mozilla
/
gecko-dev
зеркало из https://github.com/mozilla/gecko-dev.git
1
0
Форкнуть 0
Код Задачи Пакеты Проекты Релизы Вики Активность

Bug 1828517 - Vendor libwebrtc from f2a083f262 

We already cherry-picked this when we vendored 897ea04db5.

Upstream commit: https://webrtc.googlesource.com/src/+/f2a083f262d86737893e774c696716742fcab3e3
    Revert "Delete PacketReceiver::DeliverPacket from all implementations"

    This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63.

    Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200

    Original change's description:
    > Delete PacketReceiver::DeliverPacket from all implementations
    >
    > And fix tests that still depend on extensions to be known by the receiver.
    >
    > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
    >
    > Bug: webrtc:7135,webrtc:14795
    > Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
    > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
    > Commit-Queue: Per Kjellander <perkj@webrtc.org>
    > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
    > Reviewed-by: Erik Språng <sprang@webrtc.org>
    > Cr-Commit-Position: refs/heads/main@{#39184}

    Bug: webrtc:7135,webrtc:14795,b/266658815
    Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
    No-Presubmit: true
    No-Tree-Checks: true
    No-Try: true
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
    Commit-Queue: Andrey Logvin <landrey@webrtc.org>
    Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
    Owners-Override: Andrey Logvin <landrey@webrtc.org>
    Reviewed-by: Jeremy Leconte <jleconte@google.com>
    Cr-Commit-Position: refs/heads/main@{#39189}
This commit is contained in:
Michael Froman 2023-04-21 13:18:56 -05:00
Родитель 230c675c19
Коммит 9155aadeb4
100 изменённых файлов: 11320 добавлений и 11844 удалений
Показать статистику Скачать Patch файл Скачать Diff файл Показать все файлы Свернуть все файлы

3
third_party/libwebrtc/README.moz-ff-commit поставляемый
Просмотреть файл

@ -21183,3 +21183,6 @@ cc1c932f10
# MOZ_LIBWEBRTC_SRC=/home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
# base of lastest vendoring
07577b5434
# MOZ_LIBWEBRTC_SRC=/home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
# base of lastest vendoring
f2a083f262

2
third_party/libwebrtc/README.mozilla поставляемый
Просмотреть файл

@ -14144,3 +14144,5 @@ libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-l
libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-21T18:16:52.463044.
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-21T18:17:47.993957.
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
libwebrtc updated from /home/mfroman/mozilla/moz-central/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2023-04-21T18:18:44.788704.

564
third_party/libwebrtc/moz-patch-stack/0001.patch поставляемый
Просмотреть файл

@ -1,528 +1,64 @@
From: Andrey Logvin <landrey@webrtc.org>
Date: Wed, 25 Jan 2023 08:57:56 +0000
Subject: (tmp-cherry-pick) Revert "Delete PacketReceiver::DeliverPacket from
all (f2a083f262) implementations"
MIME-Version: 1.0
Content-Type: text/plain; charset=UTF-8
Content-Transfer-Encoding: 8bit
From: Alexander Cooper <alcooper@chromium.org>
Date: Wed, 8 Feb 2023 14:16:01 -0800
Subject: (cherry-pick-branch-heads/5481) Fix Destruction inside WGC Callback
This reverts commit 897ea04db5db2e591e28bd884191be58d9bcdc63.
If we are notified of the destruction of the window before a
CaptureFrame call can fail, then we may end up attempting to destroy the
underlying WGC object inside it's own event handler. This can be
problematic, as the class itself may want to run other code. Instead,
we just unsubscribe and signal that any future CaptureFrame calls should
reject.
Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200
This also removes setting "is_capture_started_=false" in the item closed
handler, as all that served to do is cause the WgcCapturerWin code to
attempt to restart the capturer, and somewhat muddies up our metrics.
Original change's description:
> Delete PacketReceiver::DeliverPacket from all implementations
>
> And fix tests that still depend on extensions to be known by the receiver.
>
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
>
> Bug: webrtc:7135,webrtc:14795
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39184}
(cherry picked from commit 318cf28945d80a0ac6f09382e507c95e649cc4c1)
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bug: chromium:1413005
No-Try: True
Change-Id: Ibccb7a2e7ce531ba80b4b331b9bc2cda0ff75f4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292762
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Original-Commit-Position: refs/heads/main@{#39275}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293246
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39189}
Cr-Commit-Position: refs/branch-heads/5481@{#5}
Cr-Branched-From: 2e1a9a4ae0234d4b1ea7a6fd4188afa1fb20379d-refs/heads/main@{#38901}
---
call/bitrate_estimator_tests.cc | 5 --
call/call.cc | 73 ++++++++++++++++++++++++++++
call/degraded_call.cc | 19 ++++++++
call/degraded_call.h | 3 ++
call/fake_network_pipe.cc | 14 ++++++
call/fake_network_pipe.h | 6 +++
call/fake_network_pipe_unittest.cc | 4 ++
call/packet_receiver.h | 26 +++++++++-
media/engine/fake_webrtc_call.cc | 15 ++++++
media/engine/fake_webrtc_call.h | 4 ++
test/direct_transport.cc | 69 ++++++++++++++++++--------
test/direct_transport.h | 9 ++++
video/end_to_end_tests/ssrc_tests.cc | 3 +-
13 files changed, 220 insertions(+), 30 deletions(-)
modules/desktop_capture/win/wgc_capture_session.cc | 13 +++++--------
1 file changed, 5 insertions(+), 8 deletions(-)
diff --git a/call/bitrate_estimator_tests.cc b/call/bitrate_estimator_tests.cc
index 6dedc59059..afa3136e0a 100644
--- a/call/bitrate_estimator_tests.cc
+++ b/call/bitrate_estimator_tests.cc
@@ -110,11 +110,6 @@ class BitrateEstimatorTest : public test::CallTest {
diff --git a/modules/desktop_capture/win/wgc_capture_session.cc b/modules/desktop_capture/win/wgc_capture_session.cc
index e16529167f..ea5565c89c 100644
--- a/modules/desktop_capture/win/wgc_capture_session.cc
+++ b/modules/desktop_capture/win/wgc_capture_session.cc
@@ -397,17 +397,14 @@ HRESULT WgcCaptureSession::OnItemClosed(WGC::IGraphicsCaptureItem* sender,
virtual void SetUp() {
SendTask(task_queue(), [this]() {
- RegisterRtpExtension(
- RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
- RegisterRtpExtension(
- RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
RTC_LOG(LS_INFO) << "Capture target has been closed.";
item_closed_ = true;
- is_capture_started_ = false;
RemoveEventHandlers();
- mapped_texture_ = nullptr;
- session_ = nullptr;
- frame_pool_ = nullptr;
- direct3d_device_ = nullptr;
- item_ = nullptr;
- d3d11_device_ = nullptr;
-
CreateCalls();
CreateSendTransport(BuiltInNetworkBehaviorConfig(), /*observer=*/nullptr);
diff --git a/call/call.cc b/call/call.cc
index e676d7a30a..218505cdea 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -241,6 +241,11 @@ class Call final : public webrtc::Call,
TaskQueueBase* network_thread() const override;
TaskQueueBase* worker_thread() const override;
+ // Implements PacketReceiver.
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) override;
+
void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override;
void DeliverRtpPacket(
@@ -334,6 +339,9 @@ class Call final : public webrtc::Call,
void DeliverRtcp(MediaType media_type, rtc::CopyOnWriteBuffer packet)
RTC_RUN_ON(network_thread_);
+ DeliveryStatus DeliverRtp(MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) RTC_RUN_ON(worker_thread_);
AudioReceiveStreamImpl* FindAudioStreamForSyncGroup(
absl::string_view sync_group) RTC_RUN_ON(worker_thread_);
@@ -343,6 +351,7 @@ class Call final : public webrtc::Call,
MediaType media_type)
RTC_RUN_ON(worker_thread_);
+ bool IdentifyReceivedPacket(RtpPacketReceived& packet);
bool RegisterReceiveStream(uint32_t ssrc, ReceiveStreamInterface* stream);
bool UnregisterReceiveStream(uint32_t ssrc);
@@ -1466,6 +1475,57 @@ void Call::DeliverRtpPacket(
}
+ // Do not attempt to free resources in the OnItemClosed handler, as this
+ // causes a race where we try to delete the item that is calling us. Removing
+ // the event handlers and setting `item_closed_` above is sufficient to ensure
+ // that the resources are no longer used, and the next time the capturer tries
+ // to get a frame, we will report a permanent failure and be destroyed.
return S_OK;
}
+PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) {
+ // TODO(perkj, https://bugs.webrtc.org/7135): Deprecate this method and
+ // direcly use DeliverRtpPacket.
+ TRACE_EVENT0("webrtc", "Call::DeliverRtp");
+ RTC_DCHECK_NE(media_type, MediaType::ANY);
+
+ RtpPacketReceived parsed_packet;
+ if (!parsed_packet.Parse(std::move(packet)))
+ return DELIVERY_PACKET_ERROR;
+
+ if (packet_time_us != -1) {
+ parsed_packet.set_arrival_time(Timestamp::Micros(packet_time_us));
+ } else {
+ parsed_packet.set_arrival_time(clock_->CurrentTime());
+ }
+
+ if (!IdentifyReceivedPacket(parsed_packet))
+ return DELIVERY_UNKNOWN_SSRC;
+ if (media_type == MediaType::VIDEO) {
+ parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+ }
+ DeliverRtpPacket(media_type, std::move(parsed_packet),
+ [](const webrtc::RtpPacketReceived& packet) {
+ // If IdentifyReceivedPacket returns true, a packet is
+ // expected to be demuxable.
+ RTC_DCHECK_NOTREACHED();
+ return false;
+ });
+ return DELIVERY_OK;
+}
+
+PacketReceiver::DeliveryStatus Call::DeliverPacket(
+ MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) {
+ if (IsRtcpPacket(packet)) {
+ RTC_DCHECK_RUN_ON(network_thread_);
+ worker_thread_->PostTask(SafeTask(
+ task_safety_.flag(), [this, packet = std::move(packet)]() mutable {
+ RTC_DCHECK_RUN_ON(worker_thread_);
+ DeliverRtcpPacket(std::move(packet));
+ }));
+ return DELIVERY_OK;
+ }
+
+ RTC_DCHECK_RUN_ON(worker_thread_);
+ return DeliverRtp(media_type, std::move(packet), packet_time_us);
+}
+
void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
MediaType media_type) {
RTC_DCHECK_RUN_ON(worker_thread_);
@@ -1489,6 +1549,19 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
}
}
+bool Call::IdentifyReceivedPacket(RtpPacketReceived& packet) {
+ RTC_DCHECK_RUN_ON(&receive_11993_checker_);
+ auto it = receive_rtp_config_.find(packet.Ssrc());
+ if (it == receive_rtp_config_.end()) {
+ RTC_DLOG(LS_WARNING) << "receive_rtp_config_ lookup failed for ssrc "
+ << packet.Ssrc();
+ return false;
+ }
+
+ packet.IdentifyExtensions(it->second->GetRtpExtensionMap());
+ return true;
+}
+
bool Call::RegisterReceiveStream(uint32_t ssrc,
ReceiveStreamInterface* stream) {
RTC_DCHECK_RUN_ON(&receive_11993_checker_);
diff --git a/call/degraded_call.cc b/call/degraded_call.cc
index fc76c7be5c..50349c1086 100644
--- a/call/degraded_call.cc
+++ b/call/degraded_call.cc
@@ -346,6 +346,25 @@ void DegradedCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
call_->OnSentPacket(sent_packet);
}
+PacketReceiver::DeliveryStatus DegradedCall::DeliverPacket(
+ MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) {
+ RTC_DCHECK_RUN_ON(&received_packet_sequence_checker_);
+ PacketReceiver::DeliveryStatus status = receive_pipe_->DeliverPacket(
+ media_type, std::move(packet), packet_time_us);
+ // This is not optimal, but there are many places where there are thread
+ // checks that fail if we're not using the worker thread call into this
+ // method. If we want to fix this we probably need a task queue to do handover
+ // of all overriden methods, which feels like overkill for the current use
+ // case.
+ // By just having this thread call out via the Process() method we work around
+ // that, with the tradeoff that a non-zero delay may become a little larger
+ // than anticipated at very low packet rates.
+ receive_pipe_->Process();
+ return status;
+}
+
void DegradedCall::DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
diff --git a/call/degraded_call.h b/call/degraded_call.h
index 98e7891d6a..6a22b69e4a 100644
--- a/call/degraded_call.h
+++ b/call/degraded_call.h
@@ -113,6 +113,9 @@ class DegradedCall : public Call, private PacketReceiver {
protected:
// Implements PacketReceiver.
+ DeliveryStatus DeliverPacket(MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) override;
void DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
diff --git a/call/fake_network_pipe.cc b/call/fake_network_pipe.cc
index 8879927a5b..76adfe3cf0 100644
--- a/call/fake_network_pipe.cc
+++ b/call/fake_network_pipe.cc
@@ -191,6 +191,16 @@ bool FakeNetworkPipe::SendRtcp(const uint8_t* packet,
return true;
}
+PacketReceiver::DeliveryStatus FakeNetworkPipe::DeliverPacket(
+ MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) {
+ return EnqueuePacket(std::move(packet), absl::nullopt, false, media_type,
+ packet_time_us)
+ ? PacketReceiver::DELIVERY_OK
+ : PacketReceiver::DELIVERY_PACKET_ERROR;
+}
+
void FakeNetworkPipe::DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
@@ -383,6 +393,10 @@ void FakeNetworkPipe::DeliverNetworkPacket(NetworkPacket* packet) {
<< packet.Ssrc() << " seq : " << packet.SequenceNumber();
return false;
});
+ } else {
+ receiver_->DeliverPacket(packet->media_type(),
+ std::move(*packet->raw_packet()),
+ packet_time_us);
}
}
}
diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index ba4c89e382..2649a00904 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h
@@ -162,6 +162,12 @@ class FakeNetworkPipe : public SimulatedPacketReceiverInterface {
OnUndemuxablePacketHandler undemuxable_packet_handler) override;
void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override;
+ // TODO(perkj, https://bugs.webrtc.org/7135): Remove once implementations
+ // dont use it.
+ PacketReceiver::DeliveryStatus DeliverPacket(MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) override;
+
// Processes the network queues and trigger PacketReceiver::IncomingPacket for
// packets ready to be delivered.
void Process() override;
diff --git a/call/fake_network_pipe_unittest.cc b/call/fake_network_pipe_unittest.cc
index 31f97fc85c..d3f7734893 100644
--- a/call/fake_network_pipe_unittest.cc
+++ b/call/fake_network_pipe_unittest.cc
@@ -31,6 +31,10 @@ using ::testing::WithArg;
namespace webrtc {
class MockReceiver : public PacketReceiver {
public:
+ MOCK_METHOD(DeliveryStatus,
+ DeliverPacket,
+ (MediaType, rtc::CopyOnWriteBuffer, int64_t),
+ (override));
MOCK_METHOD(void,
DeliverRtcpPacket,
(rtc::CopyOnWriteBuffer packet),
diff --git a/call/packet_receiver.h b/call/packet_receiver.h
index c7f55ac46c..a36ab44ea7 100644
--- a/call/packet_receiver.h
+++ b/call/packet_receiver.h
@@ -20,8 +20,26 @@ namespace webrtc {
class PacketReceiver {
public:
+ enum DeliveryStatus {
+ DELIVERY_OK,
+ DELIVERY_UNKNOWN_SSRC,
+ DELIVERY_PACKET_ERROR,
+ };
+
+ // TODO(perkj, https://bugs.webrtc.org/7135): Remove this method. This method
+ // is no longer used by PeerConnections. Some tests still use it.
+ virtual DeliveryStatus DeliverPacket(MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) {
+ RTC_CHECK_NOTREACHED();
+ }
+
// Demux RTCP packets. Must be called on the worker thread.
- virtual void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) = 0;
+ virtual void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
+ // TODO(perkj, https://bugs.webrtc.org/7135): Implement in FakeCall and
+ // FakeNetworkPipe.
+ RTC_CHECK_NOTREACHED();
+ }
// Invoked once when a packet packet is received that can not be demuxed.
// If the method returns true, a new attempt is made to demux the packet.
@@ -32,7 +50,11 @@ class PacketReceiver {
virtual void DeliverRtpPacket(
MediaType media_type,
RtpPacketReceived packet,
- OnUndemuxablePacketHandler undemuxable_packet_handler) = 0;
+ OnUndemuxablePacketHandler undemuxable_packet_handler) {
+ // TODO(perkj, https://bugs.webrtc.org/7135): Implement in FakeCall and
+ // FakeNetworkPipe.
+ RTC_CHECK_NOTREACHED();
+ }
protected:
virtual ~PacketReceiver() {}
diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc
index 6408e4e951..a20b826b41 100644
--- a/media/engine/fake_webrtc_call.cc
+++ b/media/engine/fake_webrtc_call.cc
@@ -665,6 +665,21 @@ webrtc::PacketReceiver* FakeCall::Receiver() {
return this;
}
+webrtc::PacketReceiver::DeliveryStatus FakeCall::DeliverPacket(
+ webrtc::MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) {
+ RTC_DCHECK(webrtc::IsRtpPacket(packet));
+ uint32_t ssrc = ParseRtpSsrc(packet);
+ webrtc::Timestamp arrival_time =
+ packet_time_us > -1 ? webrtc::Timestamp::Micros(packet_time_us)
+ : webrtc::Timestamp::Zero();
+ if (DeliverPacketInternal(media_type, ssrc, packet, arrival_time)) {
+ return DELIVERY_OK;
+ }
+ return DELIVERY_UNKNOWN_SSRC;
+}
+
void FakeCall::DeliverRtpPacket(
webrtc::MediaType media_type,
webrtc::RtpPacketReceived packet,
diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h
index 954bd16254..f7e3de5efb 100644
--- a/media/engine/fake_webrtc_call.h
+++ b/media/engine/fake_webrtc_call.h
@@ -442,6 +442,10 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
webrtc::PacketReceiver* Receiver() override;
+ DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
+ rtc::CopyOnWriteBuffer packet,
+ int64_t packet_time_us) override;
+
void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override {}
void DeliverRtpPacket(
diff --git a/test/direct_transport.cc b/test/direct_transport.cc
index 260497947c..3ae0216186 100644
--- a/test/direct_transport.cc
+++ b/test/direct_transport.cc
@@ -40,6 +40,18 @@ MediaType Demuxer::GetMediaType(const uint8_t* packet_data,
return MediaType::ANY;
}
+DirectTransport::DirectTransport(
+ TaskQueueBase* task_queue,
+ std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+ Call* send_call,
+ const std::map<uint8_t, MediaType>& payload_type_map)
+ : DirectTransport(task_queue,
+ std::move(pipe),
+ send_call,
+ payload_type_map,
+ {},
+ {}) {}
+
DirectTransport::DirectTransport(
TaskQueueBase* task_queue,
std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
@@ -51,6 +63,7 @@ DirectTransport::DirectTransport(
task_queue_(task_queue),
demuxer_(payload_type_map),
fake_network_(std::move(pipe)),
+ use_legacy_send_(audio_extensions.empty() && video_extensions.empty()),
audio_extensions_(audio_extensions),
video_extensions_(video_extensions) {
Start();
@@ -76,27 +89,30 @@ bool DirectTransport::SendRtp(const uint8_t* data,
send_call_->OnSentPacket(sent_packet);
}
- const RtpHeaderExtensionMap* extensions = nullptr;
- MediaType media_type = demuxer_.GetMediaType(data, length);
- switch (demuxer_.GetMediaType(data, length)) {
- case webrtc::MediaType::AUDIO:
- extensions = &audio_extensions_;
- break;
- case webrtc::MediaType::VIDEO:
- extensions = &video_extensions_;
- break;
- default:
- RTC_CHECK_NOTREACHED();
+ if (use_legacy_send_) {
+ LegacySendPacket(data, length);
+ } else {
+ const RtpHeaderExtensionMap* extensions = nullptr;
+ MediaType media_type = demuxer_.GetMediaType(data, length);
+ switch (demuxer_.GetMediaType(data, length)) {
+ case webrtc::MediaType::AUDIO:
+ extensions = &audio_extensions_;
+ break;
+ case webrtc::MediaType::VIDEO:
+ extensions = &video_extensions_;
+ break;
+ default:
+ RTC_CHECK_NOTREACHED();
+ }
+ RtpPacketReceived packet(extensions, Timestamp::Micros(rtc::TimeMicros()));
+ if (media_type == MediaType::VIDEO) {
+ packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+ }
+ RTC_CHECK(packet.Parse(rtc::CopyOnWriteBuffer(data, length)));
+ fake_network_->DeliverRtpPacket(
+ media_type, std::move(packet),
+ [](const RtpPacketReceived& packet) { return false; });
}
- RtpPacketReceived packet(extensions, Timestamp::Micros(rtc::TimeMicros()));
- if (media_type == MediaType::VIDEO) {
- packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
- }
- RTC_CHECK(packet.Parse(rtc::CopyOnWriteBuffer(data, length)));
- fake_network_->DeliverRtpPacket(
- media_type, std::move(packet),
- [](const RtpPacketReceived& packet) { return false; });
-
MutexLock lock(&process_lock_);
if (!next_process_task_.Running())
ProcessPackets();
@@ -104,13 +120,24 @@ bool DirectTransport::SendRtp(const uint8_t* data,
}
bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
- fake_network_->DeliverRtcpPacket(rtc::CopyOnWriteBuffer(data, length));
+ if (use_legacy_send_) {
+ LegacySendPacket(data, length);
+ } else {
+ fake_network_->DeliverRtcpPacket(rtc::CopyOnWriteBuffer(data, length));
+ }
MutexLock lock(&process_lock_);
if (!next_process_task_.Running())
ProcessPackets();
return true;
}
+void DirectTransport::LegacySendPacket(const uint8_t* data, size_t length) {
+ MediaType media_type = demuxer_.GetMediaType(data, length);
+ int64_t send_time_us = rtc::TimeMicros();
+ fake_network_->DeliverPacket(media_type, rtc::CopyOnWriteBuffer(data, length),
+ send_time_us);
+}
+
int DirectTransport::GetAverageDelayMs() {
return fake_network_->AverageDelay();
}
diff --git a/test/direct_transport.h b/test/direct_transport.h
index 468e339c0a..4776084ae2 100644
--- a/test/direct_transport.h
+++ b/test/direct_transport.h
@@ -44,6 +44,14 @@ class Demuxer {
// same task-queue - the one that's passed in via the constructor.
class DirectTransport : public Transport {
public:
+ // TODO(perkj, https://bugs.webrtc.org/7135): Remove header once downstream
+ // projects have been updated.
+ [[deprecated("Use ctor that provide header extensions.")]] DirectTransport(
+ TaskQueueBase* task_queue,
+ std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+ Call* send_call,
+ const std::map<uint8_t, MediaType>& payload_type_map);
+
DirectTransport(TaskQueueBase* task_queue,
std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
Call* send_call,
@@ -77,6 +85,7 @@ class DirectTransport : public Transport {
const Demuxer demuxer_;
const std::unique_ptr<SimulatedPacketReceiverInterface> fake_network_;
+ const bool use_legacy_send_;
const RtpHeaderExtensionMap audio_extensions_;
const RtpHeaderExtensionMap video_extensions_;
};
diff --git a/video/end_to_end_tests/ssrc_tests.cc b/video/end_to_end_tests/ssrc_tests.cc
index edacde115a..a3bce40fd2 100644
--- a/video/end_to_end_tests/ssrc_tests.cc
+++ b/video/end_to_end_tests/ssrc_tests.cc
@@ -108,8 +108,7 @@ TEST_F(SsrcEndToEndTest, UnknownRtpPacketTriggersUndemuxablePacketHandler) {
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())),
- receiver_call_.get(), payload_type_map_, GetRegisteredExtensions(),
- GetRegisteredExtensions());
+ receiver_call_.get(), payload_type_map_);
input_observer =
std::make_unique<PacketInputObserver>(receiver_call_->Receiver());
send_transport->SetReceiver(input_observer.get());
--
2.34.1

1990
third_party/libwebrtc/moz-patch-stack/0002.patch поставляемый
Просмотреть файл

Разница между файлами не показана из-за своего большого размера Загрузить разницу

1958
third_party/libwebrtc/moz-patch-stack/0003.patch поставляемый
Просмотреть файл

Разница между файлами не показана из-за своего большого размера Загрузить разницу

122
third_party/libwebrtc/moz-patch-stack/0004.patch поставляемый
Просмотреть файл

@ -1,46 +1,102 @@
From: Nico Grunbaum <na-g@nostrum.com>
Date: Sat, 13 Feb 2021 04:20:00 -0800
Subject: Bug 1654112 - Add pid_t to desktop_capture_types.h; r=pehrsons
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 21 Aug 2018 13:39:00 -0400
Subject: Bug 1376873 - Fix up logging in WebrtcLog.cpp; r=ng
Upstreaming bug 1697385
The webrtc::Trace code is removed by this update. We already had support for
LOG (now RTC_LOG) in WebrtcLog.cpp. This removes the trace code from
WebRtcLog.cpp and moves the aec logging code from webrtc::Trace to
rtc::LogMessage.
Also includes:
Bug 1654112 - Clarifying prev. rev that moved pid_t into the global namespace; r=dminor
This also disables logging to stderr in rtc_base/logging.cc. We could disable
it using the API, but that happens through peerconnection resulting in some
logging occuring during getusermedia.
Differential Revision: https://phabricator.services.mozilla.com/D107897
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/539b69f0e21118a1898f1ef689577c7193ff7be7
The aec logs were testing with --disable-e10s. Rather than trying to
work around sandboxing, I think it makes more sense to fix Bug 1404982 and
store the logs in memory for retrieval from about:webrtc.
Differential Revision: https://phabricator.services.mozilla.com/D7429
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e84c60e2c9373f4d2dc24e769375a92c17c2a0ad
---
modules/desktop_capture/desktop_capture_types.h | 4 ++++
modules/video_capture/video_capture.h | 1 +
2 files changed, 5 insertions(+)
.../audio_processing/logging/apm_data_dumper.cc | 2 +-
rtc_base/logging.cc | 11 ++++++++++-
rtc_base/logging.h | 14 ++++++++++++++
3 files changed, 25 insertions(+), 2 deletions(-)
diff --git a/modules/desktop_capture/desktop_capture_types.h b/modules/desktop_capture/desktop_capture_types.h
index 9627076eea..381d1021c4 100644
--- a/modules/desktop_capture/desktop_capture_types.h
+++ b/modules/desktop_capture/desktop_capture_types.h
@@ -13,6 +13,10 @@
diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc
index f787b65604..a15321ad48 100644
--- a/modules/audio_processing/logging/apm_data_dumper.cc
+++ b/modules/audio_processing/logging/apm_data_dumper.cc
@@ -42,7 +42,7 @@ std::string FormFileName(absl::string_view output_dir,
#endif
#include <stdint.h>
std::stringstream ss;
- std::string base = webrtc::Trace::aec_debug_filename();
+ std::string base = rtc::LogMessage::aec_debug_filename();
ss << base;
+#ifdef XP_WIN // Moving this into the global namespace
+typedef int pid_t; // matching what used to be in
+#endif // video_capture_defines.h
if (base.length() && base.back() != sep) {
diff --git a/rtc_base/logging.cc b/rtc_base/logging.cc
index 4bc9183d97..d71d6a3e1b 100644
--- a/rtc_base/logging.cc
+++ b/rtc_base/logging.cc
@@ -54,6 +54,15 @@ static const int kMaxLogLineSize = 1024 - 60;
#include "rtc_base/time_utils.h"
namespace rtc {
+
namespace webrtc {
+bool LogMessage::aec_debug_ = false;
+uint32_t LogMessage::aec_debug_size_ = 4*1024*1024;
+std::string LogMessage::aec_filename_base_;
+
+std::string LogMessage::aec_debug_filename() {
+ return aec_filename_base_;
+}
+
namespace {
enum class CaptureType { kWindow, kScreen };
diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
index 187eba76a0..258bc7f810 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -13,6 +13,7 @@
// By default, release builds don't log, debug builds at info level
@@ -114,7 +123,7 @@ std::string LogLineRef::DefaultLogLine() const {
// LogMessage
/////////////////////////////////////////////////////////////////////////////
#include "api/video/video_rotation.h"
#include "api/video/video_sink_interface.h"
+#include "modules/desktop_capture/desktop_capture_types.h"
#include "modules/video_capture/raw_video_sink_interface.h"
#include "modules/video_capture/video_capture_defines.h"
#include <set>
-bool LogMessage::log_to_stderr_ = true;
+bool LogMessage::log_to_stderr_ = false;
// The list of logging streams currently configured.
// Note: we explicitly do not clean this up, because of the uncertain ordering
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index d59b9a0ef7..8f490c44a2 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -581,6 +581,16 @@ class LogMessage {
}
#endif // RTC_LOG_ENABLED()
+ // Enable dumping of AEC inputs and outputs. Can be changed in mid-call
+ static void set_aec_debug(bool enable) { aec_debug_ = enable; }
+ static void set_aec_debug_size(uint32_t size) { aec_debug_size_ = size; }
+ static bool aec_debug() { return aec_debug_; }
+ static uint32_t aec_debug_size() { return aec_debug_size_; }
+ static std::string aec_debug_filename();
+ static void set_aec_debug_filename(const char* filename) {
+ aec_filename_base_ = filename;
+ }
+
private:
friend class LogMessageForTesting;
@@ -636,6 +646,10 @@ class LogMessage {
// The stringbuilder that buffers the formatted message before output
rtc::StringBuilder print_stream_;
+
+ static bool aec_debug_;
+ static uint32_t aec_debug_size_;
+ static std::string aec_filename_base_;
};
//////////////////////////////////////////////////////////////////////
--
2.34.1

115
third_party/libwebrtc/moz-patch-stack/0005.patch поставляемый
Просмотреть файл

@ -1,102 +1,37 @@
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 21 Aug 2018 13:39:00 -0400
Subject: Bug 1376873 - Fix up logging in WebrtcLog.cpp; r=ng
From: Michael Froman <mfroman@mozilla.com>
Date: Fri, 9 Jul 2021 18:14:00 -0500
Subject: Bug 1654112 - mutex changes to fix tsan errors. r=ng
The webrtc::Trace code is removed by this update. We already had support for
LOG (now RTC_LOG) in WebrtcLog.cpp. This removes the trace code from
WebRtcLog.cpp and moves the aec logging code from webrtc::Trace to
rtc::LogMessage.
This also disables logging to stderr in rtc_base/logging.cc. We could disable
it using the API, but that happens through peerconnection resulting in some
logging occuring during getusermedia.
The aec logs were testing with --disable-e10s. Rather than trying to
work around sandboxing, I think it makes more sense to fix Bug 1404982 and
store the logs in memory for retrieval from about:webrtc.
Differential Revision: https://phabricator.services.mozilla.com/D7429
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e84c60e2c9373f4d2dc24e769375a92c17c2a0ad
Differential Revision: https://phabricator.services.mozilla.com/D119674
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/3d5503acf9a4b22e02c4300f29e4fbfed406ea2c
---
.../audio_processing/logging/apm_data_dumper.cc | 2 +-
rtc_base/logging.cc | 11 ++++++++++-
rtc_base/logging.h | 14 ++++++++++++++
3 files changed, 25 insertions(+), 2 deletions(-)
rtc_base/logging.cc | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc
index f787b65604..a15321ad48 100644
--- a/modules/audio_processing/logging/apm_data_dumper.cc
+++ b/modules/audio_processing/logging/apm_data_dumper.cc
@@ -42,7 +42,7 @@ std::string FormFileName(absl::string_view output_dir,
#endif
std::stringstream ss;
- std::string base = webrtc::Trace::aec_debug_filename();
+ std::string base = rtc::LogMessage::aec_debug_filename();
ss << base;
if (base.length() && base.back() != sep) {
diff --git a/rtc_base/logging.cc b/rtc_base/logging.cc
index 4bc9183d97..d71d6a3e1b 100644
index d71d6a3e1b..d6ae7612fc 100644
--- a/rtc_base/logging.cc
+++ b/rtc_base/logging.cc
@@ -54,6 +54,15 @@ static const int kMaxLogLineSize = 1024 - 60;
#include "rtc_base/time_utils.h"
@@ -265,8 +265,8 @@ void LogMessage::LogTimestamps(bool on) {
}
namespace rtc {
+
+bool LogMessage::aec_debug_ = false;
+uint32_t LogMessage::aec_debug_size_ = 4*1024*1024;
+std::string LogMessage::aec_filename_base_;
+
+std::string LogMessage::aec_debug_filename() {
+ return aec_filename_base_;
+}
+
namespace {
void LogMessage::LogToDebug(LoggingSeverity min_sev) {
- g_dbg_sev = min_sev;
webrtc::MutexLock lock(&GetLoggingLock());
+ g_dbg_sev = min_sev;
UpdateMinLogSeverity();
}
// By default, release builds don't log, debug builds at info level
@@ -114,7 +123,7 @@ std::string LogLineRef::DefaultLogLine() const {
// LogMessage
/////////////////////////////////////////////////////////////////////////////
@@ -455,6 +455,9 @@ void LogMessage::OutputToDebug(const LogLineRef& log_line) {
-bool LogMessage::log_to_stderr_ = true;
+bool LogMessage::log_to_stderr_ = false;
// The list of logging streams currently configured.
// Note: we explicitly do not clean this up, because of the uncertain ordering
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index d59b9a0ef7..8f490c44a2 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -581,6 +581,16 @@ class LogMessage {
}
#endif // RTC_LOG_ENABLED()
+ // Enable dumping of AEC inputs and outputs. Can be changed in mid-call
+ static void set_aec_debug(bool enable) { aec_debug_ = enable; }
+ static void set_aec_debug_size(uint32_t size) { aec_debug_size_ = size; }
+ static bool aec_debug() { return aec_debug_; }
+ static uint32_t aec_debug_size() { return aec_debug_size_; }
+ static std::string aec_debug_filename();
+ static void set_aec_debug_filename(const char* filename) {
+ aec_filename_base_ = filename;
+ }
+
private:
friend class LogMessageForTesting;
@@ -636,6 +646,10 @@ class LogMessage {
// The stringbuilder that buffers the formatted message before output
rtc::StringBuilder print_stream_;
+
+ static bool aec_debug_;
+ static uint32_t aec_debug_size_;
+ static std::string aec_filename_base_;
};
//////////////////////////////////////////////////////////////////////
// static
bool LogMessage::IsNoop(LoggingSeverity severity) {
+ // Added MutexLock to fix tsan warnings on accessing g_dbg_sev. (mjf)
+ // See https://bugs.chromium.org/p/chromium/issues/detail?id=1228729
+ webrtc::MutexLock lock(&GetLoggingLock());
if (severity >= g_dbg_sev || severity >= g_min_sev)
return false;
return streams_empty_.load(std::memory_order_relaxed);
--
2.34.1

68
third_party/libwebrtc/moz-patch-stack/0006.patch поставляемый
Просмотреть файл

@ -1,37 +1,49 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Fri, 9 Jul 2021 18:14:00 -0500
Subject: Bug 1654112 - mutex changes to fix tsan errors. r=ng
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 18 Nov 2020 13:33:00 -0500
Subject: Bug 1654112 - Suppress -Wclass-varargs warning in logging.h. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D119674
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/3d5503acf9a4b22e02c4300f29e4fbfed406ea2c
This needs some investigation to see why we get this warning when it is not
present upstream.
Since both were doing the same thing for different compiler chains,
also includes:
Bug 1654112 - linux build fix (pragmas) for base-toolchains* . r=ng
Differential Revision: https://phabricator.services.mozilla.com/D130086
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8d832e832ffe513246c0763a56376a8022b2447b
---
rtc_base/logging.cc | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
rtc_base/logging.h | 13 +++++++++++++
1 file changed, 13 insertions(+)
diff --git a/rtc_base/logging.cc b/rtc_base/logging.cc
index d71d6a3e1b..d6ae7612fc 100644
--- a/rtc_base/logging.cc
+++ b/rtc_base/logging.cc
@@ -265,8 +265,8 @@ void LogMessage::LogTimestamps(bool on) {
}
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index 8f490c44a2..9340fe2c55 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -48,6 +48,14 @@
#ifndef RTC_BASE_LOGGING_H_
#define RTC_BASE_LOGGING_H_
void LogMessage::LogToDebug(LoggingSeverity min_sev) {
- g_dbg_sev = min_sev;
webrtc::MutexLock lock(&GetLoggingLock());
+ g_dbg_sev = min_sev;
UpdateMinLogSeverity();
}
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wvarargs"
+
+#if defined(__clang__)
+# pragma clang diagnostic push
+# pragma clang diagnostic ignored "-Wclass-varargs"
+#endif
+
#include <errno.h>
@@ -455,6 +455,9 @@ void LogMessage::OutputToDebug(const LogLineRef& log_line) {
#include <atomic>
@@ -769,4 +777,9 @@ inline const char* AdaptString(const std::string& str) {
// static
bool LogMessage::IsNoop(LoggingSeverity severity) {
+ // Added MutexLock to fix tsan warnings on accessing g_dbg_sev. (mjf)
+ // See https://bugs.chromium.org/p/chromium/issues/detail?id=1228729
+ webrtc::MutexLock lock(&GetLoggingLock());
if (severity >= g_dbg_sev || severity >= g_min_sev)
return false;
return streams_empty_.load(std::memory_order_relaxed);
} // namespace rtc
+#pragma GCC diagnostic pop
+#if defined(__clang__)
+# pragma clang diagnostic pop
+#endif
+
#endif // RTC_BASE_LOGGING_H_
--
2.34.1

79
third_party/libwebrtc/moz-patch-stack/0007.patch поставляемый
Просмотреть файл

@ -1,49 +1,52 @@
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 18 Nov 2020 13:33:00 -0500
Subject: Bug 1654112 - Suppress -Wclass-varargs warning in logging.h. r=ng
Date: Tue, 27 Mar 2018 15:43:00 -0400
Subject: Bug 1376873 - Disable Mid support in RtpDemuxer; r=mjf
This needs some investigation to see why we get this warning when it is not
present upstream.
The only use of Mid in the current webrtc.org code is in the unit tests.
RtpStreamReceiverController only allows adding sinks using SSRCs. Because
of this, we'll end up dropping packets in the RtpDemuxer with the current
code as none of our Mids will be recognized.
Since both were doing the same thing for different compiler chains,
also includes:
Bug 1654112 - linux build fix (pragmas) for base-toolchains* . r=ng
Tip of webrtc.org fully supports using Mids, so we'll be able to enable this
code again after the next update.
Differential Revision: https://phabricator.services.mozilla.com/D130086
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8d832e832ffe513246c0763a56376a8022b2447b
Differential Revision: https://phabricator.services.mozilla.com/D7442
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b3ba8452e77105c72f6ddbc49cbe5a53dbea1507
---
rtc_base/logging.h | 13 +++++++++++++
1 file changed, 13 insertions(+)
call/rtp_demuxer.cc | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index 8f490c44a2..9340fe2c55 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -48,6 +48,14 @@
#ifndef RTC_BASE_LOGGING_H_
#define RTC_BASE_LOGGING_H_
diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc
index 0b74f2ac0a..5c53f48144 100644
--- a/call/rtp_demuxer.cc
+++ b/call/rtp_demuxer.cc
@@ -272,13 +272,17 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
// RSID and RRID are routed to the same sinks. If an RSID is specified on a
// repair packet, it should be ignored and the RRID should be used.
std::string packet_mid, packet_rsid;
- bool has_mid = use_mid_ && packet.GetExtension<RtpMid>(&packet_mid);
+ //bool has_mid = use_mid_ && packet.GetExtension<RtpMid>(&packet_mid);
bool has_rsid = packet.GetExtension<RepairedRtpStreamId>(&packet_rsid);
if (!has_rsid) {
has_rsid = packet.GetExtension<RtpStreamId>(&packet_rsid);
}
uint32_t ssrc = packet.Ssrc();
+ // Mid support is half-baked in branch 64. RtpStreamReceiverController only
+ // supports adding sinks by ssrc, so our mids will never show up in
+ // known_mids_, causing us to drop packets here.
+#if 0
// The BUNDLE spec says to drop any packets with unknown MIDs, even if the
// SSRC is known/latched.
if (has_mid && known_mids_.find(packet_mid) == known_mids_.end()) {
@@ -352,6 +356,7 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
}
}
+#pragma GCC diagnostic push
+#pragma GCC diagnostic ignored "-Wvarargs"
+
+#if defined(__clang__)
+# pragma clang diagnostic push
+# pragma clang diagnostic ignored "-Wclass-varargs"
+#endif
+
#include <errno.h>
#include <atomic>
@@ -769,4 +777,9 @@ inline const char* AdaptString(const std::string& str) {
} // namespace rtc
+#pragma GCC diagnostic pop
+#if defined(__clang__)
+# pragma clang diagnostic pop
+#endif
+
#endif // RTC_BASE_LOGGING_H_
// We trust signaled SSRC more than payload type which is likely to conflict
// between streams.
const auto ssrc_sink_it = sink_by_ssrc_.find(ssrc);
--
2.34.1

176
third_party/libwebrtc/moz-patch-stack/0008.patch поставляемый
Просмотреть файл

@ -1,52 +1,142 @@
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 27 Mar 2018 15:43:00 -0400
Subject: Bug 1376873 - Disable Mid support in RtpDemuxer; r=mjf
Date: Wed, 7 Feb 2018 15:00:00 -0500
Subject: Bug 1376873 - Fix GetRTCPSenderReport; r=ng
The only use of Mid in the current webrtc.org code is in the unit tests.
RtpStreamReceiverController only allows adding sinks using SSRCs. Because
of this, we'll end up dropping packets in the RtpDemuxer with the current
code as none of our Mids will be recognized.
Tip of webrtc.org fully supports using Mids, so we'll be able to enable this
code again after the next update.
Differential Revision: https://phabricator.services.mozilla.com/D7442
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b3ba8452e77105c72f6ddbc49cbe5a53dbea1507
Differential Revision: https://phabricator.services.mozilla.com/D7431
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b55b0368d9f21849fa465fa9b3f028285c9ea6ae
---
call/rtp_demuxer.cc | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
call/video_receive_stream.h | 3 +++
modules/rtp_rtcp/source/rtcp_receiver.cc | 7 +++++++
modules/rtp_rtcp/source/rtcp_receiver.h | 4 ++++
modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 5 +++++
modules/rtp_rtcp/source/rtp_rtcp_impl.h | 3 +++
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc | 5 +++++
modules/rtp_rtcp/source/rtp_rtcp_impl2.h | 3 +++
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 4 ++++
8 files changed, 34 insertions(+)
diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc
index 0b74f2ac0a..5c53f48144 100644
--- a/call/rtp_demuxer.cc
+++ b/call/rtp_demuxer.cc
@@ -272,13 +272,17 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
// RSID and RRID are routed to the same sinks. If an RSID is specified on a
// repair packet, it should be ignored and the RRID should be used.
std::string packet_mid, packet_rsid;
- bool has_mid = use_mid_ && packet.GetExtension<RtpMid>(&packet_mid);
+ //bool has_mid = use_mid_ && packet.GetExtension<RtpMid>(&packet_mid);
bool has_rsid = packet.GetExtension<RepairedRtpStreamId>(&packet_rsid);
if (!has_rsid) {
has_rsid = packet.GetExtension<RtpStreamId>(&packet_rsid);
}
uint32_t ssrc = packet.Ssrc();
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 15b313a3b9..a7e82665c3 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -144,6 +144,9 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
RtpReceiveStats rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
+ // Mid support is half-baked in branch 64. RtpStreamReceiverController only
+ // supports adding sinks by ssrc, so our mids will never show up in
+ // known_mids_, causing us to drop packets here.
+#if 0
// The BUNDLE spec says to drop any packets with unknown MIDs, even if the
// SSRC is known/latched.
if (has_mid && known_mids_.find(packet_mid) == known_mids_.end()) {
@@ -352,6 +356,7 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
}
}
+ uint32_t rtcp_sender_packets_sent;
+ uint32_t rtcp_sender_octets_sent;
+
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 0a24481762..7dfe4f0b5d 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -428,6 +428,13 @@ RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
return last_xr_rtis;
}
+#endif
// We trust signaled SSRC more than payload type which is likely to conflict
// between streams.
const auto ssrc_sink_it = sink_by_ssrc_.find(ssrc);
+void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const {
+ MutexLock lock(&rtcp_receiver_lock_);
+ *packet_count = remote_sender_packet_count_;
+ *octet_count = remote_sender_octet_count_;
+}
+
std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
std::vector<ReportBlockData> result;
MutexLock lock(&rtcp_receiver_lock_);
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index cdf4cbadf8..f68e57479b 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -132,6 +132,10 @@ class RTCPReceiver final {
std::vector<rtcp::ReceiveTimeInfo> ConsumeReceivedXrReferenceTimeInfo();
+ // Get received sender packet and octet counts
+ void RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const;
+
// Get rtt.
int32_t RTT(uint32_t remote_ssrc,
int64_t* last_rtt_ms,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index b7f23236fe..54fb82c2a1 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -527,6 +527,11 @@ void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
}
// Received RTCP report.
+void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count);
+}
+
std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
const {
return rtcp_receiver_.GetLatestReportBlockData();
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index f164195168..dd916fbe40 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -192,6 +192,9 @@ class ABSL_DEPRECATED("") ModuleRtpRtcpImpl
StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const override;
+ void RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const override;
+
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
// Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index 31dd1499d5..d0f9c8ed1a 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -508,6 +508,11 @@ void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
}
// Received RTCP report.
+void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count);
+}
+
std::vector<ReportBlockData> ModuleRtpRtcpImpl2::GetLatestReportBlockData()
const {
return rtcp_receiver_.GetLatestReportBlockData();
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index e7a3ac03e8..3ef76ab66a 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -204,6 +204,9 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const override;
+ void RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const override;
+
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
// Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index 2024b308dd..f23d4d0758 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -399,6 +399,10 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const = 0;
+
+ // Returns packet count and octet count from RTCP sender report.
+ virtual void RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const = 0;
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the sender-source SSRC pair is unique and per-pair the
// ReportBlockData represents the latest Report Block that was received for
--
2.34.1

172
third_party/libwebrtc/moz-patch-stack/0009.patch поставляемый
Просмотреть файл

@ -1,139 +1,145 @@
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 7 Feb 2018 15:00:00 -0500
Subject: Bug 1376873 - Fix GetRTCPSenderReport; r=ng
From: Nico Grunbaum <na-g@nostrum.com>
Date: Fri, 5 Jun 2020 11:41:00 +0000
Subject: Bug 1615191 - P0 - implement remoteTimestamp for
RTCRemoteOutboundRtpStreamStats in libwebrtc;r=dminor
Differential Revision: https://phabricator.services.mozilla.com/D7431
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b55b0368d9f21849fa465fa9b3f028285c9ea6ae
Differential Revision: https://phabricator.services.mozilla.com/D78004
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cd901e96d23e004e4bee19b96c8c2f1ca3b42830
This patch also absorbs two additional patches that collapse into using ms for the remote timestamp:
Bug 1654112 - Plumb RemoteRTCPSenderInfo through also for audio and change unit to ms. r=ng
Bug 1654112 - Replace custom upstream code for remote received audio stats with cherry-pick. r=ng
---
call/video_receive_stream.h | 3 +++
modules/rtp_rtcp/source/rtcp_receiver.cc | 7 +++++++
modules/rtp_rtcp/source/rtcp_receiver.h | 4 ++++
modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 5 +++++
modules/rtp_rtcp/source/rtp_rtcp_impl.h | 3 +++
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc | 5 +++++
modules/rtp_rtcp/source/rtp_rtcp_impl2.h | 3 +++
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 4 ++++
8 files changed, 34 insertions(+)
call/video_receive_stream.h | 1 +
modules/rtp_rtcp/source/rtcp_receiver.cc | 4 +++-
modules/rtp_rtcp/source/rtcp_receiver.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 6 ++++--
modules/rtp_rtcp/source/rtp_rtcp_impl.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc | 6 ++++--
modules/rtp_rtcp/source/rtp_rtcp_impl2.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 5 +++--
8 files changed, 21 insertions(+), 10 deletions(-)
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 15b313a3b9..a7e82665c3 100644
index a7e82665c3..3125993d4b 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -144,6 +144,9 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
RtpReceiveStats rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
@@ -146,6 +146,7 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
uint32_t rtcp_sender_packets_sent;
uint32_t rtcp_sender_octets_sent;
+ int64_t rtcp_sender_ntp_timestamp_ms;
+ uint32_t rtcp_sender_packets_sent;
+ uint32_t rtcp_sender_octets_sent;
+
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 0a24481762..7dfe4f0b5d 100644
index 7dfe4f0b5d..68171d1c2a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -428,6 +428,13 @@ RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
return last_xr_rtis;
@@ -429,10 +429,12 @@ RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
}
+void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const {
+ MutexLock lock(&rtcp_receiver_lock_);
+ *packet_count = remote_sender_packet_count_;
+ *octet_count = remote_sender_octet_count_;
+}
+
std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
std::vector<ReportBlockData> result;
void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const {
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const {
MutexLock lock(&rtcp_receiver_lock_);
*packet_count = remote_sender_packet_count_;
*octet_count = remote_sender_octet_count_;
+ *ntp_timestamp_ms = remote_sender_ntp_time_.ToMs();
}
std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index cdf4cbadf8..f68e57479b 100644
index f68e57479b..6912912cfc 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -132,6 +132,10 @@ class RTCPReceiver final {
@@ -134,7 +134,8 @@ class RTCPReceiver final {
std::vector<rtcp::ReceiveTimeInfo> ConsumeReceivedXrReferenceTimeInfo();
// Get received sender packet and octet counts
void RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const;
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const;
+ // Get received sender packet and octet counts
+ void RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const;
+
// Get rtt.
int32_t RTT(uint32_t remote_ssrc,
int64_t* last_rtt_ms,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index b7f23236fe..54fb82c2a1 100644
index 54fb82c2a1..bf9e2b3bf9 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -527,6 +527,11 @@ void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
}
@@ -528,8 +528,10 @@ void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
// Received RTCP report.
+void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count);
+}
+
void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const {
- return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count);
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count,
+ ntp_timestamp_ms);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
const {
return rtcp_receiver_.GetLatestReportBlockData();
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index f164195168..dd916fbe40 100644
index dd916fbe40..5cf558717e 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -192,6 +192,9 @@ class ABSL_DEPRECATED("") ModuleRtpRtcpImpl
StreamDataCounters* rtp_counters,
@@ -193,7 +193,8 @@ class ABSL_DEPRECATED("") ModuleRtpRtcpImpl
StreamDataCounters* rtx_counters) const override;
+ void RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const override;
+
void RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const override;
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const override;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
// Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index 31dd1499d5..d0f9c8ed1a 100644
index d0f9c8ed1a..8378a76133 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -508,6 +508,11 @@ void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
}
@@ -509,8 +509,10 @@ void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
// Received RTCP report.
+void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count);
+}
+
void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const {
- return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count);
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count,
+ ntp_timestamp_ms);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl2::GetLatestReportBlockData()
const {
return rtcp_receiver_.GetLatestReportBlockData();
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index e7a3ac03e8..3ef76ab66a 100644
index 3ef76ab66a..4ef67d4647 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -204,6 +204,9 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
StreamDataCounters* rtp_counters,
@@ -205,7 +205,8 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
StreamDataCounters* rtx_counters) const override;
+ void RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const override;
+
void RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const override;
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const override;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
// Within this list, the ReportBlockData::RTCPReportBlock::source_ssrc(),
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index 2024b308dd..f23d4d0758 100644
index f23d4d0758..c6854937cb 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -399,6 +399,10 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
StreamDataCounters* rtp_counters,
@@ -400,9 +400,10 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
StreamDataCounters* rtx_counters) const = 0;
+
+ // Returns packet count and octet count from RTCP sender report.
+ virtual void RemoteRTCPSenderInfo(uint32_t* packet_count,
+ uint32_t* octet_count) const = 0;
- // Returns packet count and octet count from RTCP sender report.
+ // Returns packet count, octet count, and timestamp from RTCP sender report.
virtual void RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const = 0;
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const = 0;
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the sender-source SSRC pair is unique and per-pair the
// ReportBlockData represents the latest Report Block that was received for

966
third_party/libwebrtc/moz-patch-stack/0010.patch поставляемый
Просмотреть файл

@ -1,148 +1,842 @@
From: Nico Grunbaum <na-g@nostrum.com>
Date: Fri, 5 Jun 2020 11:41:00 +0000
Subject: Bug 1615191 - P0 - implement remoteTimestamp for
RTCRemoteOutboundRtpStreamStats in libwebrtc;r=dminor
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 31 Jul 2018 13:32:00 -0400
Subject: Bug 1376873 - OS X desktop capture fixes; r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D78004
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cd901e96d23e004e4bee19b96c8c2f1ca3b42830
This patch also absorbs two additional patches that collapse into using ms for the remote timestamp:
Bug 1654112 - Plumb RemoteRTCPSenderInfo through also for audio and change unit to ms. r=ng
Bug 1654112 - Replace custom upstream code for remote received audio stats with cherry-pick. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D7464
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/02c038eca65c1218b56fdf8937fdeab3d8767fe6
---
call/video_receive_stream.h | 1 +
modules/rtp_rtcp/source/rtcp_receiver.cc | 4 +++-
modules/rtp_rtcp/source/rtcp_receiver.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 6 ++++--
modules/rtp_rtcp/source/rtp_rtcp_impl.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc | 6 ++++--
modules/rtp_rtcp/source/rtp_rtcp_impl2.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 5 +++--
8 files changed, 21 insertions(+), 10 deletions(-)
.../desktop_capture/mac/screen_capturer_mac.h | 7 +
.../mac/screen_capturer_mac.mm | 4 +-
.../mouse_cursor_monitor_mac.mm | 2 +-
.../desktop_capture/screen_capturer_mac.mm | 766 ++++++++++++++++++
4 files changed, 777 insertions(+), 2 deletions(-)
create mode 100644 modules/desktop_capture/screen_capturer_mac.mm
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index a7e82665c3..3125993d4b 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -146,6 +146,7 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
diff --git a/modules/desktop_capture/mac/screen_capturer_mac.h b/modules/desktop_capture/mac/screen_capturer_mac.h
index d9a5966efa..7be05cc639 100644
--- a/modules/desktop_capture/mac/screen_capturer_mac.h
+++ b/modules/desktop_capture/mac/screen_capturer_mac.h
@@ -114,6 +114,13 @@ class ScreenCapturerMac final : public DesktopCapturer {
uint32_t rtcp_sender_packets_sent;
uint32_t rtcp_sender_octets_sent;
+ int64_t rtcp_sender_ntp_timestamp_ms;
// Start, CaptureFrame and destructor have to called in the same thread.
SequenceChecker thread_checker_;
+
+ // Used to force CaptureFrame to update it's screen configuration
+ // and reregister event handlers. This ensure that this
+ // occurs on the ScreenCapture thread. Read and written from
+ // both the VideoCapture thread and ScreenCapture thread.
+ // Protected by desktop_config_monitor_.
+ bool update_screen_configuration_ = false;
};
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 7dfe4f0b5d..68171d1c2a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -429,10 +429,12 @@ RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
}
} // namespace webrtc
diff --git a/modules/desktop_capture/mac/screen_capturer_mac.mm b/modules/desktop_capture/mac/screen_capturer_mac.mm
index 634849122e..115f6440b1 100644
--- a/modules/desktop_capture/mac/screen_capturer_mac.mm
+++ b/modules/desktop_capture/mac/screen_capturer_mac.mm
@@ -182,6 +182,7 @@ void ScreenCapturerMac::Start(Callback* callback) {
"webrtc", "ScreenCapturermac::Start", "target display id ", current_display_);
void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const {
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const {
MutexLock lock(&rtcp_receiver_lock_);
*packet_count = remote_sender_packet_count_;
*octet_count = remote_sender_octet_count_;
+ *ntp_timestamp_ms = remote_sender_ntp_time_.ToMs();
}
callback_ = callback;
+ update_screen_configuration_ = false;
// Start and operate CGDisplayStream handler all from capture thread.
if (!RegisterRefreshAndMoveHandlers()) {
RTC_LOG(LS_ERROR) << "Failed to register refresh and move handlers.";
@@ -202,7 +203,8 @@ void ScreenCapturerMac::CaptureFrame() {
}
std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index f68e57479b..6912912cfc 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -134,7 +134,8 @@ class RTCPReceiver final {
MacDesktopConfiguration new_config = desktop_config_monitor_->desktop_configuration();
- if (!desktop_config_.Equals(new_config)) {
+ if (update_screen_configuration_ || !desktop_config_.Equals(new_config)) {
+ update_screen_configuration_ = false;
desktop_config_ = new_config;
// If the display configuraiton has changed then refresh capturer data
// structures. Occasionally, the refresh and move handlers are lost when
diff --git a/modules/desktop_capture/mouse_cursor_monitor_mac.mm b/modules/desktop_capture/mouse_cursor_monitor_mac.mm
index 3db4332cd1..512103ab5e 100644
--- a/modules/desktop_capture/mouse_cursor_monitor_mac.mm
+++ b/modules/desktop_capture/mouse_cursor_monitor_mac.mm
@@ -133,7 +133,7 @@ void MouseCursorMonitorMac::CaptureImage(float scale) {
NSSize nssize = [nsimage size]; // DIP size
// Get received sender packet and octet counts
void RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const;
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const;
// No need to caputre cursor image if it's unchanged since last capture.
- if ([[nsimage TIFFRepresentation] isEqual:[last_cursor_ TIFFRepresentation]]) return;
+ if (last_cursor_ && [[nsimage TIFFRepresentation] isEqual:[last_cursor_ TIFFRepresentation]]) return;
last_cursor_ = nsimage;
// Get rtt.
int32_t RTT(uint32_t remote_ssrc,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 54fb82c2a1..bf9e2b3bf9 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -528,8 +528,10 @@ void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
// Received RTCP report.
void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const {
- return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count);
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count,
+ ntp_timestamp_ms);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index dd916fbe40..5cf558717e 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -193,7 +193,8 @@ class ABSL_DEPRECATED("") ModuleRtpRtcpImpl
StreamDataCounters* rtx_counters) const override;
void RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const override;
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const override;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index d0f9c8ed1a..8378a76133 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -509,8 +509,10 @@ void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
// Received RTCP report.
void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const {
- return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count);
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count,
+ ntp_timestamp_ms);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl2::GetLatestReportBlockData()
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index 3ef76ab66a..4ef67d4647 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -205,7 +205,8 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
StreamDataCounters* rtx_counters) const override;
void RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const override;
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const override;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index f23d4d0758..c6854937cb 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -400,9 +400,10 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
StreamDataCounters* rtx_counters) const = 0;
- // Returns packet count and octet count from RTCP sender report.
+ // Returns packet count, octet count, and timestamp from RTCP sender report.
virtual void RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count) const = 0;
+ uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const = 0;
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the sender-source SSRC pair is unique and per-pair the
// ReportBlockData represents the latest Report Block that was received for
DesktopSize size(round(nssize.width * scale),
diff --git a/modules/desktop_capture/screen_capturer_mac.mm b/modules/desktop_capture/screen_capturer_mac.mm
new file mode 100644
index 0000000000..285086ffa6
--- /dev/null
+++ b/modules/desktop_capture/screen_capturer_mac.mm
@@ -0,0 +1,766 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+
+#include <memory>
+#include <set>
+#include <utility>
+
+#include <ApplicationServices/ApplicationServices.h>
+#include <Cocoa/Cocoa.h>
+#include <CoreGraphics/CoreGraphics.h>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+#include "modules/desktop_capture/mac/scoped_pixel_buffer_object.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/screen_capturer_helper.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/macutils.h"
+#include "rtc_base/timeutils.h"
+
+namespace webrtc {
+
+namespace {
+
+// CGDisplayStreamRefs need to be destroyed asynchronously after receiving a
+// kCGDisplayStreamFrameStatusStopped callback from CoreGraphics. This may
+// happen after the ScreenCapturerMac has been destroyed. DisplayStreamManager
+// is responsible for destroying all extant CGDisplayStreamRefs, and will
+// destroy itself once it's done.
+class DisplayStreamManager {
+ public:
+ int GetUniqueId() { return ++unique_id_generator_; }
+ void DestroyStream(int unique_id) {
+ auto it = display_stream_wrappers_.find(unique_id);
+ RTC_CHECK(it != display_stream_wrappers_.end());
+ RTC_CHECK(!it->second.active);
+ CFRelease(it->second.stream);
+ display_stream_wrappers_.erase(it);
+
+ if (ready_for_self_destruction_ && display_stream_wrappers_.empty())
+ delete this;
+ }
+
+ void SaveStream(int unique_id,
+ CGDisplayStreamRef stream) {
+ RTC_CHECK(unique_id <= unique_id_generator_);
+ DisplayStreamWrapper wrapper;
+ wrapper.stream = stream;
+ display_stream_wrappers_[unique_id] = wrapper;
+ }
+
+ void UnregisterActiveStreams() {
+ for (auto& pair : display_stream_wrappers_) {
+ DisplayStreamWrapper& wrapper = pair.second;
+ if (wrapper.active) {
+ wrapper.active = false;
+ CFRunLoopSourceRef source =
+ CGDisplayStreamGetRunLoopSource(wrapper.stream);
+ CFRunLoopRemoveSource(CFRunLoopGetCurrent(), source,
+ kCFRunLoopCommonModes);
+ CGDisplayStreamStop(wrapper.stream);
+ }
+ }
+ }
+
+ void PrepareForSelfDestruction() {
+ ready_for_self_destruction_ = true;
+
+ if (display_stream_wrappers_.empty())
+ delete this;
+ }
+
+ // Once the DisplayStreamManager is ready for destruction, the
+ // ScreenCapturerMac is no longer present. Any updates should be ignored.
+ bool ShouldIgnoreUpdates() { return ready_for_self_destruction_; }
+
+ private:
+ struct DisplayStreamWrapper {
+ // The registered CGDisplayStreamRef.
+ CGDisplayStreamRef stream = nullptr;
+
+ // Set to false when the stream has been stopped. An asynchronous callback
+ // from CoreGraphics will let us destroy the CGDisplayStreamRef.
+ bool active = true;
+ };
+
+ std::map<int, DisplayStreamWrapper> display_stream_wrappers_;
+ int unique_id_generator_ = 0;
+ bool ready_for_self_destruction_ = false;
+};
+
+// Standard Mac displays have 72dpi, but we report 96dpi for
+// consistency with Windows and Linux.
+const int kStandardDPI = 96;
+
+// Scales all coordinates of a rect by a specified factor.
+DesktopRect ScaleAndRoundCGRect(const CGRect& rect, float scale) {
+ return DesktopRect::MakeLTRB(
+ static_cast<int>(floor(rect.origin.x * scale)),
+ static_cast<int>(floor(rect.origin.y * scale)),
+ static_cast<int>(ceil((rect.origin.x + rect.size.width) * scale)),
+ static_cast<int>(ceil((rect.origin.y + rect.size.height) * scale)));
+}
+
+// Copy pixels in the |rect| from |src_place| to |dest_plane|. |rect| should be
+// relative to the origin of |src_plane| and |dest_plane|.
+void CopyRect(const uint8_t* src_plane,
+ int src_plane_stride,
+ uint8_t* dest_plane,
+ int dest_plane_stride,
+ int bytes_per_pixel,
+ const DesktopRect& rect) {
+ // Get the address of the starting point.
+ const int src_y_offset = src_plane_stride * rect.top();
+ const int dest_y_offset = dest_plane_stride * rect.top();
+ const int x_offset = bytes_per_pixel * rect.left();
+ src_plane += src_y_offset + x_offset;
+ dest_plane += dest_y_offset + x_offset;
+
+ // Copy pixels in the rectangle line by line.
+ const int bytes_per_line = bytes_per_pixel * rect.width();
+ const int height = rect.height();
+ for (int i = 0 ; i < height; ++i) {
+ memcpy(dest_plane, src_plane, bytes_per_line);
+ src_plane += src_plane_stride;
+ dest_plane += dest_plane_stride;
+ }
+}
+
+// Returns an array of CGWindowID for all the on-screen windows except
+// |window_to_exclude|, or NULL if the window is not found or it fails. The
+// caller should release the returned CFArrayRef.
+CFArrayRef CreateWindowListWithExclusion(CGWindowID window_to_exclude) {
+ if (!window_to_exclude)
+ return nullptr;
+
+ CFArrayRef all_windows = CGWindowListCopyWindowInfo(
+ kCGWindowListOptionOnScreenOnly, kCGNullWindowID);
+ if (!all_windows)
+ return nullptr;
+
+ CFMutableArrayRef returned_array =
+ CFArrayCreateMutable(nullptr, CFArrayGetCount(all_windows), nullptr);
+
+ bool found = false;
+ for (CFIndex i = 0; i < CFArrayGetCount(all_windows); ++i) {
+ CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
+ CFArrayGetValueAtIndex(all_windows, i));
+
+ CFNumberRef id_ref = reinterpret_cast<CFNumberRef>(
+ CFDictionaryGetValue(window, kCGWindowNumber));
+
+ CGWindowID id;
+ CFNumberGetValue(id_ref, kCFNumberIntType, &id);
+ if (id == window_to_exclude) {
+ found = true;
+ continue;
+ }
+ CFArrayAppendValue(returned_array, reinterpret_cast<void *>(id));
+ }
+ CFRelease(all_windows);
+
+ if (!found) {
+ CFRelease(returned_array);
+ returned_array = nullptr;
+ }
+ return returned_array;
+}
+
+// Returns the bounds of |window| in physical pixels, enlarged by a small amount
+// on four edges to take account of the border/shadow effects.
+DesktopRect GetExcludedWindowPixelBounds(CGWindowID window,
+ float dip_to_pixel_scale) {
+ // The amount of pixels to add to the actual window bounds to take into
+ // account of the border/shadow effects.
+ static const int kBorderEffectSize = 20;
+ CGRect rect;
+ CGWindowID ids[1];
+ ids[0] = window;
+
+ CFArrayRef window_id_array =
+ CFArrayCreate(nullptr, reinterpret_cast<const void**>(&ids), 1, nullptr);
+ CFArrayRef window_array =
+ CGWindowListCreateDescriptionFromArray(window_id_array);
+
+ if (CFArrayGetCount(window_array) > 0) {
+ CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
+ CFArrayGetValueAtIndex(window_array, 0));
+ CFDictionaryRef bounds_ref = reinterpret_cast<CFDictionaryRef>(
+ CFDictionaryGetValue(window, kCGWindowBounds));
+ CGRectMakeWithDictionaryRepresentation(bounds_ref, &rect);
+ }
+
+ CFRelease(window_id_array);
+ CFRelease(window_array);
+
+ rect.origin.x -= kBorderEffectSize;
+ rect.origin.y -= kBorderEffectSize;
+ rect.size.width += kBorderEffectSize * 2;
+ rect.size.height += kBorderEffectSize * 2;
+ // |rect| is in DIP, so convert to physical pixels.
+ return ScaleAndRoundCGRect(rect, dip_to_pixel_scale);
+}
+
+// Create an image of the given region using the given |window_list|.
+// |pixel_bounds| should be in the primary display's coordinate in physical
+// pixels. The caller should release the returned CGImageRef and CFDataRef.
+CGImageRef CreateExcludedWindowRegionImage(const DesktopRect& pixel_bounds,
+ float dip_to_pixel_scale,
+ CFArrayRef window_list) {
+ CGRect window_bounds;
+ // The origin is in DIP while the size is in physical pixels. That's what
+ // CGWindowListCreateImageFromArray expects.
+ window_bounds.origin.x = pixel_bounds.left() / dip_to_pixel_scale;
+ window_bounds.origin.y = pixel_bounds.top() / dip_to_pixel_scale;
+ window_bounds.size.width = pixel_bounds.width();
+ window_bounds.size.height = pixel_bounds.height();
+
+ return CGWindowListCreateImageFromArray(
+ window_bounds, window_list, kCGWindowImageDefault);
+}
+
+// A class to perform video frame capturing for mac.
+class ScreenCapturerMac : public DesktopCapturer {
+ public:
+ explicit ScreenCapturerMac(
+ rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
+ bool detect_updated_region);
+ ~ScreenCapturerMac() override;
+
+ bool Init();
+
+ // DesktopCapturer interface.
+ void Start(Callback* callback) override;
+ void CaptureFrame() override;
+ void SetExcludedWindow(WindowId window) override;
+ bool GetSourceList(SourceList* screens) override;
+ bool SelectSource(SourceId id) override;
+
+ private:
+ // Returns false if the selected screen is no longer valid.
+ bool CgBlit(const DesktopFrame& frame, const DesktopRegion& region);
+
+ // Called when the screen configuration is changed.
+ void ScreenConfigurationChanged();
+
+ bool RegisterRefreshAndMoveHandlers();
+ void UnregisterRefreshAndMoveHandlers();
+
+ void ScreenRefresh(CGRectCount count,
+ const CGRect *rect_array,
+ DesktopVector display_origin);
+ void ReleaseBuffers();
+
+ std::unique_ptr<DesktopFrame> CreateFrame();
+
+ const bool detect_updated_region_;
+
+ Callback* callback_ = nullptr;
+
+ ScopedPixelBufferObject pixel_buffer_object_;
+
+ // Queue of the frames buffers.
+ ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
+
+ // Current display configuration.
+ MacDesktopConfiguration desktop_config_;
+
+ // Currently selected display, or 0 if the full desktop is selected. On OS X
+ // 10.6 and before, this is always 0.
+ CGDirectDisplayID current_display_ = 0;
+
+ // The physical pixel bounds of the current screen.
+ DesktopRect screen_pixel_bounds_;
+
+ // The dip to physical pixel scale of the current screen.
+ float dip_to_pixel_scale_ = 1.0f;
+
+ // A thread-safe list of invalid rectangles, and the size of the most
+ // recently captured screen.
+ ScreenCapturerHelper helper_;
+
+ // Contains an invalid region from the previous capture.
+ DesktopRegion last_invalid_region_;
+
+ // Monitoring display reconfiguration.
+ rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor_;
+
+ CGWindowID excluded_window_ = 0;
+
+ // A self-owned object that will destroy itself after ScreenCapturerMac and
+ // all display streams have been destroyed..
+ DisplayStreamManager* display_stream_manager_;
+
+ // Used to force CaptureFrame to update it's screen configuration
+ // and reregister event handlers. This ensure that this
+ // occurs on the ScreenCapture thread. Read and written from
+ // both the VideoCapture thread and ScreenCapture thread.
+ // Protected by desktop_config_monitor_.
+ bool update_screen_configuration_ = false;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerMac);
+};
+
+// DesktopFrame wrapper that flips wrapped frame upside down by inverting
+// stride.
+class InvertedDesktopFrame : public DesktopFrame {
+ public:
+ InvertedDesktopFrame(std::unique_ptr<DesktopFrame> frame)
+ : DesktopFrame(
+ frame->size(),
+ -frame->stride(),
+ frame->data() + (frame->size().height() - 1) * frame->stride(),
+ frame->shared_memory()) {
+ original_frame_ = std::move(frame);
+ MoveFrameInfoFrom(original_frame_.get());
+ }
+ ~InvertedDesktopFrame() override {}
+
+ private:
+ std::unique_ptr<DesktopFrame> original_frame_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(InvertedDesktopFrame);
+};
+
+ScreenCapturerMac::ScreenCapturerMac(
+ rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
+ bool detect_updated_region)
+ : detect_updated_region_(detect_updated_region),
+ desktop_config_monitor_(desktop_config_monitor) {
+ display_stream_manager_ = new DisplayStreamManager;
+}
+
+ScreenCapturerMac::~ScreenCapturerMac() {
+ ReleaseBuffers();
+ UnregisterRefreshAndMoveHandlers();
+ display_stream_manager_->PrepareForSelfDestruction();
+}
+
+bool ScreenCapturerMac::Init() {
+ desktop_config_monitor_->Lock();
+ desktop_config_ = desktop_config_monitor_->desktop_configuration();
+ desktop_config_monitor_->Unlock();
+ if (!RegisterRefreshAndMoveHandlers()) {
+ return false;
+ }
+ ScreenConfigurationChanged();
+ return true;
+}
+
+void ScreenCapturerMac::ReleaseBuffers() {
+ // The buffers might be in use by the encoder, so don't delete them here.
+ // Instead, mark them as "needs update"; next time the buffers are used by
+ // the capturer, they will be recreated if necessary.
+ queue_.Reset();
+}
+
+void ScreenCapturerMac::Start(Callback* callback) {
+ assert(!callback_);
+ assert(callback);
+
+ callback_ = callback;
+ desktop_config_monitor_->Lock();
+ update_screen_configuration_ = true;
+ desktop_config_monitor_->Unlock();
+}
+
+void ScreenCapturerMac::CaptureFrame() {
+ int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+ // Spin RunLoop for 1/100th of a second, handling at most one source
+ CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.01, true);
+
+ queue_.MoveToNextFrame();
+ RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
+
+ desktop_config_monitor_->Lock();
+ MacDesktopConfiguration new_config =
+ desktop_config_monitor_->desktop_configuration();
+ if (update_screen_configuration_ || !desktop_config_.Equals(new_config)) {
+ update_screen_configuration_ = false;
+ desktop_config_ = new_config;
+ // If the display configuraiton has changed then refresh capturer data
+ // structures. Occasionally, the refresh and move handlers are lost when
+ // the screen mode changes, so re-register them here.
+ UnregisterRefreshAndMoveHandlers();
+ RegisterRefreshAndMoveHandlers();
+ ScreenConfigurationChanged();
+ }
+
+ DesktopRegion region;
+ helper_.TakeInvalidRegion(&region);
+
+ // If the current buffer is from an older generation then allocate a new one.
+ // Note that we can't reallocate other buffers at this point, since the caller
+ // may still be reading from them.
+ if (!queue_.current_frame())
+ queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(CreateFrame()));
+
+ DesktopFrame* current_frame = queue_.current_frame();
+
+ if (!CgBlit(*current_frame, region)) {
+ desktop_config_monitor_->Unlock();
+ callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+ return;
+ }
+ std::unique_ptr<DesktopFrame> new_frame = queue_.current_frame()->Share();
+ if (detect_updated_region_) {
+ *new_frame->mutable_updated_region() = region;
+ } else {
+ new_frame->mutable_updated_region()->AddRect(
+ DesktopRect::MakeSize(new_frame->size()));
+ }
+
+ if (current_display_) {
+ const MacDisplayConfiguration* config =
+ desktop_config_.FindDisplayConfigurationById(current_display_);
+ if (config) {
+ new_frame->set_top_left(config->bounds.top_left().subtract(
+ desktop_config_.bounds.top_left()));
+ }
+ }
+
+ helper_.set_size_most_recent(new_frame->size());
+
+ // Signal that we are done capturing data from the display framebuffer,
+ // and accessing display structures.
+ desktop_config_monitor_->Unlock();
+
+ new_frame->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
+ rtc::kNumNanosecsPerMillisec);
+ callback_->OnCaptureResult(Result::SUCCESS, std::move(new_frame));
+}
+
+void ScreenCapturerMac::SetExcludedWindow(WindowId window) {
+ excluded_window_ = window;
+}
+
+bool ScreenCapturerMac::GetSourceList(SourceList* screens) {
+ assert(screens->size() == 0);
+
+ for (MacDisplayConfigurations::iterator it = desktop_config_.displays.begin();
+ it != desktop_config_.displays.end(); ++it) {
+ screens->push_back({it->id});
+ }
+ return true;
+}
+
+bool ScreenCapturerMac::SelectSource(SourceId id) {
+ if (id == kFullDesktopScreenId) {
+ current_display_ = 0;
+ } else {
+ const MacDisplayConfiguration* config =
+ desktop_config_.FindDisplayConfigurationById(
+ static_cast<CGDirectDisplayID>(id));
+ if (!config)
+ return false;
+ current_display_ = config->id;
+ }
+
+ ScreenConfigurationChanged();
+ return true;
+}
+
+bool ScreenCapturerMac::CgBlit(const DesktopFrame& frame, const DesktopRegion& region) {
+ // Copy the entire contents of the previous capture buffer, to capture over.
+ // TODO(wez): Get rid of this as per crbug.com/145064, or implement
+ // crbug.com/92354.
+ if (queue_.previous_frame()) {
+ memcpy(frame.data(), queue_.previous_frame()->data(),
+ frame.stride() * frame.size().height());
+ }
+
+ MacDisplayConfigurations displays_to_capture;
+ if (current_display_) {
+ // Capturing a single screen. Note that the screen id may change when
+ // screens are added or removed.
+ const MacDisplayConfiguration* config =
+ desktop_config_.FindDisplayConfigurationById(current_display_);
+ if (config) {
+ displays_to_capture.push_back(*config);
+ } else {
+ RTC_LOG(LS_ERROR) << "The selected screen cannot be found for capturing.";
+ return false;
+ }
+ } else {
+ // Capturing the whole desktop.
+ displays_to_capture = desktop_config_.displays;
+ }
+
+ // Create the window list once for all displays.
+ CFArrayRef window_list = CreateWindowListWithExclusion(excluded_window_);
+
+ for (size_t i = 0; i < displays_to_capture.size(); ++i) {
+ const MacDisplayConfiguration& display_config = displays_to_capture[i];
+
+ // Capturing mixed-DPI on one surface is hard, so we only return displays
+ // that match the "primary" display's DPI. The primary display is always
+ // the first in the list.
+ if (i > 0 && display_config.dip_to_pixel_scale !=
+ displays_to_capture[0].dip_to_pixel_scale) {
+ continue;
+ }
+ // Determine the display's position relative to the desktop, in pixels.
+ DesktopRect display_bounds = display_config.pixel_bounds;
+ display_bounds.Translate(-screen_pixel_bounds_.left(),
+ -screen_pixel_bounds_.top());
+
+ // Determine which parts of the blit region, if any, lay within the monitor.
+ DesktopRegion copy_region = region;
+ copy_region.IntersectWith(display_bounds);
+ if (copy_region.is_empty())
+ continue;
+
+ // Translate the region to be copied into display-relative coordinates.
+ copy_region.Translate(-display_bounds.left(), -display_bounds.top());
+
+ DesktopRect excluded_window_bounds;
+ CGImageRef excluded_image = nullptr;
+ if (excluded_window_ && window_list) {
+ // Get the region of the excluded window relative the primary display.
+ excluded_window_bounds = GetExcludedWindowPixelBounds(
+ excluded_window_, display_config.dip_to_pixel_scale);
+ excluded_window_bounds.IntersectWith(display_config.pixel_bounds);
+
+ // Create the image under the excluded window first, because it's faster
+ // than captuing the whole display.
+ if (!excluded_window_bounds.is_empty()) {
+ excluded_image = CreateExcludedWindowRegionImage(
+ excluded_window_bounds, display_config.dip_to_pixel_scale,
+ window_list);
+ }
+ }
+
+ // Create an image containing a snapshot of the display.
+ CGImageRef image = CGDisplayCreateImage(display_config.id);
+ if (!image) {
+ if (excluded_image)
+ CFRelease(excluded_image);
+ continue;
+ }
+
+ // Verify that the image has 32-bit depth.
+ int bits_per_pixel = CGImageGetBitsPerPixel(image);
+ if (bits_per_pixel / 8 != DesktopFrame::kBytesPerPixel) {
+ RTC_LOG(LS_ERROR) << "CGDisplayCreateImage() returned imaged with " << bits_per_pixel
+ << " bits per pixel. Only 32-bit depth is supported.";
+ CFRelease(image);
+ if (excluded_image)
+ CFRelease(excluded_image);
+ return false;
+ }
+
+ // Request access to the raw pixel data via the image's DataProvider.
+ CGDataProviderRef provider = CGImageGetDataProvider(image);
+ CFDataRef data = CGDataProviderCopyData(provider);
+ assert(data);
+
+ const uint8_t* display_base_address = CFDataGetBytePtr(data);
+ int src_bytes_per_row = CGImageGetBytesPerRow(image);
+
+ // |image| size may be different from display_bounds in case the screen was
+ // resized recently.
+ copy_region.IntersectWith(
+ DesktopRect::MakeWH(CGImageGetWidth(image), CGImageGetHeight(image)));
+
+ // Copy the dirty region from the display buffer into our desktop buffer.
+ uint8_t* out_ptr = frame.GetFrameDataAtPos(display_bounds.top_left());
+ for (DesktopRegion::Iterator i(copy_region); !i.IsAtEnd(); i.Advance()) {
+ CopyRect(display_base_address, src_bytes_per_row, out_ptr, frame.stride(),
+ DesktopFrame::kBytesPerPixel, i.rect());
+ }
+
+ CFRelease(data);
+ CFRelease(image);
+
+ if (excluded_image) {
+ CGDataProviderRef provider = CGImageGetDataProvider(excluded_image);
+ CFDataRef excluded_image_data = CGDataProviderCopyData(provider);
+ assert(excluded_image_data);
+ display_base_address = CFDataGetBytePtr(excluded_image_data);
+ src_bytes_per_row = CGImageGetBytesPerRow(excluded_image);
+
+ // Translate the bounds relative to the desktop, because |frame| data
+ // starts from the desktop top-left corner.
+ DesktopRect window_bounds_relative_to_desktop(excluded_window_bounds);
+ window_bounds_relative_to_desktop.Translate(-screen_pixel_bounds_.left(),
+ -screen_pixel_bounds_.top());
+
+ DesktopRect rect_to_copy =
+ DesktopRect::MakeSize(excluded_window_bounds.size());
+ rect_to_copy.IntersectWith(DesktopRect::MakeWH(
+ CGImageGetWidth(excluded_image), CGImageGetHeight(excluded_image)));
+
+ if (CGImageGetBitsPerPixel(excluded_image) / 8 ==
+ DesktopFrame::kBytesPerPixel) {
+ CopyRect(display_base_address, src_bytes_per_row,
+ frame.GetFrameDataAtPos(
+ window_bounds_relative_to_desktop.top_left()),
+ frame.stride(), DesktopFrame::kBytesPerPixel, rect_to_copy);
+ }
+
+ CFRelease(excluded_image_data);
+ CFRelease(excluded_image);
+ }
+ }
+ if (window_list)
+ CFRelease(window_list);
+ return true;
+}
+
+void ScreenCapturerMac::ScreenConfigurationChanged() {
+ if (current_display_) {
+ const MacDisplayConfiguration* config =
+ desktop_config_.FindDisplayConfigurationById(current_display_);
+ screen_pixel_bounds_ = config ? config->pixel_bounds : DesktopRect();
+ dip_to_pixel_scale_ = config ? config->dip_to_pixel_scale : 1.0f;
+ } else {
+ screen_pixel_bounds_ = desktop_config_.pixel_bounds;
+ dip_to_pixel_scale_ = desktop_config_.dip_to_pixel_scale;
+ }
+
+ // Release existing buffers, which will be of the wrong size.
+ ReleaseBuffers();
+
+ // Clear the dirty region, in case the display is down-sizing.
+ helper_.ClearInvalidRegion();
+
+ // Re-mark the entire desktop as dirty.
+ helper_.InvalidateScreen(screen_pixel_bounds_.size());
+
+ // Make sure the frame buffers will be reallocated.
+ queue_.Reset();
+}
+
+bool ScreenCapturerMac::RegisterRefreshAndMoveHandlers() {
+ desktop_config_ = desktop_config_monitor_->desktop_configuration();
+ for (const auto& config : desktop_config_.displays) {
+ size_t pixel_width = config.pixel_bounds.width();
+ size_t pixel_height = config.pixel_bounds.height();
+ if (pixel_width == 0 || pixel_height == 0)
+ continue;
+ // Using a local variable forces the block to capture the raw pointer.
+ DisplayStreamManager* manager = display_stream_manager_;
+ int unique_id = manager->GetUniqueId();
+ CGDirectDisplayID display_id = config.id;
+ DesktopVector display_origin = config.pixel_bounds.top_left();
+
+ CGDisplayStreamFrameAvailableHandler handler =
+ ^(CGDisplayStreamFrameStatus status, uint64_t display_time,
+ IOSurfaceRef frame_surface, CGDisplayStreamUpdateRef updateRef) {
+ if (status == kCGDisplayStreamFrameStatusStopped) {
+ manager->DestroyStream(unique_id);
+ return;
+ }
+
+ if (manager->ShouldIgnoreUpdates())
+ return;
+
+ // Only pay attention to frame updates.
+ if (status != kCGDisplayStreamFrameStatusFrameComplete)
+ return;
+
+ size_t count = 0;
+ const CGRect* rects = CGDisplayStreamUpdateGetRects(
+ updateRef, kCGDisplayStreamUpdateDirtyRects, &count);
+ if (count != 0) {
+ // According to CGDisplayStream.h, it's safe to call
+ // CGDisplayStreamStop() from within the callback.
+ ScreenRefresh(count, rects, display_origin);
+ }
+ };
+ CGDisplayStreamRef display_stream = CGDisplayStreamCreate(
+ display_id, pixel_width, pixel_height, 'BGRA', nullptr, handler);
+
+ if (display_stream) {
+ CGError error = CGDisplayStreamStart(display_stream);
+ if (error != kCGErrorSuccess)
+ return false;
+
+ CFRunLoopSourceRef source =
+ CGDisplayStreamGetRunLoopSource(display_stream);
+ CFRunLoopAddSource(CFRunLoopGetCurrent(), source, kCFRunLoopCommonModes);
+ display_stream_manager_->SaveStream(unique_id, display_stream);
+ }
+ }
+
+ return true;
+}
+
+void ScreenCapturerMac::UnregisterRefreshAndMoveHandlers() {
+ display_stream_manager_->UnregisterActiveStreams();
+}
+
+void ScreenCapturerMac::ScreenRefresh(CGRectCount count,
+ const CGRect* rect_array,
+ DesktopVector display_origin) {
+ if (screen_pixel_bounds_.is_empty())
+ ScreenConfigurationChanged();
+
+ // The refresh rects are in display coordinates. We want to translate to
+ // framebuffer coordinates. If a specific display is being captured, then no
+ // change is necessary. If all displays are being captured, then we want to
+ // translate by the origin of the display.
+ DesktopVector translate_vector;
+ if (!current_display_)
+ translate_vector = display_origin;
+
+ DesktopRegion region;
+ for (CGRectCount i = 0; i < count; ++i) {
+ // All rects are already in physical pixel coordinates.
+ DesktopRect rect = DesktopRect::MakeXYWH(
+ rect_array[i].origin.x, rect_array[i].origin.y,
+ rect_array[i].size.width, rect_array[i].size.height);
+
+ rect.Translate(translate_vector);
+
+ region.AddRect(rect);
+ }
+
+ helper_.InvalidateRegion(region);
+}
+
+std::unique_ptr<DesktopFrame> ScreenCapturerMac::CreateFrame() {
+ std::unique_ptr<DesktopFrame> frame(
+ new BasicDesktopFrame(screen_pixel_bounds_.size()));
+ frame->set_dpi(DesktopVector(kStandardDPI * dip_to_pixel_scale_,
+ kStandardDPI * dip_to_pixel_scale_));
+ return frame;
+}
+
+} // namespace
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
+ const DesktopCaptureOptions& options) {
+ if (!options.configuration_monitor())
+ return nullptr;
+
+ std::unique_ptr<ScreenCapturerMac> capturer(new ScreenCapturerMac(
+ options.configuration_monitor(), options.detect_updated_region()));
+ if (!capturer.get()->Init()) {
+ return nullptr;
+ }
+
+ return capturer;
+}
+
+} // namespace webrtc
--
2.34.1

878
third_party/libwebrtc/moz-patch-stack/0011.patch поставляемый
Просмотреть файл

@ -1,842 +1,56 @@
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 31 Jul 2018 13:32:00 -0400
Subject: Bug 1376873 - OS X desktop capture fixes; r=pehrsons
From: "Byron Campen [:bwc]" <docfaraday@gmail.com>
Date: Fri, 12 Feb 2021 14:27:00 +0000
Subject: Bug 1654112 - Get OS X build working.
r=ng,firefox-build-system-reviewers,glandium
Differential Revision: https://phabricator.services.mozilla.com/D7464
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/02c038eca65c1218b56fdf8937fdeab3d8767fe6
* Pull in sdk/objc/base and sdk/objc/helpers
* Add gclient_args.gni to keep build happy.
* Add a missing include path for libyuv
* Support .m files in build.
Differential Revision: https://phabricator.services.mozilla.com/D105015
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9314046d89ebc0836a50ab7addece71abbf0b5e4
---
.../desktop_capture/mac/screen_capturer_mac.h | 7 +
.../mac/screen_capturer_mac.mm | 4 +-
.../mouse_cursor_monitor_mac.mm | 2 +-
.../desktop_capture/screen_capturer_mac.mm | 766 ++++++++++++++++++
4 files changed, 777 insertions(+), 2 deletions(-)
create mode 100644 modules/desktop_capture/screen_capturer_mac.mm
modules/desktop_capture/mac/screen_capturer_mac.mm | 3 ++-
modules/desktop_capture/mac/window_list_utils.cc | 4 ++--
2 files changed, 4 insertions(+), 3 deletions(-)
diff --git a/modules/desktop_capture/mac/screen_capturer_mac.h b/modules/desktop_capture/mac/screen_capturer_mac.h
index d9a5966efa..7be05cc639 100644
--- a/modules/desktop_capture/mac/screen_capturer_mac.h
+++ b/modules/desktop_capture/mac/screen_capturer_mac.h
@@ -114,6 +114,13 @@ class ScreenCapturerMac final : public DesktopCapturer {
// Start, CaptureFrame and destructor have to called in the same thread.
SequenceChecker thread_checker_;
+
+ // Used to force CaptureFrame to update it's screen configuration
+ // and reregister event handlers. This ensure that this
+ // occurs on the ScreenCapture thread. Read and written from
+ // both the VideoCapture thread and ScreenCapture thread.
+ // Protected by desktop_config_monitor_.
+ bool update_screen_configuration_ = false;
};
} // namespace webrtc
diff --git a/modules/desktop_capture/mac/screen_capturer_mac.mm b/modules/desktop_capture/mac/screen_capturer_mac.mm
index 634849122e..115f6440b1 100644
index 115f6440b1..cad0c5b65b 100644
--- a/modules/desktop_capture/mac/screen_capturer_mac.mm
+++ b/modules/desktop_capture/mac/screen_capturer_mac.mm
@@ -182,6 +182,7 @@ void ScreenCapturerMac::Start(Callback* callback) {
"webrtc", "ScreenCapturermac::Start", "target display id ", current_display_);
callback_ = callback;
+ update_screen_configuration_ = false;
// Start and operate CGDisplayStream handler all from capture thread.
if (!RegisterRefreshAndMoveHandlers()) {
RTC_LOG(LS_ERROR) << "Failed to register refresh and move handlers.";
@@ -202,7 +203,8 @@ void ScreenCapturerMac::CaptureFrame() {
@@ -276,7 +276,8 @@ bool ScreenCapturerMac::GetSourceList(SourceList* screens) {
for (MacDisplayConfigurations::iterator it = desktop_config_.displays.begin();
it != desktop_config_.displays.end();
++it) {
- screens->push_back({it->id, std::string()});
+ Source value = {it->id, 0, std::string()};
+ screens->push_back(value);
}
MacDesktopConfiguration new_config = desktop_config_monitor_->desktop_configuration();
- if (!desktop_config_.Equals(new_config)) {
+ if (update_screen_configuration_ || !desktop_config_.Equals(new_config)) {
+ update_screen_configuration_ = false;
desktop_config_ = new_config;
// If the display configuraiton has changed then refresh capturer data
// structures. Occasionally, the refresh and move handlers are lost when
diff --git a/modules/desktop_capture/mouse_cursor_monitor_mac.mm b/modules/desktop_capture/mouse_cursor_monitor_mac.mm
index 3db4332cd1..512103ab5e 100644
--- a/modules/desktop_capture/mouse_cursor_monitor_mac.mm
+++ b/modules/desktop_capture/mouse_cursor_monitor_mac.mm
@@ -133,7 +133,7 @@ void MouseCursorMonitorMac::CaptureImage(float scale) {
NSSize nssize = [nsimage size]; // DIP size
// No need to caputre cursor image if it's unchanged since last capture.
- if ([[nsimage TIFFRepresentation] isEqual:[last_cursor_ TIFFRepresentation]]) return;
+ if (last_cursor_ && [[nsimage TIFFRepresentation] isEqual:[last_cursor_ TIFFRepresentation]]) return;
last_cursor_ = nsimage;
DesktopSize size(round(nssize.width * scale),
diff --git a/modules/desktop_capture/screen_capturer_mac.mm b/modules/desktop_capture/screen_capturer_mac.mm
new file mode 100644
index 0000000000..285086ffa6
--- /dev/null
+++ b/modules/desktop_capture/screen_capturer_mac.mm
@@ -0,0 +1,766 @@
+/*
+ * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+
+#include <memory>
+#include <set>
+#include <utility>
+
+#include <ApplicationServices/ApplicationServices.h>
+#include <Cocoa/Cocoa.h>
+#include <CoreGraphics/CoreGraphics.h>
+
+#include "modules/desktop_capture/desktop_capture_options.h"
+#include "modules/desktop_capture/desktop_capturer.h"
+#include "modules/desktop_capture/desktop_frame.h"
+#include "modules/desktop_capture/desktop_geometry.h"
+#include "modules/desktop_capture/desktop_region.h"
+#include "modules/desktop_capture/mac/desktop_configuration.h"
+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
+#include "modules/desktop_capture/mac/scoped_pixel_buffer_object.h"
+#include "modules/desktop_capture/screen_capture_frame_queue.h"
+#include "modules/desktop_capture/screen_capturer_helper.h"
+#include "modules/desktop_capture/shared_desktop_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/macutils.h"
+#include "rtc_base/timeutils.h"
+
+namespace webrtc {
+
+namespace {
+
+// CGDisplayStreamRefs need to be destroyed asynchronously after receiving a
+// kCGDisplayStreamFrameStatusStopped callback from CoreGraphics. This may
+// happen after the ScreenCapturerMac has been destroyed. DisplayStreamManager
+// is responsible for destroying all extant CGDisplayStreamRefs, and will
+// destroy itself once it's done.
+class DisplayStreamManager {
+ public:
+ int GetUniqueId() { return ++unique_id_generator_; }
+ void DestroyStream(int unique_id) {
+ auto it = display_stream_wrappers_.find(unique_id);
+ RTC_CHECK(it != display_stream_wrappers_.end());
+ RTC_CHECK(!it->second.active);
+ CFRelease(it->second.stream);
+ display_stream_wrappers_.erase(it);
+
+ if (ready_for_self_destruction_ && display_stream_wrappers_.empty())
+ delete this;
+ }
+
+ void SaveStream(int unique_id,
+ CGDisplayStreamRef stream) {
+ RTC_CHECK(unique_id <= unique_id_generator_);
+ DisplayStreamWrapper wrapper;
+ wrapper.stream = stream;
+ display_stream_wrappers_[unique_id] = wrapper;
+ }
+
+ void UnregisterActiveStreams() {
+ for (auto& pair : display_stream_wrappers_) {
+ DisplayStreamWrapper& wrapper = pair.second;
+ if (wrapper.active) {
+ wrapper.active = false;
+ CFRunLoopSourceRef source =
+ CGDisplayStreamGetRunLoopSource(wrapper.stream);
+ CFRunLoopRemoveSource(CFRunLoopGetCurrent(), source,
+ kCFRunLoopCommonModes);
+ CGDisplayStreamStop(wrapper.stream);
+ }
+ }
+ }
+
+ void PrepareForSelfDestruction() {
+ ready_for_self_destruction_ = true;
+
+ if (display_stream_wrappers_.empty())
+ delete this;
+ }
+
+ // Once the DisplayStreamManager is ready for destruction, the
+ // ScreenCapturerMac is no longer present. Any updates should be ignored.
+ bool ShouldIgnoreUpdates() { return ready_for_self_destruction_; }
+
+ private:
+ struct DisplayStreamWrapper {
+ // The registered CGDisplayStreamRef.
+ CGDisplayStreamRef stream = nullptr;
+
+ // Set to false when the stream has been stopped. An asynchronous callback
+ // from CoreGraphics will let us destroy the CGDisplayStreamRef.
+ bool active = true;
+ };
+
+ std::map<int, DisplayStreamWrapper> display_stream_wrappers_;
+ int unique_id_generator_ = 0;
+ bool ready_for_self_destruction_ = false;
+};
+
+// Standard Mac displays have 72dpi, but we report 96dpi for
+// consistency with Windows and Linux.
+const int kStandardDPI = 96;
+
+// Scales all coordinates of a rect by a specified factor.
+DesktopRect ScaleAndRoundCGRect(const CGRect& rect, float scale) {
+ return DesktopRect::MakeLTRB(
+ static_cast<int>(floor(rect.origin.x * scale)),
+ static_cast<int>(floor(rect.origin.y * scale)),
+ static_cast<int>(ceil((rect.origin.x + rect.size.width) * scale)),
+ static_cast<int>(ceil((rect.origin.y + rect.size.height) * scale)));
+}
+
+// Copy pixels in the |rect| from |src_place| to |dest_plane|. |rect| should be
+// relative to the origin of |src_plane| and |dest_plane|.
+void CopyRect(const uint8_t* src_plane,
+ int src_plane_stride,
+ uint8_t* dest_plane,
+ int dest_plane_stride,
+ int bytes_per_pixel,
+ const DesktopRect& rect) {
+ // Get the address of the starting point.
+ const int src_y_offset = src_plane_stride * rect.top();
+ const int dest_y_offset = dest_plane_stride * rect.top();
+ const int x_offset = bytes_per_pixel * rect.left();
+ src_plane += src_y_offset + x_offset;
+ dest_plane += dest_y_offset + x_offset;
+
+ // Copy pixels in the rectangle line by line.
+ const int bytes_per_line = bytes_per_pixel * rect.width();
+ const int height = rect.height();
+ for (int i = 0 ; i < height; ++i) {
+ memcpy(dest_plane, src_plane, bytes_per_line);
+ src_plane += src_plane_stride;
+ dest_plane += dest_plane_stride;
+ }
+}
+
+// Returns an array of CGWindowID for all the on-screen windows except
+// |window_to_exclude|, or NULL if the window is not found or it fails. The
+// caller should release the returned CFArrayRef.
+CFArrayRef CreateWindowListWithExclusion(CGWindowID window_to_exclude) {
+ if (!window_to_exclude)
+ return nullptr;
+
+ CFArrayRef all_windows = CGWindowListCopyWindowInfo(
+ kCGWindowListOptionOnScreenOnly, kCGNullWindowID);
+ if (!all_windows)
+ return nullptr;
+
+ CFMutableArrayRef returned_array =
+ CFArrayCreateMutable(nullptr, CFArrayGetCount(all_windows), nullptr);
+
+ bool found = false;
+ for (CFIndex i = 0; i < CFArrayGetCount(all_windows); ++i) {
+ CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
+ CFArrayGetValueAtIndex(all_windows, i));
+
+ CFNumberRef id_ref = reinterpret_cast<CFNumberRef>(
+ CFDictionaryGetValue(window, kCGWindowNumber));
+
+ CGWindowID id;
+ CFNumberGetValue(id_ref, kCFNumberIntType, &id);
+ if (id == window_to_exclude) {
+ found = true;
+ continue;
+ }
+ CFArrayAppendValue(returned_array, reinterpret_cast<void *>(id));
+ }
+ CFRelease(all_windows);
+
+ if (!found) {
+ CFRelease(returned_array);
+ returned_array = nullptr;
+ }
+ return returned_array;
+}
+
+// Returns the bounds of |window| in physical pixels, enlarged by a small amount
+// on four edges to take account of the border/shadow effects.
+DesktopRect GetExcludedWindowPixelBounds(CGWindowID window,
+ float dip_to_pixel_scale) {
+ // The amount of pixels to add to the actual window bounds to take into
+ // account of the border/shadow effects.
+ static const int kBorderEffectSize = 20;
+ CGRect rect;
+ CGWindowID ids[1];
+ ids[0] = window;
+
+ CFArrayRef window_id_array =
+ CFArrayCreate(nullptr, reinterpret_cast<const void**>(&ids), 1, nullptr);
+ CFArrayRef window_array =
+ CGWindowListCreateDescriptionFromArray(window_id_array);
+
+ if (CFArrayGetCount(window_array) > 0) {
+ CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
+ CFArrayGetValueAtIndex(window_array, 0));
+ CFDictionaryRef bounds_ref = reinterpret_cast<CFDictionaryRef>(
+ CFDictionaryGetValue(window, kCGWindowBounds));
+ CGRectMakeWithDictionaryRepresentation(bounds_ref, &rect);
+ }
+
+ CFRelease(window_id_array);
+ CFRelease(window_array);
+
+ rect.origin.x -= kBorderEffectSize;
+ rect.origin.y -= kBorderEffectSize;
+ rect.size.width += kBorderEffectSize * 2;
+ rect.size.height += kBorderEffectSize * 2;
+ // |rect| is in DIP, so convert to physical pixels.
+ return ScaleAndRoundCGRect(rect, dip_to_pixel_scale);
+}
+
+// Create an image of the given region using the given |window_list|.
+// |pixel_bounds| should be in the primary display's coordinate in physical
+// pixels. The caller should release the returned CGImageRef and CFDataRef.
+CGImageRef CreateExcludedWindowRegionImage(const DesktopRect& pixel_bounds,
+ float dip_to_pixel_scale,
+ CFArrayRef window_list) {
+ CGRect window_bounds;
+ // The origin is in DIP while the size is in physical pixels. That's what
+ // CGWindowListCreateImageFromArray expects.
+ window_bounds.origin.x = pixel_bounds.left() / dip_to_pixel_scale;
+ window_bounds.origin.y = pixel_bounds.top() / dip_to_pixel_scale;
+ window_bounds.size.width = pixel_bounds.width();
+ window_bounds.size.height = pixel_bounds.height();
+
+ return CGWindowListCreateImageFromArray(
+ window_bounds, window_list, kCGWindowImageDefault);
+}
+
+// A class to perform video frame capturing for mac.
+class ScreenCapturerMac : public DesktopCapturer {
+ public:
+ explicit ScreenCapturerMac(
+ rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
+ bool detect_updated_region);
+ ~ScreenCapturerMac() override;
+
+ bool Init();
+
+ // DesktopCapturer interface.
+ void Start(Callback* callback) override;
+ void CaptureFrame() override;
+ void SetExcludedWindow(WindowId window) override;
+ bool GetSourceList(SourceList* screens) override;
+ bool SelectSource(SourceId id) override;
+
+ private:
+ // Returns false if the selected screen is no longer valid.
+ bool CgBlit(const DesktopFrame& frame, const DesktopRegion& region);
+
+ // Called when the screen configuration is changed.
+ void ScreenConfigurationChanged();
+
+ bool RegisterRefreshAndMoveHandlers();
+ void UnregisterRefreshAndMoveHandlers();
+
+ void ScreenRefresh(CGRectCount count,
+ const CGRect *rect_array,
+ DesktopVector display_origin);
+ void ReleaseBuffers();
+
+ std::unique_ptr<DesktopFrame> CreateFrame();
+
+ const bool detect_updated_region_;
+
+ Callback* callback_ = nullptr;
+
+ ScopedPixelBufferObject pixel_buffer_object_;
+
+ // Queue of the frames buffers.
+ ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
+
+ // Current display configuration.
+ MacDesktopConfiguration desktop_config_;
+
+ // Currently selected display, or 0 if the full desktop is selected. On OS X
+ // 10.6 and before, this is always 0.
+ CGDirectDisplayID current_display_ = 0;
+
+ // The physical pixel bounds of the current screen.
+ DesktopRect screen_pixel_bounds_;
+
+ // The dip to physical pixel scale of the current screen.
+ float dip_to_pixel_scale_ = 1.0f;
+
+ // A thread-safe list of invalid rectangles, and the size of the most
+ // recently captured screen.
+ ScreenCapturerHelper helper_;
+
+ // Contains an invalid region from the previous capture.
+ DesktopRegion last_invalid_region_;
+
+ // Monitoring display reconfiguration.
+ rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor_;
+
+ CGWindowID excluded_window_ = 0;
+
+ // A self-owned object that will destroy itself after ScreenCapturerMac and
+ // all display streams have been destroyed..
+ DisplayStreamManager* display_stream_manager_;
+
+ // Used to force CaptureFrame to update it's screen configuration
+ // and reregister event handlers. This ensure that this
+ // occurs on the ScreenCapture thread. Read and written from
+ // both the VideoCapture thread and ScreenCapture thread.
+ // Protected by desktop_config_monitor_.
+ bool update_screen_configuration_ = false;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerMac);
+};
+
+// DesktopFrame wrapper that flips wrapped frame upside down by inverting
+// stride.
+class InvertedDesktopFrame : public DesktopFrame {
+ public:
+ InvertedDesktopFrame(std::unique_ptr<DesktopFrame> frame)
+ : DesktopFrame(
+ frame->size(),
+ -frame->stride(),
+ frame->data() + (frame->size().height() - 1) * frame->stride(),
+ frame->shared_memory()) {
+ original_frame_ = std::move(frame);
+ MoveFrameInfoFrom(original_frame_.get());
+ }
+ ~InvertedDesktopFrame() override {}
+
+ private:
+ std::unique_ptr<DesktopFrame> original_frame_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(InvertedDesktopFrame);
+};
+
+ScreenCapturerMac::ScreenCapturerMac(
+ rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
+ bool detect_updated_region)
+ : detect_updated_region_(detect_updated_region),
+ desktop_config_monitor_(desktop_config_monitor) {
+ display_stream_manager_ = new DisplayStreamManager;
+}
+
+ScreenCapturerMac::~ScreenCapturerMac() {
+ ReleaseBuffers();
+ UnregisterRefreshAndMoveHandlers();
+ display_stream_manager_->PrepareForSelfDestruction();
+}
+
+bool ScreenCapturerMac::Init() {
+ desktop_config_monitor_->Lock();
+ desktop_config_ = desktop_config_monitor_->desktop_configuration();
+ desktop_config_monitor_->Unlock();
+ if (!RegisterRefreshAndMoveHandlers()) {
+ return false;
+ }
+ ScreenConfigurationChanged();
+ return true;
+}
+
+void ScreenCapturerMac::ReleaseBuffers() {
+ // The buffers might be in use by the encoder, so don't delete them here.
+ // Instead, mark them as "needs update"; next time the buffers are used by
+ // the capturer, they will be recreated if necessary.
+ queue_.Reset();
+}
+
+void ScreenCapturerMac::Start(Callback* callback) {
+ assert(!callback_);
+ assert(callback);
+
+ callback_ = callback;
+ desktop_config_monitor_->Lock();
+ update_screen_configuration_ = true;
+ desktop_config_monitor_->Unlock();
+}
+
+void ScreenCapturerMac::CaptureFrame() {
+ int64_t capture_start_time_nanos = rtc::TimeNanos();
+
+ // Spin RunLoop for 1/100th of a second, handling at most one source
+ CFRunLoopRunInMode(kCFRunLoopDefaultMode, 0.01, true);
+
+ queue_.MoveToNextFrame();
+ RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
+
+ desktop_config_monitor_->Lock();
+ MacDesktopConfiguration new_config =
+ desktop_config_monitor_->desktop_configuration();
+ if (update_screen_configuration_ || !desktop_config_.Equals(new_config)) {
+ update_screen_configuration_ = false;
+ desktop_config_ = new_config;
+ // If the display configuraiton has changed then refresh capturer data
+ // structures. Occasionally, the refresh and move handlers are lost when
+ // the screen mode changes, so re-register them here.
+ UnregisterRefreshAndMoveHandlers();
+ RegisterRefreshAndMoveHandlers();
+ ScreenConfigurationChanged();
+ }
+
+ DesktopRegion region;
+ helper_.TakeInvalidRegion(&region);
+
+ // If the current buffer is from an older generation then allocate a new one.
+ // Note that we can't reallocate other buffers at this point, since the caller
+ // may still be reading from them.
+ if (!queue_.current_frame())
+ queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(CreateFrame()));
+
+ DesktopFrame* current_frame = queue_.current_frame();
+
+ if (!CgBlit(*current_frame, region)) {
+ desktop_config_monitor_->Unlock();
+ callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
+ return;
+ }
+ std::unique_ptr<DesktopFrame> new_frame = queue_.current_frame()->Share();
+ if (detect_updated_region_) {
+ *new_frame->mutable_updated_region() = region;
+ } else {
+ new_frame->mutable_updated_region()->AddRect(
+ DesktopRect::MakeSize(new_frame->size()));
+ }
+
+ if (current_display_) {
+ const MacDisplayConfiguration* config =
+ desktop_config_.FindDisplayConfigurationById(current_display_);
+ if (config) {
+ new_frame->set_top_left(config->bounds.top_left().subtract(
+ desktop_config_.bounds.top_left()));
+ }
+ }
+
+ helper_.set_size_most_recent(new_frame->size());
+
+ // Signal that we are done capturing data from the display framebuffer,
+ // and accessing display structures.
+ desktop_config_monitor_->Unlock();
+
+ new_frame->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
+ rtc::kNumNanosecsPerMillisec);
+ callback_->OnCaptureResult(Result::SUCCESS, std::move(new_frame));
+}
+
+void ScreenCapturerMac::SetExcludedWindow(WindowId window) {
+ excluded_window_ = window;
+}
+
+bool ScreenCapturerMac::GetSourceList(SourceList* screens) {
+ assert(screens->size() == 0);
+
+ for (MacDisplayConfigurations::iterator it = desktop_config_.displays.begin();
+ it != desktop_config_.displays.end(); ++it) {
+ screens->push_back({it->id});
+ }
+ return true;
+}
+
+bool ScreenCapturerMac::SelectSource(SourceId id) {
+ if (id == kFullDesktopScreenId) {
+ current_display_ = 0;
+ } else {
+ const MacDisplayConfiguration* config =
+ desktop_config_.FindDisplayConfigurationById(
+ static_cast<CGDirectDisplayID>(id));
+ if (!config)
+ return false;
+ current_display_ = config->id;
+ }
+
+ ScreenConfigurationChanged();
+ return true;
+}
+
+bool ScreenCapturerMac::CgBlit(const DesktopFrame& frame, const DesktopRegion& region) {
+ // Copy the entire contents of the previous capture buffer, to capture over.
+ // TODO(wez): Get rid of this as per crbug.com/145064, or implement
+ // crbug.com/92354.
+ if (queue_.previous_frame()) {
+ memcpy(frame.data(), queue_.previous_frame()->data(),
+ frame.stride() * frame.size().height());
+ }
+
+ MacDisplayConfigurations displays_to_capture;
+ if (current_display_) {
+ // Capturing a single screen. Note that the screen id may change when
+ // screens are added or removed.
+ const MacDisplayConfiguration* config =
+ desktop_config_.FindDisplayConfigurationById(current_display_);
+ if (config) {
+ displays_to_capture.push_back(*config);
+ } else {
+ RTC_LOG(LS_ERROR) << "The selected screen cannot be found for capturing.";
+ return false;
+ }
+ } else {
+ // Capturing the whole desktop.
+ displays_to_capture = desktop_config_.displays;
+ }
+
+ // Create the window list once for all displays.
+ CFArrayRef window_list = CreateWindowListWithExclusion(excluded_window_);
+
+ for (size_t i = 0; i < displays_to_capture.size(); ++i) {
+ const MacDisplayConfiguration& display_config = displays_to_capture[i];
+
+ // Capturing mixed-DPI on one surface is hard, so we only return displays
+ // that match the "primary" display's DPI. The primary display is always
+ // the first in the list.
+ if (i > 0 && display_config.dip_to_pixel_scale !=
+ displays_to_capture[0].dip_to_pixel_scale) {
+ continue;
+ }
+ // Determine the display's position relative to the desktop, in pixels.
+ DesktopRect display_bounds = display_config.pixel_bounds;
+ display_bounds.Translate(-screen_pixel_bounds_.left(),
+ -screen_pixel_bounds_.top());
+
+ // Determine which parts of the blit region, if any, lay within the monitor.
+ DesktopRegion copy_region = region;
+ copy_region.IntersectWith(display_bounds);
+ if (copy_region.is_empty())
+ continue;
+
+ // Translate the region to be copied into display-relative coordinates.
+ copy_region.Translate(-display_bounds.left(), -display_bounds.top());
+
+ DesktopRect excluded_window_bounds;
+ CGImageRef excluded_image = nullptr;
+ if (excluded_window_ && window_list) {
+ // Get the region of the excluded window relative the primary display.
+ excluded_window_bounds = GetExcludedWindowPixelBounds(
+ excluded_window_, display_config.dip_to_pixel_scale);
+ excluded_window_bounds.IntersectWith(display_config.pixel_bounds);
+
+ // Create the image under the excluded window first, because it's faster
+ // than captuing the whole display.
+ if (!excluded_window_bounds.is_empty()) {
+ excluded_image = CreateExcludedWindowRegionImage(
+ excluded_window_bounds, display_config.dip_to_pixel_scale,
+ window_list);
+ }
+ }
+
+ // Create an image containing a snapshot of the display.
+ CGImageRef image = CGDisplayCreateImage(display_config.id);
+ if (!image) {
+ if (excluded_image)
+ CFRelease(excluded_image);
+ continue;
+ }
+
+ // Verify that the image has 32-bit depth.
+ int bits_per_pixel = CGImageGetBitsPerPixel(image);
+ if (bits_per_pixel / 8 != DesktopFrame::kBytesPerPixel) {
+ RTC_LOG(LS_ERROR) << "CGDisplayCreateImage() returned imaged with " << bits_per_pixel
+ << " bits per pixel. Only 32-bit depth is supported.";
+ CFRelease(image);
+ if (excluded_image)
+ CFRelease(excluded_image);
+ return false;
+ }
+
+ // Request access to the raw pixel data via the image's DataProvider.
+ CGDataProviderRef provider = CGImageGetDataProvider(image);
+ CFDataRef data = CGDataProviderCopyData(provider);
+ assert(data);
+
+ const uint8_t* display_base_address = CFDataGetBytePtr(data);
+ int src_bytes_per_row = CGImageGetBytesPerRow(image);
+
+ // |image| size may be different from display_bounds in case the screen was
+ // resized recently.
+ copy_region.IntersectWith(
+ DesktopRect::MakeWH(CGImageGetWidth(image), CGImageGetHeight(image)));
+
+ // Copy the dirty region from the display buffer into our desktop buffer.
+ uint8_t* out_ptr = frame.GetFrameDataAtPos(display_bounds.top_left());
+ for (DesktopRegion::Iterator i(copy_region); !i.IsAtEnd(); i.Advance()) {
+ CopyRect(display_base_address, src_bytes_per_row, out_ptr, frame.stride(),
+ DesktopFrame::kBytesPerPixel, i.rect());
+ }
+
+ CFRelease(data);
+ CFRelease(image);
+
+ if (excluded_image) {
+ CGDataProviderRef provider = CGImageGetDataProvider(excluded_image);
+ CFDataRef excluded_image_data = CGDataProviderCopyData(provider);
+ assert(excluded_image_data);
+ display_base_address = CFDataGetBytePtr(excluded_image_data);
+ src_bytes_per_row = CGImageGetBytesPerRow(excluded_image);
+
+ // Translate the bounds relative to the desktop, because |frame| data
+ // starts from the desktop top-left corner.
+ DesktopRect window_bounds_relative_to_desktop(excluded_window_bounds);
+ window_bounds_relative_to_desktop.Translate(-screen_pixel_bounds_.left(),
+ -screen_pixel_bounds_.top());
+
+ DesktopRect rect_to_copy =
+ DesktopRect::MakeSize(excluded_window_bounds.size());
+ rect_to_copy.IntersectWith(DesktopRect::MakeWH(
+ CGImageGetWidth(excluded_image), CGImageGetHeight(excluded_image)));
+
+ if (CGImageGetBitsPerPixel(excluded_image) / 8 ==
+ DesktopFrame::kBytesPerPixel) {
+ CopyRect(display_base_address, src_bytes_per_row,
+ frame.GetFrameDataAtPos(
+ window_bounds_relative_to_desktop.top_left()),
+ frame.stride(), DesktopFrame::kBytesPerPixel, rect_to_copy);
+ }
+
+ CFRelease(excluded_image_data);
+ CFRelease(excluded_image);
+ }
+ }
+ if (window_list)
+ CFRelease(window_list);
+ return true;
+}
+
+void ScreenCapturerMac::ScreenConfigurationChanged() {
+ if (current_display_) {
+ const MacDisplayConfiguration* config =
+ desktop_config_.FindDisplayConfigurationById(current_display_);
+ screen_pixel_bounds_ = config ? config->pixel_bounds : DesktopRect();
+ dip_to_pixel_scale_ = config ? config->dip_to_pixel_scale : 1.0f;
+ } else {
+ screen_pixel_bounds_ = desktop_config_.pixel_bounds;
+ dip_to_pixel_scale_ = desktop_config_.dip_to_pixel_scale;
+ }
+
+ // Release existing buffers, which will be of the wrong size.
+ ReleaseBuffers();
+
+ // Clear the dirty region, in case the display is down-sizing.
+ helper_.ClearInvalidRegion();
+
+ // Re-mark the entire desktop as dirty.
+ helper_.InvalidateScreen(screen_pixel_bounds_.size());
+
+ // Make sure the frame buffers will be reallocated.
+ queue_.Reset();
+}
+
+bool ScreenCapturerMac::RegisterRefreshAndMoveHandlers() {
+ desktop_config_ = desktop_config_monitor_->desktop_configuration();
+ for (const auto& config : desktop_config_.displays) {
+ size_t pixel_width = config.pixel_bounds.width();
+ size_t pixel_height = config.pixel_bounds.height();
+ if (pixel_width == 0 || pixel_height == 0)
+ continue;
+ // Using a local variable forces the block to capture the raw pointer.
+ DisplayStreamManager* manager = display_stream_manager_;
+ int unique_id = manager->GetUniqueId();
+ CGDirectDisplayID display_id = config.id;
+ DesktopVector display_origin = config.pixel_bounds.top_left();
+
+ CGDisplayStreamFrameAvailableHandler handler =
+ ^(CGDisplayStreamFrameStatus status, uint64_t display_time,
+ IOSurfaceRef frame_surface, CGDisplayStreamUpdateRef updateRef) {
+ if (status == kCGDisplayStreamFrameStatusStopped) {
+ manager->DestroyStream(unique_id);
+ return;
+ }
+
+ if (manager->ShouldIgnoreUpdates())
+ return;
+
+ // Only pay attention to frame updates.
+ if (status != kCGDisplayStreamFrameStatusFrameComplete)
+ return;
+
+ size_t count = 0;
+ const CGRect* rects = CGDisplayStreamUpdateGetRects(
+ updateRef, kCGDisplayStreamUpdateDirtyRects, &count);
+ if (count != 0) {
+ // According to CGDisplayStream.h, it's safe to call
+ // CGDisplayStreamStop() from within the callback.
+ ScreenRefresh(count, rects, display_origin);
+ }
+ };
+ CGDisplayStreamRef display_stream = CGDisplayStreamCreate(
+ display_id, pixel_width, pixel_height, 'BGRA', nullptr, handler);
+
+ if (display_stream) {
+ CGError error = CGDisplayStreamStart(display_stream);
+ if (error != kCGErrorSuccess)
+ return false;
+
+ CFRunLoopSourceRef source =
+ CGDisplayStreamGetRunLoopSource(display_stream);
+ CFRunLoopAddSource(CFRunLoopGetCurrent(), source, kCFRunLoopCommonModes);
+ display_stream_manager_->SaveStream(unique_id, display_stream);
+ }
+ }
+
+ return true;
+}
+
+void ScreenCapturerMac::UnregisterRefreshAndMoveHandlers() {
+ display_stream_manager_->UnregisterActiveStreams();
+}
+
+void ScreenCapturerMac::ScreenRefresh(CGRectCount count,
+ const CGRect* rect_array,
+ DesktopVector display_origin) {
+ if (screen_pixel_bounds_.is_empty())
+ ScreenConfigurationChanged();
+
+ // The refresh rects are in display coordinates. We want to translate to
+ // framebuffer coordinates. If a specific display is being captured, then no
+ // change is necessary. If all displays are being captured, then we want to
+ // translate by the origin of the display.
+ DesktopVector translate_vector;
+ if (!current_display_)
+ translate_vector = display_origin;
+
+ DesktopRegion region;
+ for (CGRectCount i = 0; i < count; ++i) {
+ // All rects are already in physical pixel coordinates.
+ DesktopRect rect = DesktopRect::MakeXYWH(
+ rect_array[i].origin.x, rect_array[i].origin.y,
+ rect_array[i].size.width, rect_array[i].size.height);
+
+ rect.Translate(translate_vector);
+
+ region.AddRect(rect);
+ }
+
+ helper_.InvalidateRegion(region);
+}
+
+std::unique_ptr<DesktopFrame> ScreenCapturerMac::CreateFrame() {
+ std::unique_ptr<DesktopFrame> frame(
+ new BasicDesktopFrame(screen_pixel_bounds_.size()));
+ frame->set_dpi(DesktopVector(kStandardDPI * dip_to_pixel_scale_,
+ kStandardDPI * dip_to_pixel_scale_));
+ return frame;
+}
+
+} // namespace
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
+ const DesktopCaptureOptions& options) {
+ if (!options.configuration_monitor())
+ return nullptr;
+
+ std::unique_ptr<ScreenCapturerMac> capturer(new ScreenCapturerMac(
+ options.configuration_monitor(), options.detect_updated_region()));
+ if (!capturer.get()->Init()) {
+ return nullptr;
+ }
+
+ return capturer;
+}
+
+} // namespace webrtc
return true;
}
diff --git a/modules/desktop_capture/mac/window_list_utils.cc b/modules/desktop_capture/mac/window_list_utils.cc
index 5d881662ea..989ec7ea54 100644
--- a/modules/desktop_capture/mac/window_list_utils.cc
+++ b/modules/desktop_capture/mac/window_list_utils.cc
@@ -198,7 +198,7 @@ bool GetWindowList(DesktopCapturer::SourceList* windows,
// the check in the map. Also skip the window if owner name is
// empty too.
if (!owner_name.empty() && (itr == pid_itr_map.end())) {
- sources.push_back(DesktopCapturer::Source{window_id, owner_name});
+ sources.push_back(DesktopCapturer::Source{window_id, pid, owner_name});
RTC_DCHECK(!sources.empty());
// Get an iterator on the last valid element in the source list.
std::list<DesktopCapturer::Source>::const_iterator last_source =
@@ -209,7 +209,7 @@ bool GetWindowList(DesktopCapturer::SourceList* windows,
pid, last_source));
}
} else {
- sources.push_back(DesktopCapturer::Source{window_id, title});
+ sources.push_back(DesktopCapturer::Source{window_id, pid, title});
// Once the window with empty title has been removed no other empty
// windows are allowed for the same pid.
if (itr != pid_itr_map.end() && (itr->second != sources.end())) {
--
2.34.1

68
third_party/libwebrtc/moz-patch-stack/0012.patch поставляемый
Просмотреть файл

@ -1,56 +1,26 @@
From: "Byron Campen [:bwc]" <docfaraday@gmail.com>
Date: Fri, 12 Feb 2021 14:27:00 +0000
Subject: Bug 1654112 - Get OS X build working.
r=ng,firefox-build-system-reviewers,glandium
From: Dan Minor <dminor@mozilla.com>
Date: Mon, 13 Aug 2018 08:34:00 -0400
Subject: Bug 1376873 - Allow single channel opus; r=padenot
* Pull in sdk/objc/base and sdk/objc/helpers
* Add gclient_args.gni to keep build happy.
* Add a missing include path for libyuv
* Support .m files in build.
Differential Revision: https://phabricator.services.mozilla.com/D105015
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9314046d89ebc0836a50ab7addece71abbf0b5e4
Differential Revision: https://phabricator.services.mozilla.com/D7469
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/92a7c3eee9f0c80ffbf79fdab8f8f2b8f6bd7701
---
modules/desktop_capture/mac/screen_capturer_mac.mm | 3 ++-
modules/desktop_capture/mac/window_list_utils.cc | 4 ++--
2 files changed, 4 insertions(+), 3 deletions(-)
modules/audio_coding/codecs/opus/audio_encoder_opus.cc | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/modules/desktop_capture/mac/screen_capturer_mac.mm b/modules/desktop_capture/mac/screen_capturer_mac.mm
index 115f6440b1..cad0c5b65b 100644
--- a/modules/desktop_capture/mac/screen_capturer_mac.mm
+++ b/modules/desktop_capture/mac/screen_capturer_mac.mm
@@ -276,7 +276,8 @@ bool ScreenCapturerMac::GetSourceList(SourceList* screens) {
for (MacDisplayConfigurations::iterator it = desktop_config_.displays.begin();
it != desktop_config_.displays.end();
++it) {
- screens->push_back({it->id, std::string()});
+ Source value = {it->id, 0, std::string()};
+ screens->push_back(value);
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 51b0fcd492..17e0e33b1d 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -240,7 +240,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "opus") ||
- format.clockrate_hz != kRtpTimestampRateHz || format.num_channels != 2) {
+ format.clockrate_hz != kRtpTimestampRateHz) {
return absl::nullopt;
}
return true;
}
diff --git a/modules/desktop_capture/mac/window_list_utils.cc b/modules/desktop_capture/mac/window_list_utils.cc
index 5d881662ea..989ec7ea54 100644
--- a/modules/desktop_capture/mac/window_list_utils.cc
+++ b/modules/desktop_capture/mac/window_list_utils.cc
@@ -198,7 +198,7 @@ bool GetWindowList(DesktopCapturer::SourceList* windows,
// the check in the map. Also skip the window if owner name is
// empty too.
if (!owner_name.empty() && (itr == pid_itr_map.end())) {
- sources.push_back(DesktopCapturer::Source{window_id, owner_name});
+ sources.push_back(DesktopCapturer::Source{window_id, pid, owner_name});
RTC_DCHECK(!sources.empty());
// Get an iterator on the last valid element in the source list.
std::list<DesktopCapturer::Source>::const_iterator last_source =
@@ -209,7 +209,7 @@ bool GetWindowList(DesktopCapturer::SourceList* windows,
pid, last_source));
}
} else {
- sources.push_back(DesktopCapturer::Source{window_id, title});
+ sources.push_back(DesktopCapturer::Source{window_id, pid, title});
// Once the window with empty title has been removed no other empty
// windows are allowed for the same pid.
if (itr != pid_itr_map.end() && (itr->second != sources.end())) {
--
2.34.1

47
third_party/libwebrtc/moz-patch-stack/0013.patch поставляемый
Просмотреть файл

@ -1,26 +1,37 @@
From: Dan Minor <dminor@mozilla.com>
Date: Mon, 13 Aug 2018 08:34:00 -0400
Subject: Bug 1376873 - Allow single channel opus; r=padenot
Date: Mon, 13 Aug 2018 10:24:00 -0400
Subject: Bug 1376873 - Fix warning in mean_variance_estimator.cc; r=padenot
Differential Revision: https://phabricator.services.mozilla.com/D7469
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/92a7c3eee9f0c80ffbf79fdab8f8f2b8f6bd7701
Differential Revision: https://phabricator.services.mozilla.com/D7470
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cb6c555654c0bca72999f1e414d8d6d0b59186c9
---
modules/audio_coding/codecs/opus/audio_encoder_opus.cc | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
.../echo_detector/mean_variance_estimator.cc | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 51b0fcd492..17e0e33b1d 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -240,7 +240,7 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
const SdpAudioFormat& format) {
if (!absl::EqualsIgnoreCase(format.name, "opus") ||
- format.clockrate_hz != kRtpTimestampRateHz || format.num_channels != 2) {
+ format.clockrate_hz != kRtpTimestampRateHz) {
return absl::nullopt;
}
diff --git a/modules/audio_processing/echo_detector/mean_variance_estimator.cc b/modules/audio_processing/echo_detector/mean_variance_estimator.cc
index a85740387b..a9ebb8cd92 100644
--- a/modules/audio_processing/echo_detector/mean_variance_estimator.cc
+++ b/modules/audio_processing/echo_detector/mean_variance_estimator.cc
@@ -10,7 +10,7 @@
#include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
-#include <math.h>
+#include <cmath>
#include "rtc_base/checks.h"
@@ -26,8 +26,8 @@ void MeanVarianceEstimator::Update(float value) {
mean_ = (1.f - kAlpha) * mean_ + kAlpha * value;
variance_ =
(1.f - kAlpha) * variance_ + kAlpha * (value - mean_) * (value - mean_);
- RTC_DCHECK(isfinite(mean_));
- RTC_DCHECK(isfinite(variance_));
+ RTC_DCHECK(std::isfinite(mean_));
+ RTC_DCHECK(std::isfinite(variance_));
}
float MeanVarianceEstimator::std_deviation() const {
--
2.34.1

66
third_party/libwebrtc/moz-patch-stack/0014.patch поставляемый
Просмотреть файл

@ -1,37 +1,45 @@
From: Dan Minor <dminor@mozilla.com>
Date: Mon, 13 Aug 2018 10:24:00 -0400
Subject: Bug 1376873 - Fix warning in mean_variance_estimator.cc; r=padenot
Date: Wed, 19 Sep 2018 15:06:00 -0400
Subject: Bug 1376873 - Fix up rotation in video_capture_impl.cc; r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D7470
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cb6c555654c0bca72999f1e414d8d6d0b59186c9
This fixes a bug in the upstream code introduced when they removed the
ConvertToI420 helper method from webrtc_libyuv.cc. The buffer size is
passed into libyuv::ConvertI420 incorrectly when rotation is applied, which
causes bad rendering and instabilities.
Differential Revision: https://phabricator.services.mozilla.com/D7478
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c56cb66f86518dfb046c642ebeb07cf51c23a3cb
---
.../echo_detector/mean_variance_estimator.cc | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
modules/video_capture/video_capture_impl.cc | 11 ++++++++++-
1 file changed, 10 insertions(+), 1 deletion(-)
diff --git a/modules/audio_processing/echo_detector/mean_variance_estimator.cc b/modules/audio_processing/echo_detector/mean_variance_estimator.cc
index a85740387b..a9ebb8cd92 100644
--- a/modules/audio_processing/echo_detector/mean_variance_estimator.cc
+++ b/modules/audio_processing/echo_detector/mean_variance_estimator.cc
@@ -10,7 +10,7 @@
diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc
index d5ec4daae1..5dbc5e4e27 100644
--- a/modules/video_capture/video_capture_impl.cc
+++ b/modules/video_capture/video_capture_impl.cc
@@ -208,12 +208,21 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
}
}
#include "modules/audio_processing/echo_detector/mean_variance_estimator.h"
-#include <math.h>
+#include <cmath>
#include "rtc_base/checks.h"
@@ -26,8 +26,8 @@ void MeanVarianceEstimator::Update(float value) {
mean_ = (1.f - kAlpha) * mean_ + kAlpha * value;
variance_ =
(1.f - kAlpha) * variance_ + kAlpha * (value - mean_) * (value - mean_);
- RTC_DCHECK(isfinite(mean_));
- RTC_DCHECK(isfinite(variance_));
+ RTC_DCHECK(std::isfinite(mean_));
+ RTC_DCHECK(std::isfinite(variance_));
}
float MeanVarianceEstimator::std_deviation() const {
+ int dst_width = buffer->width();
+ int dst_height = buffer->height();
+
+ // LibYuv expects pre-rotation_mode values for dst.
+ // Stride values should correspond to the destination values.
+ if (rotation_mode == libyuv::kRotate90 || rotation_mode == libyuv::kRotate270) {
+ std::swap(dst_width, dst_height);
+ }
+
const int conversionResult = libyuv::ConvertToI420(
videoFrame, videoFrameLength, buffer.get()->MutableDataY(),
buffer.get()->StrideY(), buffer.get()->MutableDataU(),
buffer.get()->StrideU(), buffer.get()->MutableDataV(),
buffer.get()->StrideV(), 0, 0, // No Cropping
- width, height, target_width, target_height, rotation_mode,
+ width, height, dst_width, dst_height, rotation_mode,
ConvertVideoType(frameInfo.videoType));
if (conversionResult != 0) {
RTC_LOG(LS_ERROR) << "Failed to convert capture frame from type "
--
2.34.1

182
third_party/libwebrtc/moz-patch-stack/0015.patch поставляемый
Просмотреть файл

@ -1,45 +1,153 @@
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 19 Sep 2018 15:06:00 -0400
Subject: Bug 1376873 - Fix up rotation in video_capture_impl.cc; r=pehrsons
From: Jan Beich <jbeich@FreeBSD.org>
Date: Fri, 12 Oct 2018 12:56:00 -0400
Subject: Bug 1376873 - Unbreak WebRTC 64 build on BSDs. r=dminor f=gaston
This fixes a bug in the upstream code introduced when they removed the
ConvertToI420 helper method from webrtc_libyuv.cc. The buffer size is
passed into libyuv::ConvertI420 incorrectly when rotation is applied, which
causes bad rendering and instabilities.
Also includes:
Bug 1554949 - Fix WebRTC build failure with newer linux kernel. r=dminor
Differential Revision: https://phabricator.services.mozilla.com/D7478
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c56cb66f86518dfb046c642ebeb07cf51c23a3cb
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a636ecdcb91afb1c5d436bbcbd87da4f10f7df78
---
modules/video_capture/video_capture_impl.cc | 11 ++++++++++-
1 file changed, 10 insertions(+), 1 deletion(-)
modules/video_capture/linux/device_info_linux.cc | 6 ++++++
modules/video_capture/linux/device_info_v4l2.cc | 8 +++++++-
modules/video_capture/linux/video_capture_linux.cc | 9 ++++++++-
rtc_base/byte_order.h | 2 ++
rtc_base/physical_socket_server.cc | 7 +++++--
system_wrappers/source/cpu_info.cc | 4 ++--
6 files changed, 30 insertions(+), 6 deletions(-)
diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc
index d5ec4daae1..5dbc5e4e27 100644
--- a/modules/video_capture/video_capture_impl.cc
+++ b/modules/video_capture/video_capture_impl.cc
@@ -208,12 +208,21 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
}
}
diff --git a/modules/video_capture/linux/device_info_linux.cc b/modules/video_capture/linux/device_info_linux.cc
index ccbbeae3ab..9c12b2183e 100644
--- a/modules/video_capture/linux/device_info_linux.cc
+++ b/modules/video_capture/linux/device_info_linux.cc
@@ -16,7 +16,13 @@
#include <sys/ioctl.h>
#include <unistd.h>
// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
#include <linux/videodev2.h>
+#endif
+ int dst_width = buffer->width();
+ int dst_height = buffer->height();
+
+ // LibYuv expects pre-rotation_mode values for dst.
+ // Stride values should correspond to the destination values.
+ if (rotation_mode == libyuv::kRotate90 || rotation_mode == libyuv::kRotate270) {
+ std::swap(dst_width, dst_height);
+ }
+
const int conversionResult = libyuv::ConvertToI420(
videoFrame, videoFrameLength, buffer.get()->MutableDataY(),
buffer.get()->StrideY(), buffer.get()->MutableDataU(),
buffer.get()->StrideU(), buffer.get()->MutableDataV(),
buffer.get()->StrideV(), 0, 0, // No Cropping
- width, height, target_width, target_height, rotation_mode,
+ width, height, dst_width, dst_height, rotation_mode,
ConvertVideoType(frameInfo.videoType));
if (conversionResult != 0) {
RTC_LOG(LS_ERROR) << "Failed to convert capture frame from type "
#include <vector>
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index 28395a5a05..d836747b4a 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -18,7 +18,13 @@
#include <sys/ioctl.h>
#include <unistd.h>
// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
#include <linux/videodev2.h>
+#endif
#include <vector>
@@ -191,8 +197,8 @@ DeviceInfoV4l2::DeviceInfoV4l2() : DeviceInfoImpl()
{
_inotifyEventThread->Start();
}
-}
#endif
+}
int32_t DeviceInfoV4l2::Init() {
return 0;
diff --git a/modules/video_capture/linux/video_capture_linux.cc b/modules/video_capture/linux/video_capture_linux.cc
index b6c4017927..4895a1ab71 100644
--- a/modules/video_capture/linux/video_capture_linux.cc
+++ b/modules/video_capture/linux/video_capture_linux.cc
@@ -10,7 +10,6 @@
#include <errno.h>
#include <fcntl.h>
-#include <linux/videodev2.h>
#include <stdio.h>
#include <string.h>
#include <sys/ioctl.h>
@@ -18,6 +17,14 @@
#include <sys/select.h>
#include <time.h>
#include <unistd.h>
+// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
#include <new>
#include <string>
diff --git a/rtc_base/byte_order.h b/rtc_base/byte_order.h
index b8f8ae9f7a..382511daeb 100644
--- a/rtc_base/byte_order.h
+++ b/rtc_base/byte_order.h
@@ -90,6 +90,8 @@
#error WEBRTC_ARCH_BIG_ENDIAN or WEBRTC_ARCH_LITTLE_ENDIAN must be defined.
#endif // defined(WEBRTC_ARCH_LITTLE_ENDIAN)
+#elif defined(WEBRTC_BSD) && !defined(__OpenBSD__)
+#include <sys/endian.h>
#elif defined(WEBRTC_POSIX)
#include <endian.h>
#else
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index b7d69140e0..4ed4fd0cbb 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -73,7 +73,10 @@ typedef void* SockOptArg;
#endif // WEBRTC_POSIX
-#if defined(WEBRTC_POSIX) && !defined(WEBRTC_MAC) && !defined(__native_client__)
+#if defined(WEBRTC_POSIX) && !defined(WEBRTC_MAC) && !defined(WEBRTC_BSD) && !defined(__native_client__)
+#if defined(WEBRTC_LINUX)
+#include <linux/sockios.h>
+#endif
int64_t GetSocketRecvTimestamp(int socket) {
struct timeval tv_ioctl;
@@ -641,7 +644,7 @@ int PhysicalSocket::TranslateOption(Option opt, int* slevel, int* sopt) {
*slevel = IPPROTO_IP;
*sopt = IP_DONTFRAGMENT;
break;
-#elif defined(WEBRTC_MAC) || defined(BSD) || defined(__native_client__)
+#elif defined(WEBRTC_MAC) || defined(WEBRTC_BSD) || defined(__native_client__)
RTC_LOG(LS_WARNING) << "Socket::OPT_DONTFRAGMENT not supported.";
return -1;
#elif defined(WEBRTC_POSIX)
diff --git a/system_wrappers/source/cpu_info.cc b/system_wrappers/source/cpu_info.cc
index eff720371a..94aed09c48 100644
--- a/system_wrappers/source/cpu_info.cc
+++ b/system_wrappers/source/cpu_info.cc
@@ -12,7 +12,7 @@
#if defined(WEBRTC_WIN)
#include <windows.h>
-#elif defined(WEBRTC_LINUX)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
#include <unistd.h>
#elif defined(WEBRTC_MAC)
#include <sys/sysctl.h>
@@ -30,7 +30,7 @@ static int DetectNumberOfCores() {
SYSTEM_INFO si;
GetNativeSystemInfo(&si);
number_of_cores = static_cast<int>(si.dwNumberOfProcessors);
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID) || defined(WEBRTC_BSD)
number_of_cores = static_cast<int>(sysconf(_SC_NPROCESSORS_ONLN));
if (number_of_cores <= 0) {
RTC_LOG(LS_ERROR) << "Failed to get number of cores";
--
2.34.1

163
third_party/libwebrtc/moz-patch-stack/0016.patch поставляемый
Просмотреть файл

@ -1,153 +1,32 @@
From: Jan Beich <jbeich@FreeBSD.org>
Date: Fri, 12 Oct 2018 12:56:00 -0400
Subject: Bug 1376873 - Unbreak WebRTC 64 build on BSDs. r=dminor f=gaston
From: Jed Davis <jld@mozilla.com>
Date: Sun, 17 Nov 2019 03:40:00 +0000
Subject: Bug 1545504 - Strengthen bounds check in WebRTC PhysicalSocketServer.
r=jesup
Also includes:
Bug 1554949 - Fix WebRTC build failure with newer linux kernel. r=dminor
PhysicalSocketServer isn't currently used by Mozilla's WebRTC
integration, but just in case, let's make sure that this array index is
bounds-checked in actual use, not just in debug builds (which tend to
never see realistic test conditions).
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a636ecdcb91afb1c5d436bbcbd87da4f10f7df78
Differential Revision: https://phabricator.services.mozilla.com/D52745
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/2b079067648bfe0d064a334bf19bdcc233e26b6e
---
modules/video_capture/linux/device_info_linux.cc | 6 ++++++
modules/video_capture/linux/device_info_v4l2.cc | 8 +++++++-
modules/video_capture/linux/video_capture_linux.cc | 9 ++++++++-
rtc_base/byte_order.h | 2 ++
rtc_base/physical_socket_server.cc | 7 +++++--
system_wrappers/source/cpu_info.cc | 4 ++--
6 files changed, 30 insertions(+), 6 deletions(-)
rtc_base/physical_socket_server.cc | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/modules/video_capture/linux/device_info_linux.cc b/modules/video_capture/linux/device_info_linux.cc
index ccbbeae3ab..9c12b2183e 100644
--- a/modules/video_capture/linux/device_info_linux.cc
+++ b/modules/video_capture/linux/device_info_linux.cc
@@ -16,7 +16,13 @@
#include <sys/ioctl.h>
#include <unistd.h>
// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
#include <linux/videodev2.h>
+#endif
#include <vector>
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index 28395a5a05..d836747b4a 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -18,7 +18,13 @@
#include <sys/ioctl.h>
#include <unistd.h>
// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
#include <linux/videodev2.h>
+#endif
#include <vector>
@@ -191,8 +197,8 @@ DeviceInfoV4l2::DeviceInfoV4l2() : DeviceInfoImpl()
{
_inotifyEventThread->Start();
}
-}
#endif
+}
int32_t DeviceInfoV4l2::Init() {
return 0;
diff --git a/modules/video_capture/linux/video_capture_linux.cc b/modules/video_capture/linux/video_capture_linux.cc
index b6c4017927..4895a1ab71 100644
--- a/modules/video_capture/linux/video_capture_linux.cc
+++ b/modules/video_capture/linux/video_capture_linux.cc
@@ -10,7 +10,6 @@
#include <errno.h>
#include <fcntl.h>
-#include <linux/videodev2.h>
#include <stdio.h>
#include <string.h>
#include <sys/ioctl.h>
@@ -18,6 +17,14 @@
#include <sys/select.h>
#include <time.h>
#include <unistd.h>
+// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
#include <new>
#include <string>
diff --git a/rtc_base/byte_order.h b/rtc_base/byte_order.h
index b8f8ae9f7a..382511daeb 100644
--- a/rtc_base/byte_order.h
+++ b/rtc_base/byte_order.h
@@ -90,6 +90,8 @@
#error WEBRTC_ARCH_BIG_ENDIAN or WEBRTC_ARCH_LITTLE_ENDIAN must be defined.
#endif // defined(WEBRTC_ARCH_LITTLE_ENDIAN)
+#elif defined(WEBRTC_BSD) && !defined(__OpenBSD__)
+#include <sys/endian.h>
#elif defined(WEBRTC_POSIX)
#include <endian.h>
#else
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index b7d69140e0..4ed4fd0cbb 100644
index 4ed4fd0cbb..60d024c769 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -73,7 +73,10 @@ typedef void* SockOptArg;
@@ -1397,7 +1397,7 @@ bool PhysicalSocketServer::WaitSelect(int cmsWait, bool process_io) {
int fd = pdispatcher->GetDescriptor();
// "select"ing a file descriptor that is equal to or larger than
// FD_SETSIZE will result in undefined behavior.
- RTC_DCHECK_LT(fd, FD_SETSIZE);
+ RTC_CHECK_LT(fd, FD_SETSIZE);
if (fd > fdmax)
fdmax = fd;
#endif // WEBRTC_POSIX
-#if defined(WEBRTC_POSIX) && !defined(WEBRTC_MAC) && !defined(__native_client__)
+#if defined(WEBRTC_POSIX) && !defined(WEBRTC_MAC) && !defined(WEBRTC_BSD) && !defined(__native_client__)
+#if defined(WEBRTC_LINUX)
+#include <linux/sockios.h>
+#endif
int64_t GetSocketRecvTimestamp(int socket) {
struct timeval tv_ioctl;
@@ -641,7 +644,7 @@ int PhysicalSocket::TranslateOption(Option opt, int* slevel, int* sopt) {
*slevel = IPPROTO_IP;
*sopt = IP_DONTFRAGMENT;
break;
-#elif defined(WEBRTC_MAC) || defined(BSD) || defined(__native_client__)
+#elif defined(WEBRTC_MAC) || defined(WEBRTC_BSD) || defined(__native_client__)
RTC_LOG(LS_WARNING) << "Socket::OPT_DONTFRAGMENT not supported.";
return -1;
#elif defined(WEBRTC_POSIX)
diff --git a/system_wrappers/source/cpu_info.cc b/system_wrappers/source/cpu_info.cc
index eff720371a..94aed09c48 100644
--- a/system_wrappers/source/cpu_info.cc
+++ b/system_wrappers/source/cpu_info.cc
@@ -12,7 +12,7 @@
#if defined(WEBRTC_WIN)
#include <windows.h>
-#elif defined(WEBRTC_LINUX)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
#include <unistd.h>
#elif defined(WEBRTC_MAC)
#include <sys/sysctl.h>
@@ -30,7 +30,7 @@ static int DetectNumberOfCores() {
SYSTEM_INFO si;
GetNativeSystemInfo(&si);
number_of_cores = static_cast<int>(si.dwNumberOfProcessors);
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID) || defined(WEBRTC_BSD)
number_of_cores = static_cast<int>(sysconf(_SC_NPROCESSORS_ONLN));
if (number_of_cores <= 0) {
RTC_LOG(LS_ERROR) << "Failed to get number of cores";
--
2.34.1

67
third_party/libwebrtc/moz-patch-stack/0017.patch поставляемый
Просмотреть файл

@ -1,32 +1,49 @@
From: Jed Davis <jld@mozilla.com>
Date: Sun, 17 Nov 2019 03:40:00 +0000
Subject: Bug 1545504 - Strengthen bounds check in WebRTC PhysicalSocketServer.
r=jesup
From: Dan Minor <dminor@mozilla.com>
Date: Mon, 5 Nov 2018 10:33:00 -0500
Subject: Bug 1376873 - Reduce thread stack size in platform_thread.cc; r=bwc
PhysicalSocketServer isn't currently used by Mozilla's WebRTC
integration, but just in case, let's make sure that this array index is
bounds-checked in actual use, not just in debug builds (which tend to
never see realistic test conditions).
Summary:
The current default stack size of 1M results in intermittent OOMs on win32
builds while running web-platform tests. The value of 256k was chosen for
consistency with the default value used elsewhere in Gecko, which is defined in
nsIThreadManager.idl.
Differential Revision: https://phabricator.services.mozilla.com/D52745
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/2b079067648bfe0d064a334bf19bdcc233e26b6e
Reviewers: bwc
Tags: #secure-revision
Bug #: 1376873
Differential Revision: https://phabricator.services.mozilla.com/D11090
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e83c311e5293902be4db4ecea17cff87c633f7cf
---
rtc_base/physical_socket_server.cc | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
rtc_base/platform_thread.cc | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index 4ed4fd0cbb..60d024c769 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -1397,7 +1397,7 @@ bool PhysicalSocketServer::WaitSelect(int cmsWait, bool process_io) {
int fd = pdispatcher->GetDescriptor();
// "select"ing a file descriptor that is equal to or larger than
// FD_SETSIZE will result in undefined behavior.
- RTC_DCHECK_LT(fd, FD_SETSIZE);
+ RTC_CHECK_LT(fd, FD_SETSIZE);
if (fd > fdmax)
fdmax = fd;
diff --git a/rtc_base/platform_thread.cc b/rtc_base/platform_thread.cc
index 6d369d747e..556204ac89 100644
--- a/rtc_base/platform_thread.cc
+++ b/rtc_base/platform_thread.cc
@@ -189,15 +189,17 @@ PlatformThread PlatformThread::SpawnThread(
// Set the reserved stack stack size to 1M, which is the default on Windows
// and Linux.
DWORD thread_id = 0;
+ // Mozilla: Set to 256kb for consistency with nsIThreadManager.idl
PlatformThread::Handle handle = ::CreateThread(
- nullptr, 1024 * 1024, &RunPlatformThread, start_thread_function_ptr,
+ nullptr, 256 * 1024, &RunPlatformThread, start_thread_function_ptr,
STACK_SIZE_PARAM_IS_A_RESERVATION, &thread_id);
RTC_CHECK(handle) << "CreateThread failed";
#else
pthread_attr_t attr;
pthread_attr_init(&attr);
// Set the stack stack size to 1M.
- pthread_attr_setstacksize(&attr, 1024 * 1024);
+ // Mozilla: Set to 256kb for consistency with nsIThreadManager.idl
+ pthread_attr_setstacksize(&attr, 256 * 1024);
pthread_attr_setdetachstate(
&attr, joinable ? PTHREAD_CREATE_JOINABLE : PTHREAD_CREATE_DETACHED);
PlatformThread::Handle handle;
--
2.34.1

72
third_party/libwebrtc/moz-patch-stack/0018.patch поставляемый
Просмотреть файл

@ -1,49 +1,37 @@
From: Dan Minor <dminor@mozilla.com>
Date: Mon, 5 Nov 2018 10:33:00 -0500
Subject: Bug 1376873 - Reduce thread stack size in platform_thread.cc; r=bwc
Date: Thu, 31 Jan 2019 15:37:00 -0500
Subject: Bug 1524208 - Calculate stride based upon target_width in
video_capture_impl.cc; r=pehrsons
Summary:
The current default stack size of 1M results in intermittent OOMs on win32
builds while running web-platform tests. The value of 256k was chosen for
consistency with the default value used elsewhere in Gecko, which is defined in
nsIThreadManager.idl.
Reviewers: bwc
Tags: #secure-revision
Bug #: 1376873
Differential Revision: https://phabricator.services.mozilla.com/D11090
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e83c311e5293902be4db4ecea17cff87c633f7cf
Differential Revision: https://phabricator.services.mozilla.com/D18270
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/51d12094d825c4c4467cb132d03d4f3cad4b1b82
---
rtc_base/platform_thread.cc | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
modules/video_capture/video_capture_impl.cc | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
diff --git a/rtc_base/platform_thread.cc b/rtc_base/platform_thread.cc
index 6d369d747e..556204ac89 100644
--- a/rtc_base/platform_thread.cc
+++ b/rtc_base/platform_thread.cc
@@ -189,15 +189,17 @@ PlatformThread PlatformThread::SpawnThread(
// Set the reserved stack stack size to 1M, which is the default on Windows
// and Linux.
DWORD thread_id = 0;
+ // Mozilla: Set to 256kb for consistency with nsIThreadManager.idl
PlatformThread::Handle handle = ::CreateThread(
- nullptr, 1024 * 1024, &RunPlatformThread, start_thread_function_ptr,
+ nullptr, 256 * 1024, &RunPlatformThread, start_thread_function_ptr,
STACK_SIZE_PARAM_IS_A_RESERVATION, &thread_id);
RTC_CHECK(handle) << "CreateThread failed";
#else
pthread_attr_t attr;
pthread_attr_init(&attr);
// Set the stack stack size to 1M.
- pthread_attr_setstacksize(&attr, 1024 * 1024);
+ // Mozilla: Set to 256kb for consistency with nsIThreadManager.idl
+ pthread_attr_setstacksize(&attr, 256 * 1024);
pthread_attr_setdetachstate(
&attr, joinable ? PTHREAD_CREATE_JOINABLE : PTHREAD_CREATE_DETACHED);
PlatformThread::Handle handle;
diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc
index 5dbc5e4e27..d227d41c34 100644
--- a/modules/video_capture/video_capture_impl.cc
+++ b/modules/video_capture/video_capture_impl.cc
@@ -167,8 +167,6 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
return -1;
}
- int stride_y = width;
- int stride_uv = (width + 1) / 2;
int target_width = width;
int target_height = abs(height);
@@ -184,6 +182,9 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
}
}
+ int stride_y = target_width;
+ int stride_uv = (target_width + 1) / 2;
+
// Setting absolute height (in case it was negative).
// In Windows, the image starts bottom left, instead of top left.
// Setting a negative source height, inverts the image (within LibYuv).
--
2.34.1

98
third_party/libwebrtc/moz-patch-stack/0019.patch поставляемый
Просмотреть файл

@ -1,37 +1,81 @@
From: Dan Minor <dminor@mozilla.com>
Date: Thu, 31 Jan 2019 15:37:00 -0500
Subject: Bug 1524208 - Calculate stride based upon target_width in
video_capture_impl.cc; r=pehrsons
Date: Thu, 21 Mar 2019 15:48:00 +0000
Subject: Bug 1535584 - Restore UpdateCodecFrameSize to vp9_impl.cc; r=bwc
Differential Revision: https://phabricator.services.mozilla.com/D18270
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/51d12094d825c4c4467cb132d03d4f3cad4b1b82
Differential Revision: https://phabricator.services.mozilla.com/D23713
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/0898f9cfe09273e1d86c38abdd576cdf273009f0
---
modules/video_capture/video_capture_impl.cc | 5 +++--
1 file changed, 3 insertions(+), 2 deletions(-)
.../codecs/vp9/libvpx_vp9_encoder.cc | 50 +++++++++++++++++++
1 file changed, 50 insertions(+)
diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc
index 5dbc5e4e27..d227d41c34 100644
--- a/modules/video_capture/video_capture_impl.cc
+++ b/modules/video_capture/video_capture_impl.cc
@@ -167,8 +167,6 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
return -1;
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
index 35b13058a2..5877373b76 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
@@ -1159,6 +1159,14 @@ int LibvpxVp9Encoder::Encode(const VideoFrame& input_image,
config_changed_ = false;
}
- int stride_y = width;
- int stride_uv = (width + 1) / 2;
int target_width = width;
int target_height = abs(height);
@@ -184,6 +182,9 @@ int32_t VideoCaptureImpl::IncomingFrame(uint8_t* videoFrame,
}
}
+ int stride_y = target_width;
+ int stride_uv = (target_width + 1) / 2;
+ if (input_image.width() != codec_.width ||
+ input_image.height() != codec_.height) {
+ int ret = UpdateCodecFrameSize(input_image);
+ if (ret < 0) {
+ return ret;
+ }
+ }
+
// Setting absolute height (in case it was negative).
// In Windows, the image starts bottom left, instead of top left.
// Setting a negative source height, inverts the image (within LibYuv).
RTC_DCHECK_EQ(input_image.width(), raw_->d_w);
RTC_DCHECK_EQ(input_image.height(), raw_->d_h);
@@ -1286,6 +1294,48 @@ int LibvpxVp9Encoder::Encode(const VideoFrame& input_image,
return WEBRTC_VIDEO_CODEC_OK;
}
+int LibvpxVp9Encoder::UpdateCodecFrameSize(
+ const VideoFrame& input_image) {
+ RTC_LOG(LS_INFO) << "Reconfiging VP from " <<
+ codec_.width << "x" << codec_.height << " to " <<
+ input_image.width() << "x" << input_image.height();
+ // Preserve latest bitrate/framerate setting
+ // TODO: Mozilla - see below, we need to save more state here.
+ //uint32_t old_bitrate_kbit = config_->rc_target_bitrate;
+ //uint32_t old_framerate = codec_.maxFramerate;
+
+ codec_.width = input_image.width();
+ codec_.height = input_image.height();
+
+ vpx_img_free(raw_);
+ raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height,
+ 1, NULL);
+ // Update encoder context for new frame size.
+ config_->g_w = codec_.width;
+ config_->g_h = codec_.height;
+
+ // Determine number of threads based on the image size and #cores.
+ config_->g_threads = NumberOfThreads(codec_.width, codec_.height,
+ num_cores_);
+
+ // NOTE: We would like to do this the same way vp8 does it
+ // (with vpx_codec_enc_config_set()), but that causes asserts
+ // in AQ 3 (cyclic); and in AQ 0 it works, but on a resize to smaller
+ // than 1/2 x 1/2 original it asserts in convolve(). Given these
+ // bugs in trying to do it the "right" way, we basically re-do
+ // the initialization.
+ vpx_codec_destroy(encoder_); // clean up old state
+ int result = InitAndSetControlSettings(&codec_);
+ if (result == WEBRTC_VIDEO_CODEC_OK) {
+ // TODO: Mozilla rates have become much more complicated, we need to store
+ // more state or find another way of doing this.
+ //return SetRates(old_bitrate_kbit, old_framerate);
+ RTC_CHECK(false);
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ return result;
+}
+
bool LibvpxVp9Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
absl::optional<int>* spatial_idx,
absl::optional<int>* temporal_idx,
--
2.34.1

104
third_party/libwebrtc/moz-patch-stack/0020.patch поставляемый
Просмотреть файл

@ -1,81 +1,47 @@
From: Dan Minor <dminor@mozilla.com>
Date: Thu, 21 Mar 2019 15:48:00 +0000
Subject: Bug 1535584 - Restore UpdateCodecFrameSize to vp9_impl.cc; r=bwc
Date: Fri, 29 Mar 2019 18:43:00 +0000
Subject: Bug 1539220 - Prefer non-RGB24 capabilities when available;
r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D23713
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/0898f9cfe09273e1d86c38abdd576cdf273009f0
We've hit a number of problems with handling of RGB24 video capture on
Windows. This adds a check that will ignore any RGB24 capture capabilities
when determining a best match if there are other capabilities available to
workaround the problems.
Differential Revision: https://phabricator.services.mozilla.com/D25449
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c12307f3817bec87712ab5930493c3135c76b0a0
---
.../codecs/vp9/libvpx_vp9_encoder.cc | 50 +++++++++++++++++++
1 file changed, 50 insertions(+)
modules/video_capture/device_info_impl.cc | 13 +++++++++++++
1 file changed, 13 insertions(+)
diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
index 35b13058a2..5877373b76 100644
--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
@@ -1159,6 +1159,14 @@ int LibvpxVp9Encoder::Encode(const VideoFrame& input_image,
config_changed_ = false;
}
diff --git a/modules/video_capture/device_info_impl.cc b/modules/video_capture/device_info_impl.cc
index 7cccdb51a7..2a6afb3147 100644
--- a/modules/video_capture/device_info_impl.cc
+++ b/modules/video_capture/device_info_impl.cc
@@ -100,10 +100,23 @@ int32_t DeviceInfoImpl::GetBestMatchedCapability(
const int32_t numberOfCapabilies =
static_cast<int32_t>(_captureCapabilities.size());
+ if (input_image.width() != codec_.width ||
+ input_image.height() != codec_.height) {
+ int ret = UpdateCodecFrameSize(input_image);
+ if (ret < 0) {
+ return ret;
+ bool hasNonRGB24Capability = false;
for (int32_t tmp = 0; tmp < numberOfCapabilies;
++tmp) // Loop through all capabilities
{
VideoCaptureCapability& capability = _captureCapabilities[tmp];
+ if (capability.videoType != VideoType::kRGB24) {
+ hasNonRGB24Capability = true;
+ }
+ }
+
RTC_DCHECK_EQ(input_image.width(), raw_->d_w);
RTC_DCHECK_EQ(input_image.height(), raw_->d_h);
+ for (int32_t tmp = 0; tmp < numberOfCapabilies;
+ ++tmp) // Loop through all capabilities
+ {
+ VideoCaptureCapability& capability = _captureCapabilities[tmp];
+ if (hasNonRGB24Capability && capability.videoType == VideoType::kRGB24) {
+ continue;
+ }
@@ -1286,6 +1294,48 @@ int LibvpxVp9Encoder::Encode(const VideoFrame& input_image,
return WEBRTC_VIDEO_CODEC_OK;
}
+int LibvpxVp9Encoder::UpdateCodecFrameSize(
+ const VideoFrame& input_image) {
+ RTC_LOG(LS_INFO) << "Reconfiging VP from " <<
+ codec_.width << "x" << codec_.height << " to " <<
+ input_image.width() << "x" << input_image.height();
+ // Preserve latest bitrate/framerate setting
+ // TODO: Mozilla - see below, we need to save more state here.
+ //uint32_t old_bitrate_kbit = config_->rc_target_bitrate;
+ //uint32_t old_framerate = codec_.maxFramerate;
+
+ codec_.width = input_image.width();
+ codec_.height = input_image.height();
+
+ vpx_img_free(raw_);
+ raw_ = vpx_img_wrap(NULL, VPX_IMG_FMT_I420, codec_.width, codec_.height,
+ 1, NULL);
+ // Update encoder context for new frame size.
+ config_->g_w = codec_.width;
+ config_->g_h = codec_.height;
+
+ // Determine number of threads based on the image size and #cores.
+ config_->g_threads = NumberOfThreads(codec_.width, codec_.height,
+ num_cores_);
+
+ // NOTE: We would like to do this the same way vp8 does it
+ // (with vpx_codec_enc_config_set()), but that causes asserts
+ // in AQ 3 (cyclic); and in AQ 0 it works, but on a resize to smaller
+ // than 1/2 x 1/2 original it asserts in convolve(). Given these
+ // bugs in trying to do it the "right" way, we basically re-do
+ // the initialization.
+ vpx_codec_destroy(encoder_); // clean up old state
+ int result = InitAndSetControlSettings(&codec_);
+ if (result == WEBRTC_VIDEO_CODEC_OK) {
+ // TODO: Mozilla rates have become much more complicated, we need to store
+ // more state or find another way of doing this.
+ //return SetRates(old_bitrate_kbit, old_framerate);
+ RTC_CHECK(false);
+ return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+ }
+ return result;
+}
+
bool LibvpxVp9Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
absl::optional<int>* spatial_idx,
absl::optional<int>* temporal_idx,
const int32_t diffWidth = capability.width - requested.width;
const int32_t diffHeight = capability.height - requested.height;
--
2.34.1

119
third_party/libwebrtc/moz-patch-stack/0021.patch поставляемый
Просмотреть файл

@ -1,47 +1,96 @@
From: Dan Minor <dminor@mozilla.com>
Date: Fri, 29 Mar 2019 18:43:00 +0000
Subject: Bug 1539220 - Prefer non-RGB24 capabilities when available;
r=pehrsons
Date: Tue, 17 Sep 2019 06:47:00 +0000
Subject: Bug 1581193 - Fix devicechange event on Windows; r=achronop
We've hit a number of problems with handling of RGB24 video capture on
Windows. This adds a check that will ignore any RGB24 capture capabilities
when determining a best match if there are other capabilities available to
workaround the problems.
This restores the code for generating devicechange events that was
accidentally removed as part of updating the Windows video capture code
in Bug 1552755.
Differential Revision: https://phabricator.services.mozilla.com/D25449
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c12307f3817bec87712ab5930493c3135c76b0a0
Differential Revision: https://phabricator.services.mozilla.com/D46033
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/840c4edafa021eeac6a7e6ae0e828d0adcfea92e
---
modules/video_capture/device_info_impl.cc | 13 +++++++++++++
1 file changed, 13 insertions(+)
.../video_capture/windows/device_info_ds.cc | 39 +++++++++++++++++++
.../video_capture/windows/device_info_ds.h | 3 ++
2 files changed, 42 insertions(+)
diff --git a/modules/video_capture/device_info_impl.cc b/modules/video_capture/device_info_impl.cc
index 7cccdb51a7..2a6afb3147 100644
--- a/modules/video_capture/device_info_impl.cc
+++ b/modules/video_capture/device_info_impl.cc
@@ -100,10 +100,23 @@ int32_t DeviceInfoImpl::GetBestMatchedCapability(
const int32_t numberOfCapabilies =
static_cast<int32_t>(_captureCapabilities.size());
diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc
index fb8d55137f..8543dce746 100644
--- a/modules/video_capture/windows/device_info_ds.cc
+++ b/modules/video_capture/windows/device_info_ds.cc
@@ -20,6 +20,29 @@
namespace webrtc {
namespace videocapturemodule {
+ bool hasNonRGB24Capability = false;
for (int32_t tmp = 0; tmp < numberOfCapabilies;
++tmp) // Loop through all capabilities
{
VideoCaptureCapability& capability = _captureCapabilities[tmp];
+ if (capability.videoType != VideoType::kRGB24) {
+ hasNonRGB24Capability = true;
+LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
+{
+ DeviceInfoDS* pParent;
+ if (uiMsg == WM_CREATE)
+ {
+ pParent = (DeviceInfoDS*)((LPCREATESTRUCT)lParam)->lpCreateParams;
+ SetWindowLongPtr(hWnd, GWLP_USERDATA, (LONG_PTR)pParent);
+ }
+ }
+ else if (uiMsg == WM_DESTROY)
+ {
+ SetWindowLongPtr(hWnd, GWLP_USERDATA, NULL);
+ }
+ else if (uiMsg == WM_DEVICECHANGE)
+ {
+ pParent = (DeviceInfoDS*)GetWindowLongPtr(hWnd, GWLP_USERDATA);
+ if (pParent)
+ {
+ pParent->DeviceChange();
+ }
+ }
+ return DefWindowProc(hWnd, uiMsg, wParam, lParam);
+}
+
+ for (int32_t tmp = 0; tmp < numberOfCapabilies;
+ ++tmp) // Loop through all capabilities
+ {
+ VideoCaptureCapability& capability = _captureCapabilities[tmp];
+ if (hasNonRGB24Capability && capability.videoType == VideoType::kRGB24) {
+ continue;
+ }
// static
DeviceInfoDS* DeviceInfoDS::Create() {
DeviceInfoDS* dsInfo = new DeviceInfoDS();
@@ -77,6 +100,18 @@ DeviceInfoDS::DeviceInfoDS()
<< rtc::ToHex(hr);
}
}
+
+ _hInstance = reinterpret_cast<HINSTANCE>(GetModuleHandle(NULL));
+ _wndClass = {0};
+ _wndClass.lpfnWndProc = &WndProc;
+ _wndClass.lpszClassName = TEXT("DeviceInfoDS");
+ _wndClass.hInstance = _hInstance;
+
+ if (RegisterClass(&_wndClass)) {
+ _hwnd = CreateWindow(_wndClass.lpszClassName, NULL, 0, CW_USEDEFAULT,
+ CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, NULL,
+ NULL, _hInstance, this);
+ }
}
const int32_t diffWidth = capability.width - requested.width;
const int32_t diffHeight = capability.height - requested.height;
DeviceInfoDS::~DeviceInfoDS() {
@@ -85,6 +120,10 @@ DeviceInfoDS::~DeviceInfoDS() {
if (_CoUninitializeIsRequired) {
CoUninitialize();
}
+ if (_hwnd != NULL) {
+ DestroyWindow(_hwnd);
+ }
+ UnregisterClass(_wndClass.lpszClassName, _hInstance);
}
int32_t DeviceInfoDS::Init() {
diff --git a/modules/video_capture/windows/device_info_ds.h b/modules/video_capture/windows/device_info_ds.h
index 1b52645cde..dc7b9b1a24 100644
--- a/modules/video_capture/windows/device_info_ds.h
+++ b/modules/video_capture/windows/device_info_ds.h
@@ -93,6 +93,9 @@ class DeviceInfoDS : public DeviceInfoImpl {
IEnumMoniker* _dsMonikerDevEnum;
bool _CoUninitializeIsRequired;
std::vector<VideoCaptureCapabilityWindows> _captureCapabilitiesWindows;
+ HWND _hwnd;
+ WNDCLASS _wndClass;
+ HINSTANCE _hInstance;
};
} // namespace videocapturemodule
} // namespace webrtc
--
2.34.1

135
third_party/libwebrtc/moz-patch-stack/0022.patch поставляемый
Просмотреть файл

@ -1,93 +1,94 @@
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 17 Sep 2019 06:47:00 +0000
Subject: Bug 1581193 - Fix devicechange event on Windows; r=achronop
From: Alex Chronopoulos <achronop@gmail.com>
Date: Tue, 17 Sep 2019 14:31:00 +0000
Subject: Bug 1581806 - Trigger devicechange event for audio and video input
devices only. r=dminor
This restores the code for generating devicechange events that was
accidentally removed as part of updating the Windows video capture code
in Bug 1552755.
After Bug 1581193 devicechange notifications were triggered with any device change, not just video and audio devices. This patch limits down the notifications to only video and audio input devices change.
Differential Revision: https://phabricator.services.mozilla.com/D46033
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/840c4edafa021eeac6a7e6ae0e828d0adcfea92e
Differential Revision: https://phabricator.services.mozilla.com/D46147
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/1feec83ee6f92a35de0d4b27ebea04e68a0d7ef0
---
.../video_capture/windows/device_info_ds.cc | 39 +++++++++++++++++++
.../video_capture/windows/device_info_ds.h | 3 ++
2 files changed, 42 insertions(+)
.../video_capture/windows/device_info_ds.cc | 29 +++++++++++++++++--
.../video_capture/windows/device_info_ds.h | 1 +
2 files changed, 28 insertions(+), 2 deletions(-)
diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc
index fb8d55137f..8543dce746 100644
index 8543dce746..96db60c968 100644
--- a/modules/video_capture/windows/device_info_ds.cc
+++ b/modules/video_capture/windows/device_info_ds.cc
@@ -20,6 +20,29 @@
@@ -20,6 +20,18 @@
namespace webrtc {
namespace videocapturemodule {
+LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
+BOOL isCaptureDevice(DEV_BROADCAST_HDR *pHdr)
+{
+ DeviceInfoDS* pParent;
+ if (uiMsg == WM_CREATE)
+ {
+ pParent = (DeviceInfoDS*)((LPCREATESTRUCT)lParam)->lpCreateParams;
+ SetWindowLongPtr(hWnd, GWLP_USERDATA, (LONG_PTR)pParent);
+ }
+ else if (uiMsg == WM_DESTROY)
+ {
+ SetWindowLongPtr(hWnd, GWLP_USERDATA, NULL);
+ }
+ else if (uiMsg == WM_DEVICECHANGE)
+ {
+ pParent = (DeviceInfoDS*)GetWindowLongPtr(hWnd, GWLP_USERDATA);
+ if (pParent)
+ {
+ pParent->DeviceChange();
+ }
+ }
+ return DefWindowProc(hWnd, uiMsg, wParam, lParam);
+ if (pHdr == NULL) {
+ return FALSE;
+ }
+ if (pHdr->dbch_devicetype != DBT_DEVTYP_DEVICEINTERFACE) {
+ return FALSE;
+ }
+ DEV_BROADCAST_DEVICEINTERFACE* pDi = (DEV_BROADCAST_DEVICEINTERFACE*)pHdr;
+ return pDi->dbcc_classguid == KSCATEGORY_CAPTURE;
+}
+
// static
DeviceInfoDS* DeviceInfoDS::Create() {
DeviceInfoDS* dsInfo = new DeviceInfoDS();
@@ -77,6 +100,18 @@ DeviceInfoDS::DeviceInfoDS()
<< rtc::ToHex(hr);
}
LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
{
DeviceInfoDS* pParent;
@@ -35,7 +47,7 @@ LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
else if (uiMsg == WM_DEVICECHANGE)
{
pParent = (DeviceInfoDS*)GetWindowLongPtr(hWnd, GWLP_USERDATA);
- if (pParent)
+ if (pParent && isCaptureDevice((PDEV_BROADCAST_HDR)lParam))
{
pParent->DeviceChange();
}
@@ -56,7 +68,8 @@ DeviceInfoDS* DeviceInfoDS::Create() {
DeviceInfoDS::DeviceInfoDS()
: _dsDevEnum(NULL),
_dsMonikerDevEnum(NULL),
- _CoUninitializeIsRequired(true) {
+ _CoUninitializeIsRequired(true),
+ _hdevnotify(NULL) {
// 1) Initialize the COM library (make Windows load the DLLs).
//
// CoInitializeEx must be called at least once, and is usually called only
@@ -111,6 +124,14 @@ DeviceInfoDS::DeviceInfoDS()
_hwnd = CreateWindow(_wndClass.lpszClassName, NULL, 0, CW_USEDEFAULT,
CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, NULL,
NULL, _hInstance, this);
+
+ DEV_BROADCAST_DEVICEINTERFACE di = { 0 };
+ di.dbcc_size = sizeof(di);
+ di.dbcc_devicetype = DBT_DEVTYP_DEVICEINTERFACE;
+ di.dbcc_classguid = KSCATEGORY_CAPTURE;
+
+ _hdevnotify = RegisterDeviceNotification(_hwnd, &di,
+ DEVICE_NOTIFY_WINDOW_HANDLE);
}
+
+ _hInstance = reinterpret_cast<HINSTANCE>(GetModuleHandle(NULL));
+ _wndClass = {0};
+ _wndClass.lpfnWndProc = &WndProc;
+ _wndClass.lpszClassName = TEXT("DeviceInfoDS");
+ _wndClass.hInstance = _hInstance;
+
+ if (RegisterClass(&_wndClass)) {
+ _hwnd = CreateWindow(_wndClass.lpszClassName, NULL, 0, CW_USEDEFAULT,
+ CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, NULL,
+ NULL, _hInstance, this);
+ }
}
DeviceInfoDS::~DeviceInfoDS() {
@@ -85,6 +120,10 @@ DeviceInfoDS::~DeviceInfoDS() {
@@ -120,6 +141,10 @@ DeviceInfoDS::~DeviceInfoDS() {
if (_CoUninitializeIsRequired) {
CoUninitialize();
}
+ if (_hwnd != NULL) {
+ DestroyWindow(_hwnd);
+ if (_hdevnotify)
+ {
+ UnregisterDeviceNotification(_hdevnotify);
+ }
+ UnregisterClass(_wndClass.lpszClassName, _hInstance);
}
int32_t DeviceInfoDS::Init() {
if (_hwnd != NULL) {
DestroyWindow(_hwnd);
}
diff --git a/modules/video_capture/windows/device_info_ds.h b/modules/video_capture/windows/device_info_ds.h
index 1b52645cde..dc7b9b1a24 100644
index dc7b9b1a24..ed2a726d6f 100644
--- a/modules/video_capture/windows/device_info_ds.h
+++ b/modules/video_capture/windows/device_info_ds.h
@@ -93,6 +93,9 @@ class DeviceInfoDS : public DeviceInfoImpl {
IEnumMoniker* _dsMonikerDevEnum;
bool _CoUninitializeIsRequired;
std::vector<VideoCaptureCapabilityWindows> _captureCapabilitiesWindows;
+ HWND _hwnd;
+ WNDCLASS _wndClass;
+ HINSTANCE _hInstance;
@@ -96,6 +96,7 @@ class DeviceInfoDS : public DeviceInfoImpl {
HWND _hwnd;
WNDCLASS _wndClass;
HINSTANCE _hInstance;
+ HDEVNOTIFY _hdevnotify;
};
} // namespace videocapturemodule
} // namespace webrtc

171
third_party/libwebrtc/moz-patch-stack/0023.patch поставляемый
Просмотреть файл

@ -1,97 +1,98 @@
From: Alex Chronopoulos <achronop@gmail.com>
Date: Tue, 17 Sep 2019 14:31:00 +0000
Subject: Bug 1581806 - Trigger devicechange event for audio and video input
devices only. r=dminor
Date: Wed, 18 Sep 2019 13:16:00 +0000
Subject: Bug 1572281 - Remove audio device change notifications from video
capture in Linux. r=dminor
After Bug 1581193 devicechange notifications were triggered with any device change, not just video and audio devices. This patch limits down the notifications to only video and audio input devices change.
Video capture used to provide device change notifications for audio and video devices. From now on, CubebDeviceEnumerator will provide audio device change notifications thus video capture is updated to notify only changes of the video device. This is the Linux part.
Differential Revision: https://phabricator.services.mozilla.com/D46147
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/1feec83ee6f92a35de0d4b27ebea04e68a0d7ef0
Differential Revision: https://phabricator.services.mozilla.com/D46272
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7bf7263db30b794139332691f4fbc98b4bfcfdd7
---
.../video_capture/windows/device_info_ds.cc | 29 +++++++++++++++++--
.../video_capture/windows/device_info_ds.h | 1 +
2 files changed, 28 insertions(+), 2 deletions(-)
.../video_capture/linux/device_info_v4l2.cc | 28 ++-----------------
1 file changed, 3 insertions(+), 25 deletions(-)
diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc
index 8543dce746..96db60c968 100644
--- a/modules/video_capture/windows/device_info_ds.cc
+++ b/modules/video_capture/windows/device_info_ds.cc
@@ -20,6 +20,18 @@
namespace webrtc {
namespace videocapturemodule {
+BOOL isCaptureDevice(DEV_BROADCAST_HDR *pHdr)
+{
+ if (pHdr == NULL) {
+ return FALSE;
+ }
+ if (pHdr->dbch_devicetype != DBT_DEVTYP_DEVICEINTERFACE) {
+ return FALSE;
+ }
+ DEV_BROADCAST_DEVICEINTERFACE* pDi = (DEV_BROADCAST_DEVICEINTERFACE*)pHdr;
+ return pDi->dbcc_classguid == KSCATEGORY_CAPTURE;
+}
+
LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index d836747b4a..77968b7eaf 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -44,7 +44,7 @@ namespace videocapturemodule {
void DeviceInfoV4l2::HandleEvent(inotify_event* event, int fd)
{
DeviceInfoDS* pParent;
@@ -35,7 +47,7 @@ LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
else if (uiMsg == WM_DEVICECHANGE)
{
pParent = (DeviceInfoDS*)GetWindowLongPtr(hWnd, GWLP_USERDATA);
- if (pParent)
+ if (pParent && isCaptureDevice((PDEV_BROADCAST_HDR)lParam))
{
pParent->DeviceChange();
if (event->mask & IN_CREATE) {
- if (fd == _fd_v4l || fd == _fd_snd) {
+ if (fd == _fd_v4l) {
DeviceChange();
} else if ((event->mask & IN_ISDIR) && (fd == _fd_dev)) {
if (_wd_v4l < 0) {
@@ -56,25 +56,15 @@ void DeviceInfoV4l2::HandleEvent(inotify_event* event, int fd)
DeviceChange();
}
}
- if (_wd_snd < 0) {
- usleep(5*1000);
- _wd_snd = inotify_add_watch(_fd_snd, "/dev/snd/by-path/", IN_CREATE | IN_DELETE | IN_DELETE_SELF);
- if (_wd_snd >= 0) {
- DeviceChange();
- }
- }
}
@@ -56,7 +68,8 @@ DeviceInfoDS* DeviceInfoDS::Create() {
DeviceInfoDS::DeviceInfoDS()
: _dsDevEnum(NULL),
_dsMonikerDevEnum(NULL),
- _CoUninitializeIsRequired(true) {
+ _CoUninitializeIsRequired(true),
+ _hdevnotify(NULL) {
// 1) Initialize the COM library (make Windows load the DLLs).
//
// CoInitializeEx must be called at least once, and is usually called only
@@ -111,6 +124,14 @@ DeviceInfoDS::DeviceInfoDS()
_hwnd = CreateWindow(_wndClass.lpszClassName, NULL, 0, CW_USEDEFAULT,
CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, NULL,
NULL, _hInstance, this);
+
+ DEV_BROADCAST_DEVICEINTERFACE di = { 0 };
+ di.dbcc_size = sizeof(di);
+ di.dbcc_devicetype = DBT_DEVTYP_DEVICEINTERFACE;
+ di.dbcc_classguid = KSCATEGORY_CAPTURE;
+
+ _hdevnotify = RegisterDeviceNotification(_hwnd, &di,
+ DEVICE_NOTIFY_WINDOW_HANDLE);
}
} else if (event->mask & IN_DELETE) {
- if (fd == _fd_v4l || fd == _fd_snd) {
+ if (fd == _fd_v4l) {
DeviceChange();
}
} else if (event->mask & IN_DELETE_SELF) {
if (fd == _fd_v4l) {
inotify_rm_watch(_fd_v4l, _wd_v4l);
_wd_v4l = -1;
- } else if (fd == _fd_snd) {
- inotify_rm_watch(_fd_snd, _wd_snd);
- _wd_snd = -1;
} else {
assert(false);
}
@@ -141,11 +131,6 @@ int DeviceInfoV4l2::ProcessInotifyEvents()
break;
}
}
- if (EventCheck(_fd_snd) > 0) {
- if (HandleEvents(_fd_snd) < 0) {
- break;
- }
- }
}
return 0;
}
@@ -158,11 +143,9 @@ void DeviceInfoV4l2::InotifyEventThread(void* obj)
void DeviceInfoV4l2::InotifyProcess()
{
_fd_v4l = inotify_init();
- _fd_snd = inotify_init();
_fd_dev = inotify_init();
- if (_fd_v4l >= 0 && _fd_snd >= 0 && _fd_dev >= 0) {
+ if (_fd_v4l >= 0 && _fd_dev >= 0) {
_wd_v4l = inotify_add_watch(_fd_v4l, "/dev/v4l/by-path/", IN_CREATE | IN_DELETE | IN_DELETE_SELF);
- _wd_snd = inotify_add_watch(_fd_snd, "/dev/snd/by-path/", IN_CREATE | IN_DELETE | IN_DELETE_SELF);
_wd_dev = inotify_add_watch(_fd_dev, "/dev/", IN_CREATE);
ProcessInotifyEvents();
@@ -120,6 +141,10 @@ DeviceInfoDS::~DeviceInfoDS() {
if (_CoUninitializeIsRequired) {
CoUninitialize();
}
+ if (_hdevnotify)
+ {
+ UnregisterDeviceNotification(_hdevnotify);
+ }
if (_hwnd != NULL) {
DestroyWindow(_hwnd);
}
diff --git a/modules/video_capture/windows/device_info_ds.h b/modules/video_capture/windows/device_info_ds.h
index dc7b9b1a24..ed2a726d6f 100644
--- a/modules/video_capture/windows/device_info_ds.h
+++ b/modules/video_capture/windows/device_info_ds.h
@@ -96,6 +96,7 @@ class DeviceInfoDS : public DeviceInfoImpl {
HWND _hwnd;
WNDCLASS _wndClass;
HINSTANCE _hInstance;
+ HDEVNOTIFY _hdevnotify;
};
} // namespace videocapturemodule
} // namespace webrtc
@@ -170,16 +153,11 @@ void DeviceInfoV4l2::InotifyProcess()
inotify_rm_watch(_fd_v4l, _wd_v4l);
}
- if (_wd_snd >= 0) {
- inotify_rm_watch(_fd_snd, _wd_snd);
- }
-
if (_wd_dev >= 0) {
inotify_rm_watch(_fd_dev, _wd_dev);
}
close(_fd_v4l);
- close(_fd_snd);
close(_fd_dev);
}
}
--
2.34.1

126
third_party/libwebrtc/moz-patch-stack/0024.patch поставляемый
Просмотреть файл

@ -1,98 +1,56 @@
From: Alex Chronopoulos <achronop@gmail.com>
Date: Wed, 18 Sep 2019 13:16:00 +0000
Date: Wed, 18 Sep 2019 13:12:00 +0000
Subject: Bug 1572281 - Remove audio device change notifications from video
capture in Linux. r=dminor
capture in Windows. r=dminor
Video capture used to provide device change notifications for audio and video devices. From now on, CubebDeviceEnumerator will provide audio device change notifications thus video capture is updated to notify only changes of the video device. This is the Linux part.
Video capture used to provide device change notifications for audio and video devices. From now on, CubebDeviceEnumerator will provide audio device change notifications thus video capture is updated to notify only changes of the video device. This is the windows part.
Differential Revision: https://phabricator.services.mozilla.com/D46272
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7bf7263db30b794139332691f4fbc98b4bfcfdd7
Differential Revision: https://phabricator.services.mozilla.com/D46274
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a5c23245837ebdd99532e8bfaca05774c5c96d9d
---
.../video_capture/linux/device_info_v4l2.cc | 28 ++-----------------
1 file changed, 3 insertions(+), 25 deletions(-)
modules/video_capture/windows/device_info_ds.cc | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index d836747b4a..77968b7eaf 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -44,7 +44,7 @@ namespace videocapturemodule {
void DeviceInfoV4l2::HandleEvent(inotify_event* event, int fd)
diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc
index 96db60c968..3ab95837c0 100644
--- a/modules/video_capture/windows/device_info_ds.cc
+++ b/modules/video_capture/windows/device_info_ds.cc
@@ -20,7 +20,7 @@
namespace webrtc {
namespace videocapturemodule {
-BOOL isCaptureDevice(DEV_BROADCAST_HDR *pHdr)
+BOOL isVideoDevice(DEV_BROADCAST_HDR *pHdr)
{
if (event->mask & IN_CREATE) {
- if (fd == _fd_v4l || fd == _fd_snd) {
+ if (fd == _fd_v4l) {
DeviceChange();
} else if ((event->mask & IN_ISDIR) && (fd == _fd_dev)) {
if (_wd_v4l < 0) {
@@ -56,25 +56,15 @@ void DeviceInfoV4l2::HandleEvent(inotify_event* event, int fd)
DeviceChange();
}
}
- if (_wd_snd < 0) {
- usleep(5*1000);
- _wd_snd = inotify_add_watch(_fd_snd, "/dev/snd/by-path/", IN_CREATE | IN_DELETE | IN_DELETE_SELF);
- if (_wd_snd >= 0) {
- DeviceChange();
- }
- }
}
} else if (event->mask & IN_DELETE) {
- if (fd == _fd_v4l || fd == _fd_snd) {
+ if (fd == _fd_v4l) {
DeviceChange();
}
} else if (event->mask & IN_DELETE_SELF) {
if (fd == _fd_v4l) {
inotify_rm_watch(_fd_v4l, _wd_v4l);
_wd_v4l = -1;
- } else if (fd == _fd_snd) {
- inotify_rm_watch(_fd_snd, _wd_snd);
- _wd_snd = -1;
} else {
assert(false);
}
@@ -141,11 +131,6 @@ int DeviceInfoV4l2::ProcessInotifyEvents()
break;
}
}
- if (EventCheck(_fd_snd) > 0) {
- if (HandleEvents(_fd_snd) < 0) {
- break;
- }
- }
}
return 0;
if (pHdr == NULL) {
return FALSE;
@@ -29,7 +29,7 @@ BOOL isCaptureDevice(DEV_BROADCAST_HDR *pHdr)
return FALSE;
}
DEV_BROADCAST_DEVICEINTERFACE* pDi = (DEV_BROADCAST_DEVICEINTERFACE*)pHdr;
- return pDi->dbcc_classguid == KSCATEGORY_CAPTURE;
+ return pDi->dbcc_classguid == KSCATEGORY_VIDEO_CAMERA;
}
@@ -158,11 +143,9 @@ void DeviceInfoV4l2::InotifyEventThread(void* obj)
void DeviceInfoV4l2::InotifyProcess()
{
_fd_v4l = inotify_init();
- _fd_snd = inotify_init();
_fd_dev = inotify_init();
- if (_fd_v4l >= 0 && _fd_snd >= 0 && _fd_dev >= 0) {
+ if (_fd_v4l >= 0 && _fd_dev >= 0) {
_wd_v4l = inotify_add_watch(_fd_v4l, "/dev/v4l/by-path/", IN_CREATE | IN_DELETE | IN_DELETE_SELF);
- _wd_snd = inotify_add_watch(_fd_snd, "/dev/snd/by-path/", IN_CREATE | IN_DELETE | IN_DELETE_SELF);
_wd_dev = inotify_add_watch(_fd_dev, "/dev/", IN_CREATE);
ProcessInotifyEvents();
@@ -170,16 +153,11 @@ void DeviceInfoV4l2::InotifyProcess()
inotify_rm_watch(_fd_v4l, _wd_v4l);
LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
@@ -47,7 +47,7 @@ LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
else if (uiMsg == WM_DEVICECHANGE)
{
pParent = (DeviceInfoDS*)GetWindowLongPtr(hWnd, GWLP_USERDATA);
- if (pParent && isCaptureDevice((PDEV_BROADCAST_HDR)lParam))
+ if (pParent && isVideoDevice((PDEV_BROADCAST_HDR)lParam))
{
pParent->DeviceChange();
}
@@ -128,7 +128,7 @@ DeviceInfoDS::DeviceInfoDS()
DEV_BROADCAST_DEVICEINTERFACE di = { 0 };
di.dbcc_size = sizeof(di);
di.dbcc_devicetype = DBT_DEVTYP_DEVICEINTERFACE;
- di.dbcc_classguid = KSCATEGORY_CAPTURE;
+ di.dbcc_classguid = KSCATEGORY_VIDEO_CAMERA;
- if (_wd_snd >= 0) {
- inotify_rm_watch(_fd_snd, _wd_snd);
- }
-
if (_wd_dev >= 0) {
inotify_rm_watch(_fd_dev, _wd_dev);
}
close(_fd_v4l);
- close(_fd_snd);
close(_fd_dev);
}
}
_hdevnotify = RegisterDeviceNotification(_hwnd, &di,
DEVICE_NOTIFY_WINDOW_HANDLE);
--
2.34.1

96
third_party/libwebrtc/moz-patch-stack/0025.patch поставляемый
Просмотреть файл

@ -1,56 +1,58 @@
From: Alex Chronopoulos <achronop@gmail.com>
Date: Wed, 18 Sep 2019 13:12:00 +0000
Subject: Bug 1572281 - Remove audio device change notifications from video
capture in Windows. r=dminor
From: Wang Qing <wangqing-hf@loongson.cn>
Date: Wed, 25 Sep 2019 14:15:00 +0000
Subject: Bug 1579834 - [WebRTC] Add mips64 support; r=dminor
Video capture used to provide device change notifications for audio and video devices. From now on, CubebDeviceEnumerator will provide audio device change notifications thus video capture is updated to notify only changes of the video device. This is the windows part.
Differential Revision: https://phabricator.services.mozilla.com/D46274
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a5c23245837ebdd99532e8bfaca05774c5c96d9d
Differential Revision: https://phabricator.services.mozilla.com/D45620
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/1387b2c480b55ecca3fbdf62bd7649dafc62438d
---
modules/video_capture/windows/device_info_ds.cc | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc | 2 +-
modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc | 6 +++---
2 files changed, 4 insertions(+), 4 deletions(-)
diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc
index 96db60c968..3ab95837c0 100644
--- a/modules/video_capture/windows/device_info_ds.cc
+++ b/modules/video_capture/windows/device_info_ds.cc
@@ -20,7 +20,7 @@
namespace webrtc {
namespace videocapturemodule {
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
index 1ac71899ee..3fe86f2f85 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
@@ -47,7 +47,7 @@ const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Config-Arm";
const char kVp8PostProcFieldTrial[] = "WebRTC-VP8-Postproc-Config";
-BOOL isCaptureDevice(DEV_BROADCAST_HDR *pHdr)
+BOOL isVideoDevice(DEV_BROADCAST_HDR *pHdr)
{
if (pHdr == NULL) {
return FALSE;
@@ -29,7 +29,7 @@ BOOL isCaptureDevice(DEV_BROADCAST_HDR *pHdr)
return FALSE;
}
DEV_BROADCAST_DEVICEINTERFACE* pDi = (DEV_BROADCAST_DEVICEINTERFACE*)pHdr;
- return pDi->dbcc_classguid == KSCATEGORY_CAPTURE;
+ return pDi->dbcc_classguid == KSCATEGORY_VIDEO_CAMERA;
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
- defined(WEBRTC_ANDROID)
+ defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
constexpr bool kIsArm = true;
#else
constexpr bool kIsArm = false;
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
index 8e401fcc7b..cc84605ce7 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
@@ -687,7 +687,7 @@ int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst,
int LibvpxVp8Encoder::GetCpuSpeed(int width, int height) {
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
- defined(WEBRTC_ANDROID)
+ defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
// On mobile platform, use a lower speed setting for lower resolutions for
// CPUs with 4 or more cores.
RTC_DCHECK_GT(number_of_cores_, 0);
@@ -720,7 +720,7 @@ int LibvpxVp8Encoder::GetCpuSpeed(int width, int height) {
}
LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
@@ -47,7 +47,7 @@ LRESULT CALLBACK WndProc(HWND hWnd, UINT uiMsg, WPARAM wParam, LPARAM lParam)
else if (uiMsg == WM_DEVICECHANGE)
{
pParent = (DeviceInfoDS*)GetWindowLongPtr(hWnd, GWLP_USERDATA);
- if (pParent && isCaptureDevice((PDEV_BROADCAST_HDR)lParam))
+ if (pParent && isVideoDevice((PDEV_BROADCAST_HDR)lParam))
{
pParent->DeviceChange();
}
@@ -128,7 +128,7 @@ DeviceInfoDS::DeviceInfoDS()
DEV_BROADCAST_DEVICEINTERFACE di = { 0 };
di.dbcc_size = sizeof(di);
di.dbcc_devicetype = DBT_DEVTYP_DEVICEINTERFACE;
- di.dbcc_classguid = KSCATEGORY_CAPTURE;
+ di.dbcc_classguid = KSCATEGORY_VIDEO_CAMERA;
_hdevnotify = RegisterDeviceNotification(_hwnd, &di,
DEVICE_NOTIFY_WINDOW_HANDLE);
int LibvpxVp8Encoder::NumberOfThreads(int width, int height, int cpus) {
-#if defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
if (width * height >= 320 * 180) {
if (cpus >= 4) {
// 3 threads for CPUs with 4 and more cores since most of times only 4
@@ -794,7 +794,7 @@ int LibvpxVp8Encoder::InitAndSetControlSettings() {
// multi-res encoding feature?
denoiserState denoiser_state = kDenoiserOnYOnly;
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
- defined(WEBRTC_ANDROID)
+ defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
denoiser_state = kDenoiserOnYOnly;
#else
denoiser_state = kDenoiserOnAdaptive;
--
2.34.1

73
third_party/libwebrtc/moz-patch-stack/0026.patch поставляемый
Просмотреть файл

@ -1,58 +1,29 @@
From: Wang Qing <wangqing-hf@loongson.cn>
Date: Wed, 25 Sep 2019 14:15:00 +0000
Subject: Bug 1579834 - [WebRTC] Add mips64 support; r=dminor
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 9 Oct 2019 20:12:00 +0000
Subject: Bug 1587159 - Fix undefined shift in g722_encode.c; r=ng
Differential Revision: https://phabricator.services.mozilla.com/D45620
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/1387b2c480b55ecca3fbdf62bd7649dafc62438d
Left shifting a negative value results in undefined behaviour. It is safer to
multiply in this case.
Differential Revision: https://phabricator.services.mozilla.com/D48751
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/385d660fac359c907986e08d1d89ab5a353f30b2
---
modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc | 2 +-
modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc | 6 +++---
2 files changed, 4 insertions(+), 4 deletions(-)
modules/third_party/g722/g722_encode.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
index 1ac71899ee..3fe86f2f85 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
@@ -47,7 +47,7 @@ const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Config-Arm";
const char kVp8PostProcFieldTrial[] = "WebRTC-VP8-Postproc-Config";
diff --git a/modules/third_party/g722/g722_encode.c b/modules/third_party/g722/g722_encode.c
index 10a5bcfe7c..fedf9f5961 100644
--- a/modules/third_party/g722/g722_encode.c
+++ b/modules/third_party/g722/g722_encode.c
@@ -74,7 +74,7 @@ static void block4(G722EncoderState *s, int band, int d)
/* Block 4, UPPOL2 */
for (i = 0; i < 3; i++)
s->band[band].sg[i] = s->band[band].p[i] >> 15;
- wd1 = saturate(s->band[band].a[1] << 2);
+ wd1 = saturate(s->band[band].a[1] * 4);
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
- defined(WEBRTC_ANDROID)
+ defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
constexpr bool kIsArm = true;
#else
constexpr bool kIsArm = false;
diff --git a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
index 8e401fcc7b..cc84605ce7 100644
--- a/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
+++ b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
@@ -687,7 +687,7 @@ int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst,
int LibvpxVp8Encoder::GetCpuSpeed(int width, int height) {
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
- defined(WEBRTC_ANDROID)
+ defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
// On mobile platform, use a lower speed setting for lower resolutions for
// CPUs with 4 or more cores.
RTC_DCHECK_GT(number_of_cores_, 0);
@@ -720,7 +720,7 @@ int LibvpxVp8Encoder::GetCpuSpeed(int width, int height) {
}
int LibvpxVp8Encoder::NumberOfThreads(int width, int height, int cpus) {
-#if defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
if (width * height >= 320 * 180) {
if (cpus >= 4) {
// 3 threads for CPUs with 4 and more cores since most of times only 4
@@ -794,7 +794,7 @@ int LibvpxVp8Encoder::InitAndSetControlSettings() {
// multi-res encoding feature?
denoiserState denoiser_state = kDenoiserOnYOnly;
#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
- defined(WEBRTC_ANDROID)
+ defined(WEBRTC_ANDROID) || defined(WEBRTC_ARCH_MIPS)
denoiser_state = kDenoiserOnYOnly;
#else
denoiser_state = kDenoiserOnAdaptive;
wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
if (wd2 > 32767)
--
2.34.1

161
third_party/libwebrtc/moz-patch-stack/0027.patch поставляемый
Просмотреть файл

@ -1,29 +1,148 @@
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 9 Oct 2019 20:12:00 +0000
Subject: Bug 1587159 - Fix undefined shift in g722_encode.c; r=ng
From: Gabriele Svelto <gsvelto@mozilla.com>
Date: Mon, 28 Oct 2019 23:26:00 +0000
Subject: Bug 1590984 - Use poll() instead of select() in WebRTC code r=drno
Left shifting a negative value results in undefined behaviour. It is safer to
multiply in this case.
The use of select() was leading to crashes when the file descriptor value was
larger than FD_SETSIZE. Recent versions of glibc have checks in the FD_CLR(),
FD_SET() and FD_ISSET() macros that will abort() the program instead of doing
an out-of-bounds access. poll() doesn't have limitations on the file
descriptor values and provides behavior that is otherwise identical to
select() thus solving the problem.
Differential Revision: https://phabricator.services.mozilla.com/D48751
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/385d660fac359c907986e08d1d89ab5a353f30b2
Differential Revision: https://phabricator.services.mozilla.com/D50798
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/59fb6760bb6785a6f8a51be6fc66bf04cfba3e16
---
modules/third_party/g722/g722_encode.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
.../video_capture/linux/device_info_linux.cc | 1 +
.../video_capture/linux/device_info_v4l2.cc | 16 +++++-----
.../linux/video_capture_linux.cc | 1 +
.../video_capture/linux/video_capture_v4l2.cc | 29 +++++++++++--------
4 files changed, 26 insertions(+), 21 deletions(-)
diff --git a/modules/third_party/g722/g722_encode.c b/modules/third_party/g722/g722_encode.c
index 10a5bcfe7c..fedf9f5961 100644
--- a/modules/third_party/g722/g722_encode.c
+++ b/modules/third_party/g722/g722_encode.c
@@ -74,7 +74,7 @@ static void block4(G722EncoderState *s, int band, int d)
/* Block 4, UPPOL2 */
for (i = 0; i < 3; i++)
s->band[band].sg[i] = s->band[band].p[i] >> 15;
- wd1 = saturate(s->band[band].a[1] << 2);
+ wd1 = saturate(s->band[band].a[1] * 4);
diff --git a/modules/video_capture/linux/device_info_linux.cc b/modules/video_capture/linux/device_info_linux.cc
index 9c12b2183e..4821cbccd5 100644
--- a/modules/video_capture/linux/device_info_linux.cc
+++ b/modules/video_capture/linux/device_info_linux.cc
@@ -10,6 +10,7 @@
wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
if (wd2 > 32767)
#include <errno.h>
#include <fcntl.h>
+#include <poll.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index 77968b7eaf..f87fe53abe 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -12,6 +12,7 @@
#include <errno.h>
#include <fcntl.h>
+#include <poll.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
@@ -73,16 +74,13 @@ void DeviceInfoV4l2::HandleEvent(inotify_event* event, int fd)
int DeviceInfoV4l2::EventCheck(int fd)
{
- struct timeval timeout;
- fd_set rfds;
+ struct pollfd fds = {
+ .fd = fd,
+ .events = POLLIN,
+ .revents = 0
+ };
- timeout.tv_sec = 0;
- timeout.tv_usec = 100000;
-
- FD_ZERO(&rfds);
- FD_SET(fd, &rfds);
-
- return select(fd+1, &rfds, NULL, NULL, &timeout);
+ return poll(&fds, 1, 100);
}
int DeviceInfoV4l2::HandleEvents(int fd)
diff --git a/modules/video_capture/linux/video_capture_linux.cc b/modules/video_capture/linux/video_capture_linux.cc
index 4895a1ab71..f3324a8e68 100644
--- a/modules/video_capture/linux/video_capture_linux.cc
+++ b/modules/video_capture/linux/video_capture_linux.cc
@@ -10,6 +10,7 @@
#include <errno.h>
#include <fcntl.h>
+#include <poll.h>
#include <stdio.h>
#include <string.h>
#include <sys/ioctl.h>
diff --git a/modules/video_capture/linux/video_capture_v4l2.cc b/modules/video_capture/linux/video_capture_v4l2.cc
index 1dc13b01aa..b527a331e4 100644
--- a/modules/video_capture/linux/video_capture_v4l2.cc
+++ b/modules/video_capture/linux/video_capture_v4l2.cc
@@ -12,7 +12,7 @@
#include <errno.h>
#include <fcntl.h>
-#include <linux/videodev2.h>
+#include <poll.h>
#include <stdio.h>
#include <string.h>
#include <sys/ioctl.h>
@@ -20,6 +20,14 @@
#include <sys/select.h>
#include <time.h>
#include <unistd.h>
+// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
#include <new>
#include <string>
@@ -359,16 +367,13 @@ bool VideoCaptureModuleV4L2::CaptureStarted() {
bool VideoCaptureModuleV4L2::CaptureProcess() {
int retVal = 0;
- fd_set rSet;
- struct timeval timeout;
+ struct pollfd rSet;
- FD_ZERO(&rSet);
- FD_SET(_deviceFd, &rSet);
- timeout.tv_sec = 1;
- timeout.tv_usec = 0;
+ rSet.fd = _deviceFd;
+ rSet.events = POLLIN;
+ rSet.revents = 0;
- // _deviceFd written only in StartCapture, when this thread isn't running.
- retVal = select(_deviceFd + 1, &rSet, NULL, NULL, &timeout);
+ retVal = poll(&rSet, 1, 1000);
{
MutexLock lock(&capture_lock_);
@@ -378,12 +383,12 @@ bool VideoCaptureModuleV4L2::CaptureProcess() {
}
if (retVal < 0 && errno != EINTR) { // continue if interrupted
- // select failed
+ // poll failed
return false;
} else if (retVal == 0) {
- // select timed out
+ // poll timed out
return true;
- } else if (!FD_ISSET(_deviceFd, &rSet)) {
+ } else if (!(rSet.revents & POLLIN)) {
// not event on camera handle
return true;
}
--
2.34.1

160
third_party/libwebrtc/moz-patch-stack/0028.patch поставляемый
Просмотреть файл

@ -1,148 +1,28 @@
From: Gabriele Svelto <gsvelto@mozilla.com>
Date: Mon, 28 Oct 2019 23:26:00 +0000
Subject: Bug 1590984 - Use poll() instead of select() in WebRTC code r=drno
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 11 Feb 2020 17:04:00 +0000
Subject: Bug 1578073 - Fix warning: [cast] redundant cast to int; r=ng
The use of select() was leading to crashes when the file descriptor value was
larger than FD_SETSIZE. Recent versions of glibc have checks in the FD_CLR(),
FD_SET() and FD_ISSET() macros that will abort() the program instead of doing
an out-of-bounds access. poll() doesn't have limitations on the file
descriptor values and provides behavior that is otherwise identical to
select() thus solving the problem.
Differential Revision: https://phabricator.services.mozilla.com/D50798
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/59fb6760bb6785a6f8a51be6fc66bf04cfba3e16
Differential Revision: https://phabricator.services.mozilla.com/D61852
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/82877c8a864072c03f45234c6649645ed609e098
---
.../video_capture/linux/device_info_linux.cc | 1 +
.../video_capture/linux/device_info_v4l2.cc | 16 +++++-----
.../linux/video_capture_linux.cc | 1 +
.../video_capture/linux/video_capture_v4l2.cc | 29 +++++++++++--------
4 files changed, 26 insertions(+), 21 deletions(-)
sdk/android/api/org/webrtc/TextureBufferImpl.java | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/modules/video_capture/linux/device_info_linux.cc b/modules/video_capture/linux/device_info_linux.cc
index 9c12b2183e..4821cbccd5 100644
--- a/modules/video_capture/linux/device_info_linux.cc
+++ b/modules/video_capture/linux/device_info_linux.cc
@@ -10,6 +10,7 @@
diff --git a/sdk/android/api/org/webrtc/TextureBufferImpl.java b/sdk/android/api/org/webrtc/TextureBufferImpl.java
index c08fc4c29b..8e0e40ef70 100644
--- a/sdk/android/api/org/webrtc/TextureBufferImpl.java
+++ b/sdk/android/api/org/webrtc/TextureBufferImpl.java
@@ -136,8 +136,8 @@ public class TextureBufferImpl implements VideoFrame.TextureBuffer {
cropAndScaleMatrix.preScale(cropWidth / (float) width, cropHeight / (float) height);
#include <errno.h>
#include <fcntl.h>
+#include <poll.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index 77968b7eaf..f87fe53abe 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -12,6 +12,7 @@
return applyTransformMatrix(cropAndScaleMatrix,
- (int) Math.round(unscaledWidth * cropWidth / (float) width),
- (int) Math.round(unscaledHeight * cropHeight / (float) height), scaleWidth, scaleHeight);
+ Math.round(unscaledWidth * cropWidth / (float) width),
+ Math.round(unscaledHeight * cropHeight / (float) height), scaleWidth, scaleHeight);
}
#include <errno.h>
#include <fcntl.h>
+#include <poll.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
@@ -73,16 +74,13 @@ void DeviceInfoV4l2::HandleEvent(inotify_event* event, int fd)
int DeviceInfoV4l2::EventCheck(int fd)
{
- struct timeval timeout;
- fd_set rfds;
+ struct pollfd fds = {
+ .fd = fd,
+ .events = POLLIN,
+ .revents = 0
+ };
- timeout.tv_sec = 0;
- timeout.tv_usec = 100000;
-
- FD_ZERO(&rfds);
- FD_SET(fd, &rfds);
-
- return select(fd+1, &rfds, NULL, NULL, &timeout);
+ return poll(&fds, 1, 100);
}
int DeviceInfoV4l2::HandleEvents(int fd)
diff --git a/modules/video_capture/linux/video_capture_linux.cc b/modules/video_capture/linux/video_capture_linux.cc
index 4895a1ab71..f3324a8e68 100644
--- a/modules/video_capture/linux/video_capture_linux.cc
+++ b/modules/video_capture/linux/video_capture_linux.cc
@@ -10,6 +10,7 @@
#include <errno.h>
#include <fcntl.h>
+#include <poll.h>
#include <stdio.h>
#include <string.h>
#include <sys/ioctl.h>
diff --git a/modules/video_capture/linux/video_capture_v4l2.cc b/modules/video_capture/linux/video_capture_v4l2.cc
index 1dc13b01aa..b527a331e4 100644
--- a/modules/video_capture/linux/video_capture_v4l2.cc
+++ b/modules/video_capture/linux/video_capture_v4l2.cc
@@ -12,7 +12,7 @@
#include <errno.h>
#include <fcntl.h>
-#include <linux/videodev2.h>
+#include <poll.h>
#include <stdio.h>
#include <string.h>
#include <sys/ioctl.h>
@@ -20,6 +20,14 @@
#include <sys/select.h>
#include <time.h>
#include <unistd.h>
+// v4l includes
+#if defined(__NetBSD__) || defined(__OpenBSD__) // WEBRTC_BSD
+#include <sys/videoio.h>
+#elif defined(__sun)
+#include <sys/videodev2.h>
+#else
+#include <linux/videodev2.h>
+#endif
#include <new>
#include <string>
@@ -359,16 +367,13 @@ bool VideoCaptureModuleV4L2::CaptureStarted() {
bool VideoCaptureModuleV4L2::CaptureProcess() {
int retVal = 0;
- fd_set rSet;
- struct timeval timeout;
+ struct pollfd rSet;
- FD_ZERO(&rSet);
- FD_SET(_deviceFd, &rSet);
- timeout.tv_sec = 1;
- timeout.tv_usec = 0;
+ rSet.fd = _deviceFd;
+ rSet.events = POLLIN;
+ rSet.revents = 0;
- // _deviceFd written only in StartCapture, when this thread isn't running.
- retVal = select(_deviceFd + 1, &rSet, NULL, NULL, &timeout);
+ retVal = poll(&rSet, 1, 1000);
{
MutexLock lock(&capture_lock_);
@@ -378,12 +383,12 @@ bool VideoCaptureModuleV4L2::CaptureProcess() {
}
if (retVal < 0 && errno != EINTR) { // continue if interrupted
- // select failed
+ // poll failed
return false;
} else if (retVal == 0) {
- // select timed out
+ // poll timed out
return true;
- } else if (!FD_ISSET(_deviceFd, &rSet)) {
+ } else if (!(rSet.revents & POLLIN)) {
// not event on camera handle
return true;
}
/**
--
2.34.1

53
third_party/libwebrtc/moz-patch-stack/0029.patch поставляемый
Просмотреть файл

@ -1,28 +1,43 @@
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 11 Feb 2020 17:04:00 +0000
Subject: Bug 1578073 - Fix warning: [cast] redundant cast to int; r=ng
Date: Tue, 11 Feb 2020 17:07:00 +0000
Subject: Bug 1578073 - Remove native calls in Histogram.java; r=ng
Differential Revision: https://phabricator.services.mozilla.com/D61852
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/82877c8a864072c03f45234c6649645ed609e098
Getting the JNI calls here to work requires a good amount of webrtc.org
machinery. It might be worth setting that up the next time we do an upstream
merge, but for now, it is a lot simpler to just remove the affected code,
given that we are not interested in collecting this data anyway.
Differential Revision: https://phabricator.services.mozilla.com/D61860
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7a9b07dec9f9d435416b06829fa5063aca3a476e
---
sdk/android/api/org/webrtc/TextureBufferImpl.java | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
sdk/android/src/java/org/webrtc/Histogram.java | 9 ++-------
1 file changed, 2 insertions(+), 7 deletions(-)
diff --git a/sdk/android/api/org/webrtc/TextureBufferImpl.java b/sdk/android/api/org/webrtc/TextureBufferImpl.java
index c08fc4c29b..8e0e40ef70 100644
--- a/sdk/android/api/org/webrtc/TextureBufferImpl.java
+++ b/sdk/android/api/org/webrtc/TextureBufferImpl.java
@@ -136,8 +136,8 @@ public class TextureBufferImpl implements VideoFrame.TextureBuffer {
cropAndScaleMatrix.preScale(cropWidth / (float) width, cropHeight / (float) height);
return applyTransformMatrix(cropAndScaleMatrix,
- (int) Math.round(unscaledWidth * cropWidth / (float) width),
- (int) Math.round(unscaledHeight * cropHeight / (float) height), scaleWidth, scaleHeight);
+ Math.round(unscaledWidth * cropWidth / (float) width),
+ Math.round(unscaledHeight * cropHeight / (float) height), scaleWidth, scaleHeight);
diff --git a/sdk/android/src/java/org/webrtc/Histogram.java b/sdk/android/src/java/org/webrtc/Histogram.java
index 877986134a..c1d2d61a71 100644
--- a/sdk/android/src/java/org/webrtc/Histogram.java
+++ b/sdk/android/src/java/org/webrtc/Histogram.java
@@ -27,18 +27,13 @@ class Histogram {
}
/**
static public Histogram createCounts(String name, int min, int max, int bucketCount) {
- return new Histogram(nativeCreateCounts(name, min, max, bucketCount));
+ return new Histogram(0);
}
static public Histogram createEnumeration(String name, int max) {
- return new Histogram(nativeCreateEnumeration(name, max));
+ return new Histogram(0);
}
public void addSample(int sample) {
- nativeAddSample(handle, sample);
}
-
- private static native long nativeCreateCounts(String name, int min, int max, int bucketCount);
- private static native long nativeCreateEnumeration(String name, int max);
- private static native void nativeAddSample(long handle, int sample);
}
--
2.34.1

57
third_party/libwebrtc/moz-patch-stack/0030.patch поставляемый
Просмотреть файл

@ -1,43 +1,36 @@
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 11 Feb 2020 17:07:00 +0000
Subject: Bug 1578073 - Remove native calls in Histogram.java; r=ng
Date: Wed, 12 Feb 2020 17:19:00 +0000
Subject: Bug 1578073 - Suppress MissingPermission lint in Camera2Session;
r=snorp
Getting the JNI calls here to work requires a good amount of webrtc.org
machinery. It might be worth setting that up the next time we do an upstream
merge, but for now, it is a lot simpler to just remove the affected code,
given that we are not interested in collecting this data anyway.
Depends on D61861
Differential Revision: https://phabricator.services.mozilla.com/D61860
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7a9b07dec9f9d435416b06829fa5063aca3a476e
Differential Revision: https://phabricator.services.mozilla.com/D62457
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/722c4a6d1dad867f9ce47fe96d71b5dedb4cbaa8
---
sdk/android/src/java/org/webrtc/Histogram.java | 9 ++-------
1 file changed, 2 insertions(+), 7 deletions(-)
sdk/android/src/java/org/webrtc/Camera2Session.java | 2 ++
1 file changed, 2 insertions(+)
diff --git a/sdk/android/src/java/org/webrtc/Histogram.java b/sdk/android/src/java/org/webrtc/Histogram.java
index 877986134a..c1d2d61a71 100644
--- a/sdk/android/src/java/org/webrtc/Histogram.java
+++ b/sdk/android/src/java/org/webrtc/Histogram.java
@@ -27,18 +27,13 @@ class Histogram {
diff --git a/sdk/android/src/java/org/webrtc/Camera2Session.java b/sdk/android/src/java/org/webrtc/Camera2Session.java
index dec97a2c25..d5ee80c73e 100644
--- a/sdk/android/src/java/org/webrtc/Camera2Session.java
+++ b/sdk/android/src/java/org/webrtc/Camera2Session.java
@@ -10,6 +10,7 @@
package org.webrtc;
+import android.annotation.SuppressLint;
import android.content.Context;
import android.hardware.camera2.CameraAccessException;
import android.hardware.camera2.CameraCaptureSession;
@@ -347,6 +348,7 @@ class Camera2Session implements CameraSession {
Logging.d(TAG, "Using capture format: " + captureFormat);
}
static public Histogram createCounts(String name, int min, int max, int bucketCount) {
- return new Histogram(nativeCreateCounts(name, min, max, bucketCount));
+ return new Histogram(0);
}
+ @SuppressLint("MissingPermission")
private void openCamera() {
checkIsOnCameraThread();
static public Histogram createEnumeration(String name, int max) {
- return new Histogram(nativeCreateEnumeration(name, max));
+ return new Histogram(0);
}
public void addSample(int sample) {
- nativeAddSample(handle, sample);
}
-
- private static native long nativeCreateCounts(String name, int min, int max, int bucketCount);
- private static native long nativeCreateEnumeration(String name, int max);
- private static native void nativeAddSample(long handle, int sample);
}
--
2.34.1

102
third_party/libwebrtc/moz-patch-stack/0031.patch поставляемый
Просмотреть файл

@ -1,36 +1,90 @@
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 12 Feb 2020 17:19:00 +0000
Subject: Bug 1578073 - Suppress MissingPermission lint in Camera2Session;
r=snorp
Date: Wed, 8 Jul 2020 17:35:00 +0000
Subject: Bug 1650572 - Check V4L2_CAP_DEVICE_CAPS before accessing
device_caps; r=ng
Depends on D61861
The capabilities field is for the physical device, device_caps is for the
specific /dev/videoX device that has been opened. The device_caps field is
only populated if V4L2_CAP_DEVICE_CAPS is set, so we should check that, and
fall back to capabilities if it is not set.
Differential Revision: https://phabricator.services.mozilla.com/D62457
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/722c4a6d1dad867f9ce47fe96d71b5dedb4cbaa8
Differential Revision: https://phabricator.services.mozilla.com/D82377
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b5acbf536c46a66c939a61bde34ad93b1977a604
---
sdk/android/src/java/org/webrtc/Camera2Session.java | 2 ++
1 file changed, 2 insertions(+)
modules/video_capture/linux/device_info_v4l2.cc | 17 ++++++++++++-----
modules/video_capture/linux/device_info_v4l2.h | 3 +++
2 files changed, 15 insertions(+), 5 deletions(-)
diff --git a/sdk/android/src/java/org/webrtc/Camera2Session.java b/sdk/android/src/java/org/webrtc/Camera2Session.java
index dec97a2c25..d5ee80c73e 100644
--- a/sdk/android/src/java/org/webrtc/Camera2Session.java
+++ b/sdk/android/src/java/org/webrtc/Camera2Session.java
@@ -10,6 +10,7 @@
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index f87fe53abe..d506f3a448 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -202,8 +202,7 @@ uint32_t DeviceInfoV4l2::NumberOfDevices() {
snprintf(device, sizeof(device), "/dev/video%d", n);
if ((fd = open(device, O_RDONLY)) != -1) {
// query device capabilities and make sure this is a video capture device
- if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 ||
- !(cap.device_caps & V4L2_CAP_VIDEO_CAPTURE)) {
+ if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 || !IsVideoCaptureDevice(&cap)) {
close(fd);
continue;
}
@@ -235,8 +234,7 @@ int32_t DeviceInfoV4l2::GetDeviceName(uint32_t deviceNumber,
sprintf(device, "/dev/video%d", device_index);
if ((fd = open(device, O_RDONLY)) != -1) {
// query device capabilities and make sure this is a video capture device
- if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 ||
- !(cap.device_caps & V4L2_CAP_VIDEO_CAPTURE)) {
+ if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 || !IsVideoCaptureDevice(&cap)) {
close(fd);
continue;
}
@@ -321,7 +319,7 @@ int32_t DeviceInfoV4l2::CreateCapabilityMap(const char* deviceUniqueIdUTF8) {
struct v4l2_capability cap;
if (ioctl(fd, VIDIOC_QUERYCAP, &cap) == 0) {
// skip devices without video capture capability
- if (!(cap.device_caps & V4L2_CAP_VIDEO_CAPTURE)) {
+ if (!IsVideoCaptureDevice(&cap)) {
continue;
}
package org.webrtc;
@@ -383,6 +381,15 @@ bool DeviceInfoV4l2::IsDeviceNameMatches(const char* name,
return false;
}
+import android.annotation.SuppressLint;
import android.content.Context;
import android.hardware.camera2.CameraAccessException;
import android.hardware.camera2.CameraCaptureSession;
@@ -347,6 +348,7 @@ class Camera2Session implements CameraSession {
Logging.d(TAG, "Using capture format: " + captureFormat);
}
+bool DeviceInfoV4l2::IsVideoCaptureDevice(struct v4l2_capability* cap)
+{
+ if (cap->capabilities & V4L2_CAP_DEVICE_CAPS) {
+ return cap->device_caps & V4L2_CAP_VIDEO_CAPTURE;
+ } else {
+ return cap->capabilities & V4L2_CAP_VIDEO_CAPTURE;
+ }
+}
+
int32_t DeviceInfoV4l2::FillCapabilities(int fd) {
// set image format
struct v4l2_format video_fmt;
diff --git a/modules/video_capture/linux/device_info_v4l2.h b/modules/video_capture/linux/device_info_v4l2.h
index 95432a509d..e3c2395f49 100644
--- a/modules/video_capture/linux/device_info_v4l2.h
+++ b/modules/video_capture/linux/device_info_v4l2.h
@@ -18,6 +18,8 @@
#include "rtc_base/platform_thread.h"
#include <sys/inotify.h>
+ @SuppressLint("MissingPermission")
private void openCamera() {
checkIsOnCameraThread();
+struct v4l2_capability;
+
namespace webrtc {
namespace videocapturemodule {
class DeviceInfoV4l2 : public DeviceInfoImpl {
@@ -49,6 +51,7 @@ class DeviceInfoV4l2 : public DeviceInfoImpl {
private:
bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
+ bool IsVideoCaptureDevice(struct v4l2_capability* cap);
#ifdef WEBRTC_LINUX
void HandleEvent(inotify_event* event, int fd);
--
2.34.1

124
third_party/libwebrtc/moz-patch-stack/0032.patch поставляемый
Просмотреть файл

@ -1,90 +1,66 @@
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 8 Jul 2020 17:35:00 +0000
Subject: Bug 1650572 - Check V4L2_CAP_DEVICE_CAPS before accessing
device_caps; r=ng
From: Randell Jesup <rjesup@wgate.com>
Date: Sat, 11 Jul 2020 12:31:00 +0000
Subject: Bug 1112392 - Move webrtc Tab Sharing to work in e10s/fission
r=dminor
The capabilities field is for the physical device, device_caps is for the
specific /dev/videoX device that has been opened. The device_caps field is
only populated if V4L2_CAP_DEVICE_CAPS is set, so we should check that, and
fall back to capabilities if it is not set.
Also we drop support for an independent-of-scroll/viewport capture, which
the old Tab Sharing supported, for security reasons (and we don't need it).
Differential Revision: https://phabricator.services.mozilla.com/D82377
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b5acbf536c46a66c939a61bde34ad93b1977a604
Differential Revision: https://phabricator.services.mozilla.com/D80974
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d57a030e6e3ae9ff56f14e8cc732b0e1d3869858
---
modules/video_capture/linux/device_info_v4l2.cc | 17 ++++++++++++-----
modules/video_capture/linux/device_info_v4l2.h | 3 +++
2 files changed, 15 insertions(+), 5 deletions(-)
modules/desktop_capture/desktop_capturer.cc | 11 +++++++++++
modules/desktop_capture/desktop_capturer.h | 8 ++++++++
2 files changed, 19 insertions(+)
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index f87fe53abe..d506f3a448 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -202,8 +202,7 @@ uint32_t DeviceInfoV4l2::NumberOfDevices() {
snprintf(device, sizeof(device), "/dev/video%d", n);
if ((fd = open(device, O_RDONLY)) != -1) {
// query device capabilities and make sure this is a video capture device
- if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 ||
- !(cap.device_caps & V4L2_CAP_VIDEO_CAPTURE)) {
+ if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 || !IsVideoCaptureDevice(&cap)) {
close(fd);
continue;
}
@@ -235,8 +234,7 @@ int32_t DeviceInfoV4l2::GetDeviceName(uint32_t deviceNumber,
sprintf(device, "/dev/video%d", device_index);
if ((fd = open(device, O_RDONLY)) != -1) {
// query device capabilities and make sure this is a video capture device
- if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 ||
- !(cap.device_caps & V4L2_CAP_VIDEO_CAPTURE)) {
+ if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0 || !IsVideoCaptureDevice(&cap)) {
close(fd);
continue;
}
@@ -321,7 +319,7 @@ int32_t DeviceInfoV4l2::CreateCapabilityMap(const char* deviceUniqueIdUTF8) {
struct v4l2_capability cap;
if (ioctl(fd, VIDIOC_QUERYCAP, &cap) == 0) {
// skip devices without video capture capability
- if (!(cap.device_caps & V4L2_CAP_VIDEO_CAPTURE)) {
+ if (!IsVideoCaptureDevice(&cap)) {
continue;
}
@@ -383,6 +381,15 @@ bool DeviceInfoV4l2::IsDeviceNameMatches(const char* name,
return false;
diff --git a/modules/desktop_capture/desktop_capturer.cc b/modules/desktop_capture/desktop_capturer.cc
index dc24b387d5..4debc75394 100644
--- a/modules/desktop_capture/desktop_capturer.cc
+++ b/modules/desktop_capture/desktop_capturer.cc
@@ -95,6 +95,17 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
return capturer;
}
+bool DeviceInfoV4l2::IsVideoCaptureDevice(struct v4l2_capability* cap)
+{
+ if (cap->capabilities & V4L2_CAP_DEVICE_CAPS) {
+ return cap->device_caps & V4L2_CAP_VIDEO_CAPTURE;
+ } else {
+ return cap->capabilities & V4L2_CAP_VIDEO_CAPTURE;
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateTabCapturer(
+ const DesktopCaptureOptions& options) {
+ std::unique_ptr<DesktopCapturer> capturer = CreateRawTabCapturer(options);
+ if (capturer && options.detect_updated_region()) {
+ capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
+ }
+
+ return capturer;
+}
+
int32_t DeviceInfoV4l2::FillCapabilities(int fd) {
// set image format
struct v4l2_format video_fmt;
diff --git a/modules/video_capture/linux/device_info_v4l2.h b/modules/video_capture/linux/device_info_v4l2.h
index 95432a509d..e3c2395f49 100644
--- a/modules/video_capture/linux/device_info_v4l2.h
+++ b/modules/video_capture/linux/device_info_v4l2.h
@@ -18,6 +18,8 @@
#include "rtc_base/platform_thread.h"
#include <sys/inotify.h>
#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
bool DesktopCapturer::IsRunningUnderWayland() {
const char* xdg_session_type = getenv("XDG_SESSION_TYPE");
diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
index 558217badd..3fdcd2c5d0 100644
--- a/modules/desktop_capture/desktop_capturer.h
+++ b/modules/desktop_capture/desktop_capturer.h
@@ -176,6 +176,10 @@ class RTC_EXPORT DesktopCapturer {
static std::unique_ptr<DesktopCapturer> CreateScreenCapturer(
const DesktopCaptureOptions& options);
+struct v4l2_capability;
+ // Creates a DesktopCapturer instance which targets to capture tab.
+ static std::unique_ptr<DesktopCapturer> CreateTabCapturer(
+ const DesktopCaptureOptions& options);
+
namespace webrtc {
namespace videocapturemodule {
class DeviceInfoV4l2 : public DeviceInfoImpl {
@@ -49,6 +51,7 @@ class DeviceInfoV4l2 : public DeviceInfoImpl {
#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
static bool IsRunningUnderWayland();
private:
bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
+ bool IsVideoCaptureDevice(struct v4l2_capability* cap);
@@ -201,6 +205,10 @@ class RTC_EXPORT DesktopCapturer {
// capture screens.
static std::unique_ptr<DesktopCapturer> CreateRawScreenCapturer(
const DesktopCaptureOptions& options);
+
+ // Creates a DesktopCapturer instance which targets to capture tabs
+ static std::unique_ptr<DesktopCapturer> CreateRawTabCapturer(
+ const DesktopCaptureOptions& options);
};
#ifdef WEBRTC_LINUX
void HandleEvent(inotify_event* event, int fd);
} // namespace webrtc
--
2.34.1

90
third_party/libwebrtc/moz-patch-stack/0033.patch поставляемый
Просмотреть файл

@ -1,66 +1,40 @@
From: Randell Jesup <rjesup@wgate.com>
Date: Sat, 11 Jul 2020 12:31:00 +0000
Subject: Bug 1112392 - Move webrtc Tab Sharing to work in e10s/fission
r=dminor
From: James Willcox <snorp@snorp.net>
Date: Fri, 18 Sep 2020 22:29:00 +0000
Subject: Bug 1553459 - Migrate to AndroidX r=geckoview-reviewers,agi
Also we drop support for an independent-of-scroll/viewport capture, which
the old Tab Sharing supported, for security reasons (and we don't need it).
Differential Revision: https://phabricator.services.mozilla.com/D80974
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d57a030e6e3ae9ff56f14e8cc732b0e1d3869858
Differential Revision: https://phabricator.services.mozilla.com/D90711
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cf8917364050fc7d981b190cdc1db5ab57357f9b
---
modules/desktop_capture/desktop_capturer.cc | 11 +++++++++++
modules/desktop_capture/desktop_capturer.h | 8 ++++++++
2 files changed, 19 insertions(+)
sdk/android/src/java/org/webrtc/EglBase14Impl.java | 1 -
sdk/android/src/java/org/webrtc/GlGenericDrawer.java | 3 ---
2 files changed, 4 deletions(-)
diff --git a/modules/desktop_capture/desktop_capturer.cc b/modules/desktop_capture/desktop_capturer.cc
index dc24b387d5..4debc75394 100644
--- a/modules/desktop_capture/desktop_capturer.cc
+++ b/modules/desktop_capture/desktop_capturer.cc
@@ -95,6 +95,17 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
return capturer;
}
diff --git a/sdk/android/src/java/org/webrtc/EglBase14Impl.java b/sdk/android/src/java/org/webrtc/EglBase14Impl.java
index e53dda6e4c..caf45b091e 100644
--- a/sdk/android/src/java/org/webrtc/EglBase14Impl.java
+++ b/sdk/android/src/java/org/webrtc/EglBase14Impl.java
@@ -21,7 +21,6 @@ import android.opengl.GLException;
import android.os.Build;
import android.view.Surface;
import androidx.annotation.Nullable;
-import org.webrtc.EglBase;
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateTabCapturer(
+ const DesktopCaptureOptions& options) {
+ std::unique_ptr<DesktopCapturer> capturer = CreateRawTabCapturer(options);
+ if (capturer && options.detect_updated_region()) {
+ capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
+ }
+
+ return capturer;
+}
+
#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
bool DesktopCapturer::IsRunningUnderWayland() {
const char* xdg_session_type = getenv("XDG_SESSION_TYPE");
diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
index 558217badd..3fdcd2c5d0 100644
--- a/modules/desktop_capture/desktop_capturer.h
+++ b/modules/desktop_capture/desktop_capturer.h
@@ -176,6 +176,10 @@ class RTC_EXPORT DesktopCapturer {
static std::unique_ptr<DesktopCapturer> CreateScreenCapturer(
const DesktopCaptureOptions& options);
/**
* Holds EGL state and utility methods for handling an EGL14 EGLContext, an EGLDisplay,
diff --git a/sdk/android/src/java/org/webrtc/GlGenericDrawer.java b/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
index b70a3728b9..34144e2f75 100644
--- a/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
+++ b/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
@@ -14,9 +14,6 @@ import android.opengl.GLES11Ext;
import android.opengl.GLES20;
import androidx.annotation.Nullable;
import java.nio.FloatBuffer;
-import org.webrtc.GlShader;
-import org.webrtc.GlUtil;
-import org.webrtc.RendererCommon;
+ // Creates a DesktopCapturer instance which targets to capture tab.
+ static std::unique_ptr<DesktopCapturer> CreateTabCapturer(
+ const DesktopCaptureOptions& options);
+
#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
static bool IsRunningUnderWayland();
@@ -201,6 +205,10 @@ class RTC_EXPORT DesktopCapturer {
// capture screens.
static std::unique_ptr<DesktopCapturer> CreateRawScreenCapturer(
const DesktopCaptureOptions& options);
+
+ // Creates a DesktopCapturer instance which targets to capture tabs
+ static std::unique_ptr<DesktopCapturer> CreateRawTabCapturer(
+ const DesktopCaptureOptions& options);
};
} // namespace webrtc
/**
* Helper class to implement an instance of RendererCommon.GlDrawer that can accept multiple input
--
2.34.1

91
third_party/libwebrtc/moz-patch-stack/0034.patch поставляемый
Просмотреть файл

@ -1,40 +1,67 @@
From: James Willcox <snorp@snorp.net>
Date: Fri, 18 Sep 2020 22:29:00 +0000
Subject: Bug 1553459 - Migrate to AndroidX r=geckoview-reviewers,agi
From: Paul Adenot <paul@paul.cx>
Date: Wed, 4 Nov 2020 13:03:00 +0000
Subject: Bug 1675042 - Put IR camera last in the device selection list, so
that they are never the default. r=dminor
Differential Revision: https://phabricator.services.mozilla.com/D90711
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cf8917364050fc7d981b190cdc1db5ab57357f9b
Differential Revision: https://phabricator.services.mozilla.com/D95764
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/754900ff2a2b1c32878baf3c76d7c0e8219419ff
---
sdk/android/src/java/org/webrtc/EglBase14Impl.java | 1 -
sdk/android/src/java/org/webrtc/GlGenericDrawer.java | 3 ---
2 files changed, 4 deletions(-)
sdk/android/api/org/webrtc/Camera1Enumerator.java | 5 +++++
sdk/android/api/org/webrtc/Camera2Enumerator.java | 12 ++++++++++++
sdk/android/api/org/webrtc/CameraEnumerator.java | 1 +
3 files changed, 18 insertions(+)
diff --git a/sdk/android/src/java/org/webrtc/EglBase14Impl.java b/sdk/android/src/java/org/webrtc/EglBase14Impl.java
index e53dda6e4c..caf45b091e 100644
--- a/sdk/android/src/java/org/webrtc/EglBase14Impl.java
+++ b/sdk/android/src/java/org/webrtc/EglBase14Impl.java
@@ -21,7 +21,6 @@ import android.opengl.GLException;
import android.os.Build;
import android.view.Surface;
import androidx.annotation.Nullable;
-import org.webrtc.EglBase;
diff --git a/sdk/android/api/org/webrtc/Camera1Enumerator.java b/sdk/android/api/org/webrtc/Camera1Enumerator.java
index fb1a21f323..4a1aacdb05 100644
--- a/sdk/android/api/org/webrtc/Camera1Enumerator.java
+++ b/sdk/android/api/org/webrtc/Camera1Enumerator.java
@@ -63,6 +63,11 @@ public class Camera1Enumerator implements CameraEnumerator {
return info != null && info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_BACK;
}
/**
* Holds EGL state and utility methods for handling an EGL14 EGLContext, an EGLDisplay,
diff --git a/sdk/android/src/java/org/webrtc/GlGenericDrawer.java b/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
index b70a3728b9..34144e2f75 100644
--- a/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
+++ b/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
@@ -14,9 +14,6 @@ import android.opengl.GLES11Ext;
import android.opengl.GLES20;
import androidx.annotation.Nullable;
import java.nio.FloatBuffer;
-import org.webrtc.GlShader;
-import org.webrtc.GlUtil;
-import org.webrtc.RendererCommon;
+ @Override
+ public boolean isInfrared(String deviceName) {
+ return false;
+ }
+
@Override
public List<CaptureFormat> getSupportedFormats(String deviceName) {
return getSupportedFormats(getCameraIndex(deviceName));
diff --git a/sdk/android/api/org/webrtc/Camera2Enumerator.java b/sdk/android/api/org/webrtc/Camera2Enumerator.java
index 456d8cd060..44e239ad8e 100644
--- a/sdk/android/api/org/webrtc/Camera2Enumerator.java
+++ b/sdk/android/api/org/webrtc/Camera2Enumerator.java
@@ -74,6 +74,18 @@ public class Camera2Enumerator implements CameraEnumerator {
== CameraMetadata.LENS_FACING_BACK;
}
/**
* Helper class to implement an instance of RendererCommon.GlDrawer that can accept multiple input
+ @Override
+ public boolean isInfrared(String deviceName) {
+ CameraCharacteristics characteristics = getCameraCharacteristics(deviceName);
+
+ if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.Q) {
+ Integer colors = characteristics.get(CameraCharacteristics.SENSOR_INFO_COLOR_FILTER_ARRANGEMENT);
+ return colors != null && colors.equals(CameraCharacteristics.SENSOR_INFO_COLOR_FILTER_ARRANGEMENT_NIR);
+ }
+
+ return false;
+ }
+
@Nullable
@Override
public List<CaptureFormat> getSupportedFormats(String deviceName) {
diff --git a/sdk/android/api/org/webrtc/CameraEnumerator.java b/sdk/android/api/org/webrtc/CameraEnumerator.java
index dc954b62e0..db34d542c8 100644
--- a/sdk/android/api/org/webrtc/CameraEnumerator.java
+++ b/sdk/android/api/org/webrtc/CameraEnumerator.java
@@ -18,6 +18,7 @@ public interface CameraEnumerator {
public String[] getDeviceNames();
public boolean isFrontFacing(String deviceName);
public boolean isBackFacing(String deviceName);
+ public boolean isInfrared(String deviceName);
public List<CaptureFormat> getSupportedFormats(String deviceName);
public CameraVideoCapturer createCapturer(
--
2.34.1

1148
third_party/libwebrtc/moz-patch-stack/0035.patch поставляемый
Просмотреть файл

Разница между файлами не показана из-за своего большого размера Загрузить разницу

1833
third_party/libwebrtc/moz-patch-stack/0036.patch поставляемый
Просмотреть файл

Разница между файлами не показана из-за своего большого размера Загрузить разницу

787
third_party/libwebrtc/moz-patch-stack/0037.patch поставляемый
Просмотреть файл

@ -1,778 +1,29 @@
From: stransky <stransky@redhat.com>
Date: Thu, 5 Nov 2020 07:47:00 +0000
Subject: Bug 1654112 - Tweak upstream gn files for Firefox build. r=ng
From: Dan Minor <dminor@mozilla.com>
Date: Fri, 13 Nov 2020 14:34:00 -0500
Subject: Bug 1654112 - Fully quality AudioLevel::kUri in channel_send.cc. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D130075
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/127ace4d8887f11abb201d300a849772a2b519f8
Differential Revision: https://phabricator.services.mozilla.com/D130082
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7163801a480d607005042292ed9e4fbb892f440d
---
.gn | 2 ++
BUILD.gn | 26 +++++++++++++-
api/BUILD.gn | 8 ++++-
api/task_queue/BUILD.gn | 2 ++
api/transport/BUILD.gn | 2 ++
common_audio/BUILD.gn | 4 ---
common_audio/fir_filter_avx2.cc | 2 ++
common_audio/intrin.h | 8 +++++
media/BUILD.gn | 30 ++++++++++++++++
modules/audio_coding/BUILD.gn | 2 +-
modules/audio_device/BUILD.gn | 8 ++---
modules/audio_processing/aec3/BUILD.gn | 13 +++----
.../aec3/adaptive_fir_filter_avx2.cc | 3 +-
.../audio_processing/agc2/rnn_vad/BUILD.gn | 2 +-
modules/desktop_capture/BUILD.gn | 29 +---------------
modules/portal/BUILD.gn | 24 +++++++++++++
modules/video_capture/BUILD.gn | 11 +-----
modules/video_coding/BUILD.gn | 6 ++++
rtc_base/BUILD.gn | 4 ++-
rtc_base/system/BUILD.gn | 2 +-
test/BUILD.gn | 14 ++++++++
webrtc.gni | 34 ++++++++++++-------
22 files changed, 162 insertions(+), 74 deletions(-)
create mode 100644 common_audio/intrin.h
audio/channel_send.cc | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/.gn b/.gn
index c9824916ad..d35ea79ced 100644
--- a/.gn
+++ b/.gn
@@ -69,6 +69,8 @@ default_args = {
# Prevent jsoncpp to pass -Wno-deprecated-declarations to users
jsoncpp_no_deprecated_declarations = false
+ use_custom_libcxx = false
+
# Fixes the abi-revision issue.
# TODO(https://bugs.webrtc.org/14437): Remove this section if general
# Chromium fix resolves the problem.
diff --git a/BUILD.gn b/BUILD.gn
index c41059f288..02e9d1d4a0 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -33,7 +33,7 @@ if (is_android) {
import("//build/config/android/rules.gni")
}
-if (!build_with_chromium) {
+if (!build_with_chromium && !build_with_mozilla) {
# This target should (transitively) cause everything to be built; if you run
# 'ninja default' and then 'ninja all', the second build should do no work.
group("default") {
@@ -152,6 +152,10 @@ config("common_inherited_config") {
defines += [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ]
}
+ if (build_with_mozilla) {
+ defines += [ "WEBRTC_MOZILLA_BUILD" ]
+ }
+
if (!rtc_builtin_ssl_root_certificates) {
defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ]
}
@@ -460,9 +464,11 @@ config("common_config") {
}
}
+if (is_mac) {
config("common_objc") {
frameworks = [ "Foundation.framework" ]
}
+}
if (!build_with_chromium) {
# Target to build all the WebRTC production code.
@@ -502,6 +508,23 @@ if (!build_with_chromium) {
"sdk",
"video",
]
+ if (build_with_mozilla) {
+ deps -= [
+ "api:create_peerconnection_factory",
+ "api:rtc_error",
+ "api:transport_api",
+ "api/crypto",
+ "api/rtc_event_log:rtc_event_log_factory",
+ "api/task_queue",
+ "api/task_queue:default_task_queue_factory",
+ "api/test/metrics",
+ "logging:rtc_event_log_api",
+ "p2p:rtc_p2p",
+ "pc:libjingle_peerconnection",
+ "pc:rtc_pc",
+ "sdk",
+ ]
+ }
if (rtc_include_builtin_audio_codecs) {
deps += [
@@ -521,6 +544,7 @@ if (!build_with_chromium) {
deps += [
"api/video:video_frame",
"api/video:video_rtp_headers",
+ "test:rtp_test_utils",
]
} else {
deps += [
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 33a6b0aaa6..ab5d6c91ce 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -35,7 +35,7 @@ rtc_source_set("callfactory_api") {
]
}
-if (!build_with_chromium) {
+if (!build_with_chromium && !build_with_mozilla) {
rtc_library("create_peerconnection_factory") {
visibility = [ "*" ]
allow_poison = [ "default_task_queue" ]
@@ -227,6 +227,7 @@ rtc_library("rtp_sender_interface") {
}
rtc_library("libjingle_peerconnection_api") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
cflags = []
sources = [
@@ -343,6 +344,7 @@ rtc_library("libjingle_peerconnection_api") {
"//third_party/abseil-cpp/absl/types:optional",
]
}
+}
rtc_source_set("frame_transformer_interface") {
visibility = [ "*" ]
@@ -550,6 +552,7 @@ rtc_source_set("peer_network_dependencies") {
}
rtc_source_set("peer_connection_quality_test_fixture_api") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
testonly = true
sources = [ "test/peerconnection_quality_test_fixture.h" ]
@@ -600,6 +603,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
"//third_party/abseil-cpp/absl/types:optional",
]
}
+}
rtc_source_set("frame_generator_api") {
visibility = [ "*" ]
@@ -885,6 +889,7 @@ rtc_source_set("refcountedbase") {
]
}
+if (!build_with_mozilla) {
rtc_library("ice_transport_factory") {
visibility = [ "*" ]
sources = [
@@ -903,6 +908,7 @@ rtc_library("ice_transport_factory") {
"rtc_event_log:rtc_event_log",
]
}
+}
rtc_library("neteq_simulator_api") {
visibility = [ "*" ]
diff --git a/api/task_queue/BUILD.gn b/api/task_queue/BUILD.gn
index 69393b80ff..c9b4a5d0ec 100644
--- a/api/task_queue/BUILD.gn
+++ b/api/task_queue/BUILD.gn
@@ -30,6 +30,7 @@ rtc_library("task_queue") {
]
}
+if (rtc_include_tests) {
rtc_library("task_queue_test") {
visibility = [ "*" ]
testonly = true
@@ -78,6 +79,7 @@ rtc_library("task_queue_test") {
]
}
}
+}
rtc_library("default_task_queue_factory") {
visibility = [ "*" ]
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 08f3d6d1d0..e0b31122b2 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -90,6 +90,7 @@ rtc_source_set("sctp_transport_factory_interface") {
}
rtc_source_set("stun_types") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"stun.cc",
@@ -110,6 +111,7 @@ rtc_source_set("stun_types") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
if (rtc_include_tests) {
rtc_source_set("test_feedback_generator_interface") {
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index 2ae6d32710..a45214f754 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -267,14 +267,10 @@ if (current_cpu == "x86" || current_cpu == "x64") {
"resampler/sinc_resampler_avx2.cc",
]
- if (is_win) {
- cflags = [ "/arch:AVX2" ]
- } else {
cflags = [
"-mavx2",
"-mfma",
]
- }
deps = [
":fir_filter",
diff --git a/common_audio/fir_filter_avx2.cc b/common_audio/fir_filter_avx2.cc
index 9cb0f770ca..0031392f8a 100644
--- a/common_audio/fir_filter_avx2.cc
+++ b/common_audio/fir_filter_avx2.cc
@@ -15,6 +15,8 @@
#include <string.h>
#include <xmmintrin.h>
+#include "common_audio/intrin.h"
+
#include "rtc_base/checks.h"
#include "rtc_base/memory/aligned_malloc.h"
diff --git a/common_audio/intrin.h b/common_audio/intrin.h
new file mode 100644
index 0000000000..f6ff7f218f
--- /dev/null
+++ b/common_audio/intrin.h
@@ -0,0 +1,8 @@
+#if defined (__SSE__)
+ #include <immintrin.h>
+ #if defined (__clang__)
+ #include <avxintrin.h>
+ #include <avx2intrin.h>
+ #include <fmaintrin.h>
+ #endif
+#endif
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 2e5c0bb872..4ddc8349a8 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -147,6 +147,27 @@ rtc_library("rtc_media_base") {
"base/video_source_base.cc",
"base/video_source_base.h",
]
+ if (build_with_mozilla) {
+ sources -= [
+ "base/adapted_video_track_source.cc",
+ "base/adapted_video_track_source.h",
+ "base/audio_source.h",
+ "base/delayable.h",
+ "base/media_channel.h",
+ "base/media_channel_impl.cc",
+ "base/media_channel_impl.h",
+ "base/media_engine.cc",
+ "base/media_engine.h",
+ "base/rid_description.cc",
+ "base/rid_description.h",
+ "base/rtp_utils.cc",
+ "base/rtp_utils.h",
+ "base/stream_params.cc",
+ "base/stream_params.h",
+ "base/turn_utils.cc",
+ "base/turn_utils.h",
+ ]
+ }
}
rtc_library("media_channel_impl") {
@@ -420,6 +441,9 @@ rtc_library("rtc_internal_video_codecs") {
"../system_wrappers:field_trial",
"../test:fake_video_codecs",
]
+ if (build_with_mozilla) {
+ deps -= [ "../test:fake_video_codecs" ]
+ }
if (enable_libaom) {
defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ]
@@ -449,6 +473,12 @@ rtc_library("rtc_internal_video_codecs") {
# targets depend on :rtc_encoder_simulcast_proxy directly.
"engine/encoder_simulcast_proxy.h",
]
+ if (build_with_mozilla) {
+ sources -= [
+ "engine/fake_video_codec_factory.cc",
+ "engine/fake_video_codec_factory.h",
+ ]
+ }
}
rtc_library("rtc_audio_video") {
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index dddc3edd83..eac0650a26 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -553,7 +553,7 @@ rtc_library("webrtc_opus_wrapper") {
deps += [ rtc_opus_dir ]
public_configs = [ "//third_party/opus:opus_config" ]
} else if (build_with_mozilla) {
- include_dirs = [ getenv("DIST") + "/include/opus" ]
+ public_configs = [ "//third_party/opus:opus_config" ]
}
}
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index 4a6a0ab41c..e35a442025 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -233,9 +233,9 @@ rtc_library("audio_device_impl") {
]
if (build_with_mozilla) {
- sources += [
- "opensl/single_rw_fifo.cc",
- "opensl/single_rw_fifo.h",
+ sources -= [
+ "include/test_audio_device.cc",
+ "include/test_audio_device.h",
]
}
@@ -477,7 +477,7 @@ if (rtc_include_tests && !build_with_chromium) {
}
}
-if (!build_with_chromium && is_android) {
+if ((!build_with_chromium && !build_with_mozilla) && is_android) {
rtc_android_library("audio_device_java") {
sources = [
"android/java/src/org/webrtc/voiceengine/BuildInfo.java",
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index f5eb5d5951..3e11a245a1 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -264,14 +264,11 @@ if (current_cpu == "x86" || current_cpu == "x64") {
"vector_math_avx2.cc",
]
- if (is_win) {
- cflags = [ "/arch:AVX2" ]
- } else {
- cflags = [
- "-mavx2",
- "-mfma",
- ]
- }
+ cflags = [
+ "-mavx",
+ "-mavx2",
+ "-mfma",
+ ]
deps = [
":adaptive_fir_filter",
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
index 6c8c948026..44d4514275 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
@@ -10,8 +10,7 @@
#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
-#include <immintrin.h>
-
+#include "common_audio/intrin.h"
#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 9093a68cf3..3003a585bd 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -122,7 +122,7 @@ rtc_source_set("vector_math") {
if (current_cpu == "x86" || current_cpu == "x64") {
rtc_library("vector_math_avx2") {
sources = [ "vector_math_avx2.cc" ]
- if (is_win) {
+ if (is_win && !build_with_mozilla) {
cflags = [ "/arch:AVX2" ]
} else {
cflags = [
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index c9993dd47b..4aa9186527 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -342,37 +342,12 @@ rtc_library("desktop_capture") {
]
deps += [ ":desktop_capture_objc" ]
}
-
- if (build_with_mozilla) {
- sources += [
- "desktop_device_info.cc",
- "desktop_device_info.h",
- ]
- if (is_win) {
- sources += [
- "app_capturer_win.cc",
- "win/desktop_device_info_win.cc",
- "win/win_shared.cc",
- ]
- }
- }
if (rtc_use_x11_extensions || rtc_use_pipewire) {
sources += [
"mouse_cursor_monitor_linux.cc",
"screen_capturer_linux.cc",
"window_capturer_linux.cc",
]
-
- if (build_with_mozilla && (is_linux || is_chromeos)) {
- sources += [
- "app_capturer_linux.cc",
- "linux/x11/app_capturer_x11.cc",
- "linux/x11/desktop_device_info_linux.cc",
- "linux/x11/desktop_device_info_linux.h",
- "linux/x11/shared_x_util.cc",
- "linux/x11/shared_x_util.h",
- ]
- }
}
if (rtc_use_x11_extensions) {
@@ -536,9 +511,7 @@ rtc_library("desktop_capture") {
deps += [ "../../rtc_base:sanitizer" ]
}
- if (!build_with_mozilla) {
- deps += [ "//third_party/libyuv" ]
- }
+ deps += [ "//third_party/libyuv" ]
if (use_desktop_capture_differ_sse2) {
deps += [ ":desktop_capture_differ_sse2" ]
diff --git a/modules/portal/BUILD.gn b/modules/portal/BUILD.gn
index d0756f269b..d7768b2323 100644
--- a/modules/portal/BUILD.gn
+++ b/modules/portal/BUILD.gn
@@ -11,6 +11,7 @@ import("//tools/generate_stubs/rules.gni")
import("../../webrtc.gni")
if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+if (!build_with_mozilla) {
pkg_config("gio") {
packages = [
"gio-2.0",
@@ -88,6 +89,12 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) {
defines += [ "WEBRTC_USE_GIO" ]
}
}
+} else {
+ config("pipewire_all") {
+ }
+ config("pipewire_config") {
+ }
+}
rtc_library("portal") {
sources = [
@@ -120,5 +127,22 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) {
deps += [ ":pipewire_stubs" ]
}
+
+ if (build_with_mozilla) {
+ configs -= [
+ ":gio",
+ ":pipewire",
+ ":pipewire_config",
+ ]
+ deps -= [ ":pipewire_stubs" ]
+ defines -= [ "WEBRTC_DLOPEN_PIPEWIRE" ]
+ public_deps = [
+ "//third_party/pipewire",
+ "//third_party/drm",
+ "//third_party/gbm",
+ "//third_party/libepoxy"
+ ]
+ }
}
}
+
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index cfa5184d71..95548906c4 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -93,21 +93,12 @@ if (!build_with_chromium) {
"strmiids.lib",
"user32.lib",
]
-
- if (build_with_mozilla) {
- sources += [
- "windows/BaseFilter.cpp",
- "windows/BaseInputPin.cpp",
- "windows/BasePin.cpp",
- "windows/MediaType.cpp",
- ]
- }
}
if (is_fuchsia) {
sources = [ "video_capture_factory_null.cc" ]
}
- if (build_with_mozilla && is_android) {
+ if (!build_with_mozilla && is_android) {
include_dirs = [
"/config/external/nspr",
"/nsprpub/lib/ds",
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index fe63804b19..e1b5e4ba84 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -237,6 +237,12 @@ rtc_library("video_coding") {
"video_receiver2.cc",
"video_receiver2.h",
]
+ if (build_with_mozilla) {
+ sources += [
+ "event_wrapper.cc",
+ "event_wrapper.h",
+ ]
+ }
deps = [
":codec_globals_headers",
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 0d8bd4f759..7e162cecbb 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -820,7 +820,9 @@ rtc_library("rtc_json") {
"strings/json.h",
]
deps = [ ":stringutils" ]
+if (!build_with_mozilla) {
all_dependent_configs = [ "//third_party/jsoncpp:jsoncpp_config" ]
+}
if (rtc_build_json) {
deps += [ "//third_party/jsoncpp" ]
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 361880d68f..f39e57cc39 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -690,9 +690,9 @@ void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
include_audio_level_indication_.store(enable);
if (enable) {
- rtp_rtcp_->RegisterRtpHeaderExtension(AudioLevel::Uri(), id);
+ rtp_rtcp_->RegisterRtpHeaderExtension(webrtc::AudioLevel::Uri(), id);
} else {
@@ -2073,7 +2075,7 @@ if (rtc_include_tests) {
- rtp_rtcp_->DeregisterSendRtpHeaderExtension(AudioLevel::Uri());
+ rtp_rtcp_->DeregisterSendRtpHeaderExtension(webrtc::AudioLevel::Uri());
}
}
-if (is_android) {
+if (is_android && !build_with_mozilla) {
rtc_android_library("base_java") {
visibility = [ "*" ]
sources = [
diff --git a/rtc_base/system/BUILD.gn b/rtc_base/system/BUILD.gn
index 77f5139a2f..486b37590c 100644
--- a/rtc_base/system/BUILD.gn
+++ b/rtc_base/system/BUILD.gn
@@ -101,7 +101,7 @@ if (is_mac || is_ios) {
rtc_source_set("warn_current_thread_is_deadlocked") {
sources = [ "warn_current_thread_is_deadlocked.h" ]
deps = []
- if (is_android && !build_with_chromium) {
+ if (is_android && (!build_with_chromium && !build_with_mozilla)) {
sources += [ "warn_current_thread_is_deadlocked.cc" ]
deps += [
"..:logging",
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 0c71115f0d..04a718c411 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -156,6 +156,7 @@ rtc_library("audio_test_common") {
absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
}
+if (!build_with_mozilla) {
if (!build_with_chromium) {
if (is_mac || is_ios) {
rtc_library("video_test_mac") {
@@ -204,8 +205,12 @@ if (!build_with_chromium) {
}
}
}
+}
rtc_library("rtp_test_utils") {
+ if (build_with_mozilla) {
+ sources = []
+ } else {
testonly = true
sources = [
"rtcp_packet_parser.cc",
@@ -215,6 +220,7 @@ rtc_library("rtp_test_utils") {
"rtp_file_writer.cc",
"rtp_file_writer.h",
]
+ }
deps = [
"../api:array_view",
@@ -467,7 +473,9 @@ rtc_library("video_test_support") {
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
if (!is_ios) {
+ if (!build_with_mozilla) {
deps += [ "//third_party:jpeg" ]
+ }
sources += [ "testsupport/jpeg_frame_writer.cc" ]
} else {
sources += [ "testsupport/jpeg_frame_writer_ios.cc" ]
@@ -1036,6 +1044,10 @@ rtc_library("test_common") {
if (!is_android && !build_with_chromium) {
deps += [ "../modules/video_capture:video_capture_internal_impl" ]
}
+ # This, or some form of it should be upstreamed.
+ if (!rtc_include_tests) {
+ deps -= [ "../rtc_base:task_queue_for_test" ]
+ }
}
rtc_library("mock_transport") {
@@ -1203,6 +1215,7 @@ rtc_library("copy_to_file_audio_capturer_unittest") {
]
}
+if (!build_with_mozilla) {
if (!build_with_chromium && is_android) {
rtc_android_library("native_test_java") {
testonly = true
@@ -1216,6 +1229,7 @@ if (!build_with_chromium && is_android) {
]
}
}
+}
rtc_library("call_config_utils") {
# TODO(bugs.webrtc.org/10814): Remove rtc_json_suppressions as soon as it
diff --git a/webrtc.gni b/webrtc.gni
index 8dfcc9d244..6ae1b2329c 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -35,6 +35,11 @@ if (is_mac) {
import("//build/config/mac/rules.gni")
}
+if (is_android) {
+ import("//build/config/android/config.gni")
+ import("//build/config/android/rules.gni")
+}
+
if (is_fuchsia) {
import("//build/config/fuchsia/config.gni")
}
@@ -42,6 +47,11 @@ if (is_fuchsia) {
# This declare_args is separated from the next one because args declared
# in this one, can be read from the next one (args defined in the same
# declare_args cannot be referenced in that scope).
+declare_args() {
+ # Enable to use the Mozilla internal settings.
+ build_with_mozilla = true
+}
+
declare_args() {
# Setting this to true will make RTC_EXPORT (see rtc_base/system/rtc_export.h)
# expand to code that will manage symbols visibility.
@@ -84,7 +94,7 @@ declare_args() {
# will tell the pre-processor to remove the default definition of the
# SystemTimeNanos() which is defined in rtc_base/system_time.cc. In
# that case a new implementation needs to be provided.
- rtc_exclude_system_time = build_with_chromium
+ rtc_exclude_system_time = build_with_chromium || build_with_mozilla
# Setting this to false will require the API user to pass in their own
# SSLCertificateVerifier to verify the certificates presented from a
@@ -110,7 +120,7 @@ declare_args() {
# Used to specify an external OpenSSL include path when not compiling the
# library that comes with WebRTC (i.e. rtc_build_ssl == 0).
- rtc_ssl_root = ""
+ rtc_ssl_root = "unused"
# Enable when an external authentication mechanism is used for performing
# packet authentication for RTP packets instead of libsrtp.
@@ -127,13 +137,13 @@ declare_args() {
rtc_enable_bwe_test_logging = false
# Set this to false to skip building examples.
- rtc_build_examples = true
+ rtc_build_examples = false
# Set this to false to skip building tools.
- rtc_build_tools = true
+ rtc_build_tools = false
# Set this to false to skip building code that requires X11.
- rtc_use_x11 = ozone_platform_x11
+ rtc_use_x11 = use_x11
# Set this to use PipeWire on the Wayland display server.
# By default it's only enabled on desktop Linux (excludes ChromeOS) and
@@ -144,9 +154,6 @@ declare_args() {
# Set this to link PipeWire and required libraries directly instead of using the dlopen.
rtc_link_pipewire = false
- # Enable to use the Mozilla internal settings.
- build_with_mozilla = false
-
# Experimental: enable use of Android AAudio which requires Android SDK 26 or above
# and NDK r16 or above.
rtc_enable_android_aaudio = false
@@ -227,7 +234,7 @@ declare_args() {
# When set to true, a capturer implementation that uses the
# Windows.Graphics.Capture APIs will be available for use. This introduces a
# dependency on the Win 10 SDK v10.0.17763.0.
- rtc_enable_win_wgc = is_win
+ rtc_enable_win_wgc = false
# Includes the dav1d decoder in the internal decoder factory when set to true.
rtc_include_dav1d_in_internal_decoder_factory = true
@@ -256,7 +263,7 @@ declare_args() {
rtc_build_json = !build_with_mozilla
rtc_build_libsrtp = !build_with_mozilla
rtc_build_libvpx = !build_with_mozilla
- rtc_libvpx_build_vp9 = !build_with_mozilla
+ rtc_libvpx_build_vp9 = true
rtc_build_opus = !build_with_mozilla
rtc_build_ssl = !build_with_mozilla
@@ -275,7 +282,7 @@ declare_args() {
# Chromium uses its own IO handling, so the internal ADM is only built for
# standalone WebRTC.
- rtc_include_internal_audio_device = !build_with_chromium
+ rtc_include_internal_audio_device = !build_with_chromium && !build_with_mozilla
# Set this to true to enable the avx2 support in webrtc.
# TODO: Make sure that AVX2 works also for non-clang compilers.
@@ -319,6 +326,9 @@ declare_args() {
rtc_enable_grpc = rtc_enable_protobuf && (is_linux || is_mac)
}
+# Enable liboam only on non-mozilla builds.
+enable_libaom = !build_with_mozilla
+
# Make it possible to provide custom locations for some libraries (move these
# up into declare_args should we need to actually use them for the GN build).
rtc_libvpx_dir = "//third_party/libvpx"
@@ -1114,7 +1124,7 @@ if (is_mac || is_ios) {
}
}
-if (is_android) {
+if (is_android && !build_with_mozilla) {
template("rtc_android_library") {
android_library(target_name) {
forward_variables_from(invoker,
--
2.34.1

63
third_party/libwebrtc/moz-patch-stack/0038.patch поставляемый
Просмотреть файл

@ -1,29 +1,52 @@
From: Dan Minor <dminor@mozilla.com>
Date: Fri, 13 Nov 2020 14:34:00 -0500
Subject: Bug 1654112 - Fully quality AudioLevel::kUri in channel_send.cc. r=ng
Date: Wed, 18 Nov 2020 15:19:00 -0500
Subject: Bug 1654112 - Fully qualify kIvfHeaderSize. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D130082
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7163801a480d607005042292ed9e4fbb892f440d
Differential Revision: https://phabricator.services.mozilla.com/D130087
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9452e226943fea4b0f6ff67e0ee85587d9c23a44
---
audio/channel_send.cc | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
modules/video_coding/utility/ivf_file_writer.cc | 10 +++++-----
1 file changed, 5 insertions(+), 5 deletions(-)
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 361880d68f..f39e57cc39 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -690,9 +690,9 @@ void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
include_audio_level_indication_.store(enable);
if (enable) {
- rtp_rtcp_->RegisterRtpHeaderExtension(AudioLevel::Uri(), id);
+ rtp_rtcp_->RegisterRtpHeaderExtension(webrtc::AudioLevel::Uri(), id);
} else {
- rtp_rtcp_->DeregisterSendRtpHeaderExtension(AudioLevel::Uri());
+ rtp_rtcp_->DeregisterSendRtpHeaderExtension(webrtc::AudioLevel::Uri());
}
diff --git a/modules/video_coding/utility/ivf_file_writer.cc b/modules/video_coding/utility/ivf_file_writer.cc
index 668390a78c..5b27ef3ef7 100644
--- a/modules/video_coding/utility/ivf_file_writer.cc
+++ b/modules/video_coding/utility/ivf_file_writer.cc
@@ -39,7 +39,7 @@ IvfFileWriter::IvfFileWriter(FileWrapper file, size_t byte_limit)
last_timestamp_(-1),
using_capture_timestamps_(false),
file_(std::move(file)) {
- RTC_DCHECK(byte_limit == 0 || kIvfHeaderSize <= byte_limit)
+ RTC_DCHECK(byte_limit == 0 || webrtc::kIvfHeaderSize <= byte_limit)
<< "The byte_limit is too low, not even the header will fit.";
}
@@ -59,7 +59,7 @@ bool IvfFileWriter::WriteHeader() {
return false;
}
- uint8_t ivf_header[kIvfHeaderSize] = {0};
+ uint8_t ivf_header[webrtc::kIvfHeaderSize] = {0};
ivf_header[0] = 'D';
ivf_header[1] = 'K';
ivf_header[2] = 'I';
@@ -113,13 +113,13 @@ bool IvfFileWriter::WriteHeader() {
static_cast<uint32_t>(num_frames_));
ByteWriter<uint32_t>::WriteLittleEndian(&ivf_header[28], 0); // Reserved.
- if (!file_.Write(ivf_header, kIvfHeaderSize)) {
+ if (!file_.Write(ivf_header, webrtc::kIvfHeaderSize)) {
RTC_LOG(LS_ERROR) << "Unable to write IVF header for ivf output file.";
return false;
}
- if (bytes_written_ < kIvfHeaderSize) {
- bytes_written_ = kIvfHeaderSize;
+ if (bytes_written_ < webrtc::kIvfHeaderSize) {
+ bytes_written_ = webrtc::kIvfHeaderSize;
}
return true;
--
2.34.1

68
third_party/libwebrtc/moz-patch-stack/0039.patch поставляемый
Просмотреть файл

@ -1,52 +1,32 @@
From: Dan Minor <dminor@mozilla.com>
Date: Wed, 18 Nov 2020 15:19:00 -0500
Subject: Bug 1654112 - Fully qualify kIvfHeaderSize. r=ng
Date: Tue, 1 Dec 2020 09:36:00 -0500
Subject: Bug 1654112 - Disable creating av1 encoder and decoder. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D130087
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9452e226943fea4b0f6ff67e0ee85587d9c23a44
Differential Revision: https://phabricator.services.mozilla.com/D130089
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/ef548d7758c7de6e78d38af299c2296bf9d20ec9
---
modules/video_coding/utility/ivf_file_writer.cc | 10 +++++-----
1 file changed, 5 insertions(+), 5 deletions(-)
media/engine/internal_decoder_factory.cc | 2 ++
1 file changed, 2 insertions(+)
diff --git a/modules/video_coding/utility/ivf_file_writer.cc b/modules/video_coding/utility/ivf_file_writer.cc
index 668390a78c..5b27ef3ef7 100644
--- a/modules/video_coding/utility/ivf_file_writer.cc
+++ b/modules/video_coding/utility/ivf_file_writer.cc
@@ -39,7 +39,7 @@ IvfFileWriter::IvfFileWriter(FileWrapper file, size_t byte_limit)
last_timestamp_(-1),
using_capture_timestamps_(false),
file_(std::move(file)) {
- RTC_DCHECK(byte_limit == 0 || kIvfHeaderSize <= byte_limit)
+ RTC_DCHECK(byte_limit == 0 || webrtc::kIvfHeaderSize <= byte_limit)
<< "The byte_limit is too low, not even the header will fit.";
diff --git a/media/engine/internal_decoder_factory.cc b/media/engine/internal_decoder_factory.cc
index e761fd60c8..001c666313 100644
--- a/media/engine/internal_decoder_factory.cc
+++ b/media/engine/internal_decoder_factory.cc
@@ -49,12 +49,14 @@ std::vector<SdpVideoFormat> InternalDecoderFactory::GetSupportedFormats()
for (const SdpVideoFormat& h264_format : SupportedH264DecoderCodecs())
formats.push_back(h264_format);
+#if !defined(WEBRTC_MOZILLA_BUILD)
if (kDav1dIsIncluded) {
formats.push_back(SdpVideoFormat(cricket::kAv1CodecName));
formats.push_back(SdpVideoFormat(
cricket::kAv1CodecName,
{{kAV1FmtpProfile, AV1ProfileToString(AV1Profile::kProfile1).data()}}));
}
+#endif
return formats;
}
@@ -59,7 +59,7 @@ bool IvfFileWriter::WriteHeader() {
return false;
}
- uint8_t ivf_header[kIvfHeaderSize] = {0};
+ uint8_t ivf_header[webrtc::kIvfHeaderSize] = {0};
ivf_header[0] = 'D';
ivf_header[1] = 'K';
ivf_header[2] = 'I';
@@ -113,13 +113,13 @@ bool IvfFileWriter::WriteHeader() {
static_cast<uint32_t>(num_frames_));
ByteWriter<uint32_t>::WriteLittleEndian(&ivf_header[28], 0); // Reserved.
- if (!file_.Write(ivf_header, kIvfHeaderSize)) {
+ if (!file_.Write(ivf_header, webrtc::kIvfHeaderSize)) {
RTC_LOG(LS_ERROR) << "Unable to write IVF header for ivf output file.";
return false;
}
- if (bytes_written_ < kIvfHeaderSize) {
- bytes_written_ = kIvfHeaderSize;
+ if (bytes_written_ < webrtc::kIvfHeaderSize) {
+ bytes_written_ = webrtc::kIvfHeaderSize;
}
return true;
--
2.34.1

149
third_party/libwebrtc/moz-patch-stack/0040.patch поставляемый
Просмотреть файл

@ -1,32 +1,133 @@
From: Dan Minor <dminor@mozilla.com>
Date: Tue, 1 Dec 2020 09:36:00 -0500
Subject: Bug 1654112 - Disable creating av1 encoder and decoder. r=ng
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 18 Jan 2021 11:04:00 +0100
Subject: Bug 1654112 - Include RtcpPacketTypeCounter in audio send stats, to
not regress nackCount. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D130089
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/ef548d7758c7de6e78d38af299c2296bf9d20ec9
This is similar to how it's already included for video send.
Differential Revision: https://phabricator.services.mozilla.com/D102273
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d380a43d59f4f7cbc001f4eab9b63ee993b32cd8
---
media/engine/internal_decoder_factory.cc | 2 ++
1 file changed, 2 insertions(+)
audio/audio_send_stream.cc | 1 +
audio/channel_send.cc | 31 +++++++++++++++++++++++++++++++
audio/channel_send.h | 1 +
call/audio_send_stream.h | 2 ++
4 files changed, 35 insertions(+)
diff --git a/media/engine/internal_decoder_factory.cc b/media/engine/internal_decoder_factory.cc
index e761fd60c8..001c666313 100644
--- a/media/engine/internal_decoder_factory.cc
+++ b/media/engine/internal_decoder_factory.cc
@@ -49,12 +49,14 @@ std::vector<SdpVideoFormat> InternalDecoderFactory::GetSupportedFormats()
for (const SdpVideoFormat& h264_format : SupportedH264DecoderCodecs())
formats.push_back(h264_format);
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 7d6ec794d4..20af3f7722 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -442,6 +442,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
stats.target_bitrate_bps = channel_send_->GetTargetBitrate();
+#if !defined(WEBRTC_MOZILLA_BUILD)
if (kDav1dIsIncluded) {
formats.push_back(SdpVideoFormat(cricket::kAv1CodecName));
formats.push_back(SdpVideoFormat(
cricket::kAv1CodecName,
{{kAV1FmtpProfile, AV1ProfileToString(AV1Profile::kProfile1).data()}}));
}
+#endif
webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
+ stats.rtcp_packet_type_counts = call_stats.rtcp_packet_type_counts;
stats.payload_bytes_sent = call_stats.payload_bytes_sent;
stats.header_and_padding_bytes_sent =
call_stats.header_and_padding_bytes_sent;
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index f39e57cc39..bf8e81fda4 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -56,6 +56,31 @@ class RtpPacketSenderProxy;
class TransportSequenceNumberProxy;
class VoERtcpObserver;
return formats;
}
+class RtcpCounterObserver : public RtcpPacketTypeCounterObserver {
+ public:
+ explicit RtcpCounterObserver(uint32_t ssrc) : ssrc_(ssrc) {}
+
+ void RtcpPacketTypesCounterUpdated(
+ uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) override {
+ if (ssrc_ != ssrc) {
+ return;
+ }
+
+ MutexLock lock(&mutex_);
+ packet_counter_ = packet_counter;
+ }
+
+ RtcpPacketTypeCounter GetCounts() {
+ MutexLock lock(&mutex_);
+ return packet_counter_;
+ }
+
+ private:
+ Mutex mutex_;
+ const uint32_t ssrc_;
+ RtcpPacketTypeCounter packet_counter_;
+};
+
class ChannelSend : public ChannelSendInterface,
public AudioPacketizationCallback, // receive encoded
// packets from the ACM
@@ -208,6 +233,8 @@ class ChannelSend : public ChannelSendInterface,
// RtcpBandwidthObserver
const std::unique_ptr<VoERtcpObserver> rtcp_observer_;
+ const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_;
+
PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
nullptr;
TransportFeedbackObserver* const feedback_observer_;
@@ -460,6 +487,7 @@ ChannelSend::ChannelSend(
: ssrc_(ssrc),
event_log_(rtc_event_log),
rtcp_observer_(new VoERtcpObserver(this)),
+ rtcp_counter_observer_(new RtcpCounterObserver(ssrc)),
feedback_observer_(feedback_observer),
rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(
@@ -482,6 +510,8 @@ ChannelSend::ChannelSend(
configuration.event_log = event_log_;
configuration.rtt_stats = rtcp_rtt_stats;
+ configuration.rtcp_packet_type_counter_observer =
+ rtcp_counter_observer_.get();
configuration.retransmission_rate_limiter =
retransmission_rate_limiter_.get();
configuration.extmap_allow_mixed = extmap_allow_mixed;
@@ -757,6 +787,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
CallSendStatistics stats = {0};
stats.rttMs = GetRTT();
+ stats.rtcp_packet_type_counts = rtcp_counter_observer_->GetCounts();
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
diff --git a/audio/channel_send.h b/audio/channel_send.h
index cf9a273f70..9b3969161c 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -43,6 +43,7 @@ struct CallSendStatistics {
TimeDelta total_packet_send_delay = TimeDelta::Zero();
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
uint64_t retransmitted_packets_sent;
+ RtcpPacketTypeCounter rtcp_packet_type_counts;
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the sender-source SSRC pair is unique and per-pair the
// ReportBlockData represents the latest Report Block that was received for
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 0f42d0fb82..bafa22d312 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -31,6 +31,7 @@
#include "call/rtp_config.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
namespace webrtc {
@@ -65,6 +66,7 @@ class AudioSendStream : public AudioSender {
ANAStats ana_statistics;
AudioProcessingStats apm_statistics;
+ RtcpPacketTypeCounter rtcp_packet_type_counts;
int64_t target_bitrate_bps = 0;
// A snapshot of Report Blocks with additional data of interest to
--
2.34.1

155
third_party/libwebrtc/moz-patch-stack/0041.patch поставляемый
Просмотреть файл

@ -1,133 +1,44 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 18 Jan 2021 11:04:00 +0100
Subject: Bug 1654112 - Include RtcpPacketTypeCounter in audio send stats, to
not regress nackCount. r=ng
Date: Fri, 19 Feb 2021 13:45:00 +0100
Subject: Bug 1654112 - libwebrtc: Add a REMB on/off switch to
VideoReceiveStream. r=ng
This is similar to how it's already included for video send.
Differential Revision: https://phabricator.services.mozilla.com/D102273
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d380a43d59f4f7cbc001f4eab9b63ee993b32cd8
Differential Revision: https://phabricator.services.mozilla.com/D105774
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7330681cf4de6d8dd868cc661cbdd6679bbc07b3
---
audio/audio_send_stream.cc | 1 +
audio/channel_send.cc | 31 +++++++++++++++++++++++++++++++
audio/channel_send.h | 1 +
call/audio_send_stream.h | 2 ++
4 files changed, 35 insertions(+)
call/video_receive_stream.h | 3 +++
video/rtp_video_stream_receiver2.cc | 3 +--
2 files changed, 4 insertions(+), 2 deletions(-)
diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
index 7d6ec794d4..20af3f7722 100644
--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
@@ -442,6 +442,7 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
stats.target_bitrate_bps = channel_send_->GetTargetBitrate();
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 3125993d4b..cda8b1f6af 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -203,6 +203,9 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
// disabled.
KeyFrameReqMethod keyframe_method = KeyFrameReqMethod::kPliRtcp;
webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
+ stats.rtcp_packet_type_counts = call_stats.rtcp_packet_type_counts;
stats.payload_bytes_sent = call_stats.payload_bytes_sent;
stats.header_and_padding_bytes_sent =
call_stats.header_and_padding_bytes_sent;
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index f39e57cc39..bf8e81fda4 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -56,6 +56,31 @@ class RtpPacketSenderProxy;
class TransportSequenceNumberProxy;
class VoERtcpObserver;
+class RtcpCounterObserver : public RtcpPacketTypeCounterObserver {
+ public:
+ explicit RtcpCounterObserver(uint32_t ssrc) : ssrc_(ssrc) {}
+ // See draft-alvestrand-rmcat-remb for information.
+ bool remb = false;
+
+ void RtcpPacketTypesCounterUpdated(
+ uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) override {
+ if (ssrc_ != ssrc) {
+ return;
+ }
+
+ MutexLock lock(&mutex_);
+ packet_counter_ = packet_counter;
+ }
+
+ RtcpPacketTypeCounter GetCounts() {
+ MutexLock lock(&mutex_);
+ return packet_counter_;
+ }
+
+ private:
+ Mutex mutex_;
+ const uint32_t ssrc_;
+ RtcpPacketTypeCounter packet_counter_;
+};
+
class ChannelSend : public ChannelSendInterface,
public AudioPacketizationCallback, // receive encoded
// packets from the ACM
@@ -208,6 +233,8 @@ class ChannelSend : public ChannelSendInterface,
// RtcpBandwidthObserver
const std::unique_ptr<VoERtcpObserver> rtcp_observer_;
bool tmmbr = false;
+ const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_;
+
PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
nullptr;
TransportFeedbackObserver* const feedback_observer_;
@@ -460,6 +487,7 @@ ChannelSend::ChannelSend(
: ssrc_(ssrc),
event_log_(rtc_event_log),
rtcp_observer_(new VoERtcpObserver(this)),
+ rtcp_counter_observer_(new RtcpCounterObserver(ssrc)),
feedback_observer_(feedback_observer),
rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
retransmission_rate_limiter_(
@@ -482,6 +510,8 @@ ChannelSend::ChannelSend(
// See LntfConfig for description.
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index e7f88665b6..eed9770d93 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -297,9 +297,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
frames_decryptable_(false),
absolute_capture_time_interpolator_(clock) {
packet_sequence_checker_.Detach();
- constexpr bool remb_candidate = true;
if (packet_router_)
- packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
+ packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), config_.rtp.remb);
configuration.event_log = event_log_;
configuration.rtt_stats = rtcp_rtt_stats;
+ configuration.rtcp_packet_type_counter_observer =
+ rtcp_counter_observer_.get();
configuration.retransmission_rate_limiter =
retransmission_rate_limiter_.get();
configuration.extmap_allow_mixed = extmap_allow_mixed;
@@ -757,6 +787,7 @@ CallSendStatistics ChannelSend::GetRTCPStatistics() const {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
CallSendStatistics stats = {0};
stats.rttMs = GetRTT();
+ stats.rtcp_packet_type_counts = rtcp_counter_observer_->GetCounts();
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
diff --git a/audio/channel_send.h b/audio/channel_send.h
index cf9a273f70..9b3969161c 100644
--- a/audio/channel_send.h
+++ b/audio/channel_send.h
@@ -43,6 +43,7 @@ struct CallSendStatistics {
TimeDelta total_packet_send_delay = TimeDelta::Zero();
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
uint64_t retransmitted_packets_sent;
+ RtcpPacketTypeCounter rtcp_packet_type_counts;
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the sender-source SSRC pair is unique and per-pair the
// ReportBlockData represents the latest Report Block that was received for
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index 0f42d0fb82..bafa22d312 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -31,6 +31,7 @@
#include "call/rtp_config.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
namespace webrtc {
@@ -65,6 +66,7 @@ class AudioSendStream : public AudioSender {
ANAStats ana_statistics;
AudioProcessingStats apm_statistics;
+ RtcpPacketTypeCounter rtcp_packet_type_counts;
int64_t target_bitrate_bps = 0;
// A snapshot of Report Blocks with additional data of interest to
RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
<< "A stream should not be configured with RTCP disabled. This value is "
--
2.34.1

65
third_party/libwebrtc/moz-patch-stack/0042.patch поставляемый
Просмотреть файл

@ -1,44 +1,41 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Fri, 19 Feb 2021 13:45:00 +0100
Subject: Bug 1654112 - libwebrtc: Add a REMB on/off switch to
VideoReceiveStream. r=ng
From: Nico Grunbaum <na-g@nostrum.com>
Date: Wed, 10 Feb 2021 12:24:00 -0800
Subject: Bug 1654112 - Use newer thread run callback, and adapt
PlatformUIThread; r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D105774
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7330681cf4de6d8dd868cc661cbdd6679bbc07b3
Differential Revision: https://phabricator.services.mozilla.com/D107879
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/bb6417a4cfac1416a8e2565bd68b66c40be4827b
---
call/video_receive_stream.h | 3 +++
video/rtp_video_stream_receiver2.cc | 3 +--
2 files changed, 4 insertions(+), 2 deletions(-)
rtc_base/platform_thread.h | 7 +++++++
1 file changed, 7 insertions(+)
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 3125993d4b..cda8b1f6af 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -203,6 +203,9 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
// disabled.
KeyFrameReqMethod keyframe_method = KeyFrameReqMethod::kPliRtcp;
diff --git a/rtc_base/platform_thread.h b/rtc_base/platform_thread.h
index 3ab2761f43..52497ffd0c 100644
--- a/rtc_base/platform_thread.h
+++ b/rtc_base/platform_thread.h
@@ -18,8 +18,13 @@
#include "absl/types/optional.h"
#include "rtc_base/platform_thread_types.h"
+ // See draft-alvestrand-rmcat-remb for information.
+ bool remb = false;
+#include "rtc_base/deprecated/recursive_critical_section.h"
+
bool tmmbr = false;
namespace rtc {
// See LntfConfig for description.
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index e7f88665b6..eed9770d93 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -297,9 +297,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
frames_decryptable_(false),
absolute_capture_time_interpolator_(clock) {
packet_sequence_checker_.Detach();
- constexpr bool remb_candidate = true;
if (packet_router_)
- packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
+ packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), config_.rtp.remb);
+// Bug 1691641
+class PlatformUIThread;
+
enum class ThreadPriority {
kLow = 1,
kNormal,
@@ -110,6 +115,8 @@ class PlatformThread final {
RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
<< "A stream should not be configured with RTCP disabled. This value is "
absl::optional<Handle> handle_;
bool joinable_ = false;
+ // Bug 1691641
+ friend PlatformUIThread;
};
} // namespace rtc
--
2.34.1

76
third_party/libwebrtc/moz-patch-stack/0043.patch поставляемый
Просмотреть файл

@ -1,41 +1,57 @@
From: Nico Grunbaum <na-g@nostrum.com>
Date: Wed, 10 Feb 2021 12:24:00 -0800
Subject: Bug 1654112 - Use newer thread run callback, and adapt
PlatformUIThread; r=pehrsons
Date: Thu, 18 Feb 2021 17:23:00 -0800
Subject: Bug 1654112 - fix device_info_ds pid and Windows constants includes;
r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D107879
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/bb6417a4cfac1416a8e2565bd68b66c40be4827b
Upstreaming bug 1697385
Differential Revision: https://phabricator.services.mozilla.com/D107899
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/24809d566449907edea49ea47528065ad0f76910
---
rtc_base/platform_thread.h | 7 +++++++
1 file changed, 7 insertions(+)
modules/video_capture/windows/device_info_ds.cc | 3 ++-
modules/video_capture/windows/device_info_ds.h | 6 +++++-
2 files changed, 7 insertions(+), 2 deletions(-)
diff --git a/rtc_base/platform_thread.h b/rtc_base/platform_thread.h
index 3ab2761f43..52497ffd0c 100644
--- a/rtc_base/platform_thread.h
+++ b/rtc_base/platform_thread.h
@@ -18,8 +18,13 @@
#include "absl/types/optional.h"
#include "rtc_base/platform_thread_types.h"
diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc
index 3ab95837c0..6b4c57d01e 100644
--- a/modules/video_capture/windows/device_info_ds.cc
+++ b/modules/video_capture/windows/device_info_ds.cc
@@ -172,7 +172,8 @@ int32_t DeviceInfoDS::GetDeviceName(uint32_t deviceNumber,
char* deviceUniqueIdUTF8,
uint32_t deviceUniqueIdUTF8Length,
char* productUniqueIdUTF8,
- uint32_t productUniqueIdUTF8Length) {
+ uint32_t productUniqueIdUTF8Length,
+ pid_t* pid) {
MutexLock lock(&_apiLock);
const int32_t result = GetDeviceInfo(
deviceNumber, deviceNameUTF8, deviceNameLength, deviceUniqueIdUTF8,
diff --git a/modules/video_capture/windows/device_info_ds.h b/modules/video_capture/windows/device_info_ds.h
index ed2a726d6f..e6dfaed366 100644
--- a/modules/video_capture/windows/device_info_ds.h
+++ b/modules/video_capture/windows/device_info_ds.h
@@ -12,8 +12,11 @@
#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_DS_H_
+#include "rtc_base/deprecated/recursive_critical_section.h"
+
namespace rtc {
#include <dshow.h>
+#include <Ks.h>
+#include <dbt.h>
+// Bug 1691641
+class PlatformUIThread;
+
enum class ThreadPriority {
kLow = 1,
kNormal,
@@ -110,6 +115,8 @@ class PlatformThread final {
#include "modules/video_capture/device_info_impl.h"
+#include "modules/video_capture/video_capture.h"
#include "modules/video_capture/video_capture_impl.h"
absl::optional<Handle> handle_;
bool joinable_ = false;
+ // Bug 1691641
+ friend PlatformUIThread;
};
namespace webrtc {
@@ -47,7 +50,8 @@ class DeviceInfoDS : public DeviceInfoImpl {
char* deviceUniqueIdUTF8,
uint32_t deviceUniqueIdUTF8Length,
char* productUniqueIdUTF8,
- uint32_t productUniqueIdUTF8Length) override;
+ uint32_t productUniqueIdUTF8Length,
+ pid_t* pid) override;
} // namespace rtc
/*
* Display OS /capture device specific settings dialog
--
2.34.1

95
third_party/libwebrtc/moz-patch-stack/0044.patch поставляемый
Просмотреть файл

@ -1,57 +1,50 @@
From: Nico Grunbaum <na-g@nostrum.com>
Date: Thu, 18 Feb 2021 17:23:00 -0800
Subject: Bug 1654112 - fix device_info_ds pid and Windows constants includes;
r=pehrsons
From: "Byron Campen [:bwc]" <docfaraday@gmail.com>
Date: Thu, 29 Apr 2021 18:25:00 +0000
Subject: Bug 1654112 - Work around the old (<1.5) libxrandr headers on our
build machines. r=mjf,ng
Upstreaming bug 1697385
Differential Revision: https://phabricator.services.mozilla.com/D107899
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/24809d566449907edea49ea47528065ad0f76910
Differential Revision: https://phabricator.services.mozilla.com/D113830
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c91f12b557a1d23b468c75c4f2fc00eb0f8d541a
---
modules/video_capture/windows/device_info_ds.cc | 3 ++-
modules/video_capture/windows/device_info_ds.h | 6 +++++-
2 files changed, 7 insertions(+), 2 deletions(-)
X11/extensions/Xrandr.h | 29 +++++++++++++++++++++++++++++
1 file changed, 29 insertions(+)
create mode 100644 X11/extensions/Xrandr.h
diff --git a/modules/video_capture/windows/device_info_ds.cc b/modules/video_capture/windows/device_info_ds.cc
index 3ab95837c0..6b4c57d01e 100644
--- a/modules/video_capture/windows/device_info_ds.cc
+++ b/modules/video_capture/windows/device_info_ds.cc
@@ -172,7 +172,8 @@ int32_t DeviceInfoDS::GetDeviceName(uint32_t deviceNumber,
char* deviceUniqueIdUTF8,
uint32_t deviceUniqueIdUTF8Length,
char* productUniqueIdUTF8,
- uint32_t productUniqueIdUTF8Length) {
+ uint32_t productUniqueIdUTF8Length,
+ pid_t* pid) {
MutexLock lock(&_apiLock);
const int32_t result = GetDeviceInfo(
deviceNumber, deviceNameUTF8, deviceNameLength, deviceUniqueIdUTF8,
diff --git a/modules/video_capture/windows/device_info_ds.h b/modules/video_capture/windows/device_info_ds.h
index ed2a726d6f..e6dfaed366 100644
--- a/modules/video_capture/windows/device_info_ds.h
+++ b/modules/video_capture/windows/device_info_ds.h
@@ -12,8 +12,11 @@
#define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_DS_H_
#include <dshow.h>
+#include <Ks.h>
+#include <dbt.h>
#include "modules/video_capture/device_info_impl.h"
+#include "modules/video_capture/video_capture.h"
#include "modules/video_capture/video_capture_impl.h"
namespace webrtc {
@@ -47,7 +50,8 @@ class DeviceInfoDS : public DeviceInfoImpl {
char* deviceUniqueIdUTF8,
uint32_t deviceUniqueIdUTF8Length,
char* productUniqueIdUTF8,
- uint32_t productUniqueIdUTF8Length) override;
+ uint32_t productUniqueIdUTF8Length,
+ pid_t* pid) override;
/*
* Display OS /capture device specific settings dialog
diff --git a/X11/extensions/Xrandr.h b/X11/extensions/Xrandr.h
new file mode 100644
index 0000000000..876e8b4c7f
--- /dev/null
+++ b/X11/extensions/Xrandr.h
@@ -0,0 +1,29 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+// Hack to compensate for the old (<1.5) Xrandr development headers on
+// Mozilla's build boxes.
+
+#ifndef _XRANDR_H_WRAPPER_HACK_
+#define _XRANDR_H_WRAPPER_HACK_
+
+#include_next <X11/extensions/Xrandr.h>
+
+#if RANDR_MAJOR == 1 && RANDR_MINOR < 5 // defined in randr.h
+typedef struct _XRRMonitorInfo {
+ Atom name;
+ Bool primary;
+ Bool automatic;
+ int noutput;
+ int x;
+ int y;
+ int width;
+ int height;
+ int mwidth;
+ int mheight;
+ RROutput *outputs;
+} XRRMonitorInfo;
+#endif
+
+#endif // _XRANDR_H_WRAPPER_HACK_
--
2.34.1

99
third_party/libwebrtc/moz-patch-stack/0045.patch поставляемый
Просмотреть файл

@ -1,50 +1,63 @@
From: "Byron Campen [:bwc]" <docfaraday@gmail.com>
Date: Thu, 29 Apr 2021 18:25:00 +0000
Subject: Bug 1654112 - Work around the old (<1.5) libxrandr headers on our
build machines. r=mjf,ng
From: Michael Froman <mfroman@mozilla.com>
Date: Fri, 16 Apr 2021 17:35:00 -0500
Subject: Bug 1654112 - General build fixes for paths and naming changes. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D113830
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c91f12b557a1d23b468c75c4f2fc00eb0f8d541a
Differential Revision: https://phabricator.services.mozilla.com/D113438
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/99b99cca6b7b2c2ebffb5472457a4f927bda11c4
---
X11/extensions/Xrandr.h | 29 +++++++++++++++++++++++++++++
1 file changed, 29 insertions(+)
create mode 100644 X11/extensions/Xrandr.h
modules/video_capture/video_capture_impl.h | 4 +++-
rtc_base/system/warn_current_thread_is_deadlocked.h | 2 +-
sdk/android/api/org/webrtc/VideoCodecInfo.java | 2 +-
3 files changed, 5 insertions(+), 3 deletions(-)
diff --git a/X11/extensions/Xrandr.h b/X11/extensions/Xrandr.h
new file mode 100644
index 0000000000..876e8b4c7f
--- /dev/null
+++ b/X11/extensions/Xrandr.h
@@ -0,0 +1,29 @@
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
diff --git a/modules/video_capture/video_capture_impl.h b/modules/video_capture/video_capture_impl.h
index f874580471..b9af5f2441 100644
--- a/modules/video_capture/video_capture_impl.h
+++ b/modules/video_capture/video_capture_impl.h
@@ -81,6 +81,9 @@ class VideoCaptureImpl : public VideoCaptureModule {
VideoCaptureImpl();
~VideoCaptureImpl() override;
+ // moved DeliverCapturedFrame to protected for VideoCaptureAndroid (mjf)
+ int32_t DeliverCapturedFrame(VideoFrame& captureFrame);
+
+// Hack to compensate for the old (<1.5) Xrandr development headers on
+// Mozilla's build boxes.
+
+#ifndef _XRANDR_H_WRAPPER_HACK_
+#define _XRANDR_H_WRAPPER_HACK_
+
+#include_next <X11/extensions/Xrandr.h>
+
+#if RANDR_MAJOR == 1 && RANDR_MINOR < 5 // defined in randr.h
+typedef struct _XRRMonitorInfo {
+ Atom name;
+ Bool primary;
+ Bool automatic;
+ int noutput;
+ int x;
+ int y;
+ int width;
+ int height;
+ int mwidth;
+ int mheight;
+ RROutput *outputs;
+} XRRMonitorInfo;
+#endif
+
+#endif // _XRANDR_H_WRAPPER_HACK_
char* _deviceUniqueId; // current Device unique name;
Mutex api_lock_;
VideoCaptureCapability _requestedCapability; // Should be set by platform
@@ -89,7 +92,6 @@ class VideoCaptureImpl : public VideoCaptureModule {
private:
void UpdateFrameCount();
uint32_t CalculateFrameRate(int64_t now_ns);
- int32_t DeliverCapturedFrame(VideoFrame& captureFrame);
void DeliverRawFrame(uint8_t* videoFrame,
size_t videoFrameLength,
const VideoCaptureCapability& frameInfo,
diff --git a/rtc_base/system/warn_current_thread_is_deadlocked.h b/rtc_base/system/warn_current_thread_is_deadlocked.h
index 4a0ba9dc09..eac12022ed 100644
--- a/rtc_base/system/warn_current_thread_is_deadlocked.h
+++ b/rtc_base/system/warn_current_thread_is_deadlocked.h
@@ -13,7 +13,7 @@
namespace webrtc {
-#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(MOZ_WIDGET_ANDROID)
void WarnThatTheCurrentThreadIsProbablyDeadlocked();
#else
inline void WarnThatTheCurrentThreadIsProbablyDeadlocked() {}
diff --git a/sdk/android/api/org/webrtc/VideoCodecInfo.java b/sdk/android/api/org/webrtc/VideoCodecInfo.java
index 4f97cf74cf..363be347b5 100644
--- a/sdk/android/api/org/webrtc/VideoCodecInfo.java
+++ b/sdk/android/api/org/webrtc/VideoCodecInfo.java
@@ -80,7 +80,7 @@ public class VideoCodecInfo {
}
@CalledByNative
- Map getParams() {
+ Map<String, String> getParams() {
return params;
}
}
--
2.34.1

102
third_party/libwebrtc/moz-patch-stack/0046.patch поставляемый
Просмотреть файл

@ -1,63 +1,61 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Fri, 16 Apr 2021 17:35:00 -0500
Subject: Bug 1654112 - General build fixes for paths and naming changes. r=ng
Date: Tue, 15 Jun 2021 12:18:00 -0500
Subject: Bug 1654112 - suppress android lint warnings for WrongConstant in 2
libwebrtc java files. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D113438
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/99b99cca6b7b2c2ebffb5472457a4f927bda11c4
Differential Revision: https://phabricator.services.mozilla.com/D118050
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e60e2f295fb722f69a3a9fe9af34219880afe772
---
modules/video_capture/video_capture_impl.h | 4 +++-
rtc_base/system/warn_current_thread_is_deadlocked.h | 2 +-
sdk/android/api/org/webrtc/VideoCodecInfo.java | 2 +-
3 files changed, 5 insertions(+), 3 deletions(-)
.../java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java | 5 +++++
sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java | 5 +++++
2 files changed, 10 insertions(+)
diff --git a/modules/video_capture/video_capture_impl.h b/modules/video_capture/video_capture_impl.h
index f874580471..b9af5f2441 100644
--- a/modules/video_capture/video_capture_impl.h
+++ b/modules/video_capture/video_capture_impl.h
@@ -81,6 +81,9 @@ class VideoCaptureImpl : public VideoCaptureModule {
VideoCaptureImpl();
~VideoCaptureImpl() override;
diff --git a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
index 0472114297..afd3d429af 100644
--- a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
+++ b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
@@ -15,6 +15,7 @@ import static android.media.AudioManager.MODE_IN_COMMUNICATION;
import static android.media.AudioManager.MODE_NORMAL;
import static android.media.AudioManager.MODE_RINGTONE;
+ // moved DeliverCapturedFrame to protected for VideoCaptureAndroid (mjf)
+ int32_t DeliverCapturedFrame(VideoFrame& captureFrame);
+
char* _deviceUniqueId; // current Device unique name;
Mutex api_lock_;
VideoCaptureCapability _requestedCapability; // Should be set by platform
@@ -89,7 +92,6 @@ class VideoCaptureImpl : public VideoCaptureModule {
private:
void UpdateFrameCount();
uint32_t CalculateFrameRate(int64_t now_ns);
- int32_t DeliverCapturedFrame(VideoFrame& captureFrame);
void DeliverRawFrame(uint8_t* videoFrame,
size_t videoFrameLength,
const VideoCaptureCapability& frameInfo,
diff --git a/rtc_base/system/warn_current_thread_is_deadlocked.h b/rtc_base/system/warn_current_thread_is_deadlocked.h
index 4a0ba9dc09..eac12022ed 100644
--- a/rtc_base/system/warn_current_thread_is_deadlocked.h
+++ b/rtc_base/system/warn_current_thread_is_deadlocked.h
@@ -13,7 +13,7 @@
namespace webrtc {
-#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(MOZ_WIDGET_ANDROID)
void WarnThatTheCurrentThreadIsProbablyDeadlocked();
#else
inline void WarnThatTheCurrentThreadIsProbablyDeadlocked() {}
diff --git a/sdk/android/api/org/webrtc/VideoCodecInfo.java b/sdk/android/api/org/webrtc/VideoCodecInfo.java
index 4f97cf74cf..363be347b5 100644
--- a/sdk/android/api/org/webrtc/VideoCodecInfo.java
+++ b/sdk/android/api/org/webrtc/VideoCodecInfo.java
@@ -80,7 +80,7 @@ public class VideoCodecInfo {
+import android.annotation.SuppressLint;
import android.content.Context;
import android.content.pm.PackageManager;
import android.media.AudioDeviceInfo;
@@ -247,6 +248,10 @@ public final class WebRtcAudioUtils {
}
}
@CalledByNative
- Map getParams() {
+ Map<String, String> getParams() {
return params;
+ // Moz linting complains even though AudioManager.GET_DEVICES_ALL is
+ // listed in the docs here:
+ // https://developer.android.com/reference/android/media/AudioManager#GET_DEVICES_ALL
+ @SuppressLint("WrongConstant")
private static void logAudioDeviceInfo(String tag, AudioManager audioManager) {
if (Build.VERSION.SDK_INT < 23) {
return;
diff --git a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
index 7894659926..7b4b809ab1 100644
--- a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
+++ b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
@@ -15,6 +15,7 @@ import static android.media.AudioManager.MODE_IN_COMMUNICATION;
import static android.media.AudioManager.MODE_NORMAL;
import static android.media.AudioManager.MODE_RINGTONE;
+import android.annotation.SuppressLint;
import android.annotation.TargetApi;
import android.content.Context;
import android.content.pm.PackageManager;
@@ -229,6 +230,10 @@ final class WebRtcAudioUtils {
}
}
}
+ // Moz linting complains even though AudioManager.GET_DEVICES_ALL is
+ // listed in the docs here:
+ // https://developer.android.com/reference/android/media/AudioManager#GET_DEVICES_ALL
+ @SuppressLint("WrongConstant")
private static void logAudioDeviceInfo(String tag, AudioManager audioManager) {
if (Build.VERSION.SDK_INT < 23) {
return;
--
2.34.1

79
third_party/libwebrtc/moz-patch-stack/0047.patch поставляемый
Просмотреть файл

@ -1,61 +1,38 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Tue, 15 Jun 2021 12:18:00 -0500
Subject: Bug 1654112 - suppress android lint warnings for WrongConstant in 2
libwebrtc java files. r=ng
Date: Fri, 25 Jun 2021 15:12:00 -0500
Subject: Bug 1654112 - Mirror Bug 1714577 - Part 3 - Register WebRTC threads
with the Gecko Profiler. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D118050
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e60e2f295fb722f69a3a9fe9af34219880afe772
Differential Revision: https://phabricator.services.mozilla.com/D119412
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d881b16dd8a6813feb5ce1516c2a7ebe0270e72d
---
.../java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java | 5 +++++
sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java | 5 +++++
2 files changed, 10 insertions(+)
rtc_base/platform_thread.cc | 6 ++++++
1 file changed, 6 insertions(+)
diff --git a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
index 0472114297..afd3d429af 100644
--- a/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
+++ b/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
@@ -15,6 +15,7 @@ import static android.media.AudioManager.MODE_IN_COMMUNICATION;
import static android.media.AudioManager.MODE_NORMAL;
import static android.media.AudioManager.MODE_RINGTONE;
diff --git a/rtc_base/platform_thread.cc b/rtc_base/platform_thread.cc
index 556204ac89..71a9f1b224 100644
--- a/rtc_base/platform_thread.cc
+++ b/rtc_base/platform_thread.cc
@@ -19,6 +19,8 @@
+import android.annotation.SuppressLint;
import android.content.Context;
import android.content.pm.PackageManager;
import android.media.AudioDeviceInfo;
@@ -247,6 +248,10 @@ public final class WebRtcAudioUtils {
}
}
#include "rtc_base/checks.h"
+ // Moz linting complains even though AudioManager.GET_DEVICES_ALL is
+ // listed in the docs here:
+ // https://developer.android.com/reference/android/media/AudioManager#GET_DEVICES_ALL
+ @SuppressLint("WrongConstant")
private static void logAudioDeviceInfo(String tag, AudioManager audioManager) {
if (Build.VERSION.SDK_INT < 23) {
return;
diff --git a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
index 7894659926..7b4b809ab1 100644
--- a/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
+++ b/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
@@ -15,6 +15,7 @@ import static android.media.AudioManager.MODE_IN_COMMUNICATION;
import static android.media.AudioManager.MODE_NORMAL;
import static android.media.AudioManager.MODE_RINGTONE;
+#include "MicroGeckoProfiler.h"
+
namespace rtc {
namespace {
+import android.annotation.SuppressLint;
import android.annotation.TargetApi;
import android.content.Context;
import android.content.pm.PackageManager;
@@ -229,6 +230,10 @@ final class WebRtcAudioUtils {
}
}
+ // Moz linting complains even though AudioManager.GET_DEVICES_ALL is
+ // listed in the docs here:
+ // https://developer.android.com/reference/android/media/AudioManager#GET_DEVICES_ALL
+ @SuppressLint("WrongConstant")
private static void logAudioDeviceInfo(String tag, AudioManager audioManager) {
if (Build.VERSION.SDK_INT < 23) {
return;
@@ -181,6 +183,10 @@ PlatformThread PlatformThread::SpawnThread(
new std::function<void()>([thread_function = std::move(thread_function),
name = std::string(name), attributes] {
rtc::SetCurrentThreadName(name.c_str());
+
+ char stacktop;
+ AutoRegisterProfiler profiler(name.c_str(), &stacktop);
+
SetPriority(attributes.priority);
thread_function();
});
--
2.34.1

320
third_party/libwebrtc/moz-patch-stack/0048.patch поставляемый
Просмотреть файл

@ -1,38 +1,300 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Fri, 25 Jun 2021 15:12:00 -0500
Subject: Bug 1654112 - Mirror Bug 1714577 - Part 3 - Register WebRTC threads
with the Gecko Profiler. r=ng
From: Nico Grunbaum <na-g@nostrum.com>
Date: Wed, 14 Jul 2021 22:26:00 +0000
Subject: Bug 1654112 - deconflate the target and host architectures in
libwebrtc build files; r=mjf
Differential Revision: https://phabricator.services.mozilla.com/D119412
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d881b16dd8a6813feb5ce1516c2a7ebe0270e72d
Differential Revision: https://phabricator.services.mozilla.com/D119707
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/58f47eacaf10d12e21dff7362743b6f4cdd1696b
---
rtc_base/platform_thread.cc | 6 ++++++
1 file changed, 6 insertions(+)
BUILD.gn | 6 +++---
common_audio/BUILD.gn | 16 ++++++++--------
common_audio/third_party/ooura/BUILD.gn | 6 +++---
common_audio/third_party/spl_sqrt_floor/BUILD.gn | 4 ++--
modules/audio_processing/aec3/BUILD.gn | 6 +++---
modules/audio_processing/aecm/BUILD.gn | 4 ++--
modules/audio_processing/agc/BUILD.gn | 2 +-
modules/audio_processing/agc2/rnn_vad/BUILD.gn | 2 +-
modules/audio_processing/ns/BUILD.gn | 2 +-
modules/desktop_capture/BUILD.gn | 2 +-
webrtc.gni | 4 ++--
11 files changed, 27 insertions(+), 27 deletions(-)
diff --git a/rtc_base/platform_thread.cc b/rtc_base/platform_thread.cc
index 556204ac89..71a9f1b224 100644
--- a/rtc_base/platform_thread.cc
+++ b/rtc_base/platform_thread.cc
@@ -19,6 +19,8 @@
diff --git a/BUILD.gn b/BUILD.gn
index 02e9d1d4a0..4e2a1f1e70 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -407,12 +407,12 @@ config("common_config") {
}
}
#include "rtc_base/checks.h"
- if (current_cpu == "arm64") {
+ if (target_cpu == "arm64") {
defines += [ "WEBRTC_ARCH_ARM64" ]
defines += [ "WEBRTC_HAS_NEON" ]
}
+#include "MicroGeckoProfiler.h"
+
namespace rtc {
namespace {
- if (current_cpu == "arm") {
+ if (target_cpu == "arm") {
defines += [ "WEBRTC_ARCH_ARM" ]
if (arm_version >= 7) {
defines += [ "WEBRTC_ARCH_ARM_V7" ]
@@ -422,7 +422,7 @@ config("common_config") {
}
}
@@ -181,6 +183,10 @@ PlatformThread PlatformThread::SpawnThread(
new std::function<void()>([thread_function = std::move(thread_function),
name = std::string(name), attributes] {
rtc::SetCurrentThreadName(name.c_str());
+
+ char stacktop;
+ AutoRegisterProfiler profiler(name.c_str(), &stacktop);
+
SetPriority(attributes.priority);
thread_function();
});
- if (current_cpu == "mipsel") {
+ if (target_cpu == "mipsel") {
defines += [ "MIPS32_LE" ]
if (mips_float_abi == "hard") {
defines += [ "MIPS_FPU_LE" ]
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index a45214f754..79d9321bbd 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -66,7 +66,7 @@ rtc_library("common_audio") {
deps += [ ":common_audio_neon" ]
}
- if (current_cpu == "x86" || current_cpu == "x64") {
+ if (target_cpu == "x86" || target_cpu == "x64") {
deps += [ ":common_audio_sse2" ]
deps += [ ":common_audio_avx2" ]
}
@@ -88,7 +88,7 @@ rtc_source_set("mock_common_audio") {
rtc_source_set("common_audio_c_arm_asm") {
sources = []
deps = []
- if (current_cpu == "arm") {
+ if (target_cpu == "arm") {
sources += [ "signal_processing/complex_bit_reverse_arm.S" ]
if (arm_version >= 7) {
@@ -152,7 +152,7 @@ rtc_library("common_audio_c") {
"vad/webrtc_vad.c",
]
- if (current_cpu == "mipsel") {
+ if (target_cpu == "mipsel") {
sources += [
"signal_processing/complex_bit_reverse_mips.c",
"signal_processing/complex_fft_mips.c",
@@ -170,7 +170,7 @@ rtc_library("common_audio_c") {
sources += [ "signal_processing/complex_fft.c" ]
}
- if (current_cpu != "arm" && current_cpu != "mipsel") {
+ if (target_cpu != "arm" && target_cpu != "mipsel") {
sources += [
"signal_processing/complex_bit_reverse.c",
"signal_processing/filter_ar_fast_q12.c",
@@ -231,7 +231,7 @@ rtc_library("fir_filter_factory") {
"../rtc_base/system:arch",
"../system_wrappers",
]
- if (current_cpu == "x86" || current_cpu == "x64") {
+ if (target_cpu == "x86" || target_cpu == "x64") {
deps += [ ":common_audio_sse2" ]
deps += [ ":common_audio_avx2" ]
}
@@ -240,7 +240,7 @@ rtc_library("fir_filter_factory") {
}
}
-if (current_cpu == "x86" || current_cpu == "x64") {
+if (target_cpu == "x86" || target_cpu == "x64") {
rtc_library("common_audio_sse2") {
sources = [
"fir_filter_sse.cc",
@@ -289,7 +289,7 @@ if (rtc_build_with_neon) {
"resampler/sinc_resampler_neon.cc",
]
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
@@ -312,7 +312,7 @@ if (rtc_build_with_neon) {
"signal_processing/min_max_operations_neon.c",
]
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
diff --git a/common_audio/third_party/ooura/BUILD.gn b/common_audio/third_party/ooura/BUILD.gn
index 0cdf98e591..a0ddf777db 100644
--- a/common_audio/third_party/ooura/BUILD.gn
+++ b/common_audio/third_party/ooura/BUILD.gn
@@ -20,7 +20,7 @@ rtc_library("fft_size_128") {
]
cflags = []
- if (current_cpu == "x86" || current_cpu == "x64") {
+ if (target_cpu == "x86" || target_cpu == "x64") {
sources += [
"fft_size_128/ooura_fft_sse2.cc",
"fft_size_128/ooura_fft_tables_neon_sse2.h",
@@ -38,14 +38,14 @@ rtc_library("fft_size_128") {
deps += [ "../../../common_audio" ]
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags += [ "-mfpu=neon" ]
}
}
- if (current_cpu == "mipsel" && mips_float_abi == "hard") {
+ if (target_cpu == "mipsel" && mips_float_abi == "hard") {
sources += [ "fft_size_128/ooura_fft_mips.cc" ]
}
}
diff --git a/common_audio/third_party/spl_sqrt_floor/BUILD.gn b/common_audio/third_party/spl_sqrt_floor/BUILD.gn
index ac862c65a8..e66ed2796e 100644
--- a/common_audio/third_party/spl_sqrt_floor/BUILD.gn
+++ b/common_audio/third_party/spl_sqrt_floor/BUILD.gn
@@ -12,11 +12,11 @@ rtc_library("spl_sqrt_floor") {
visibility = [ "../..:common_audio_c" ]
sources = [ "spl_sqrt_floor.h" ]
deps = []
- if (current_cpu == "arm") {
+ if (target_cpu == "arm") {
sources += [ "spl_sqrt_floor_arm.S" ]
deps += [ "../../../rtc_base/system:asm_defines" ]
- } else if (current_cpu == "mipsel") {
+ } else if (target_cpu == "mipsel") {
sources += [ "spl_sqrt_floor_mips.c" ]
} else {
sources += [ "spl_sqrt_floor.c" ]
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index 3e11a245a1..c29b893b7d 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -123,7 +123,7 @@ rtc_library("aec3") {
]
defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
+ if (rtc_build_with_neon && target_cpu != "arm64") {
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
@@ -162,7 +162,7 @@ rtc_library("aec3") {
"//third_party/abseil-cpp/absl/types:optional",
]
- if (current_cpu == "x86" || current_cpu == "x64") {
+ if (target_cpu == "x86" || target_cpu == "x64") {
deps += [ ":aec3_avx2" ]
}
}
@@ -253,7 +253,7 @@ rtc_source_set("fft_data") {
]
}
-if (current_cpu == "x86" || current_cpu == "x64") {
+if (target_cpu == "x86" || target_cpu == "x64") {
rtc_library("aec3_avx2") {
configs += [ "..:apm_debug_dump" ]
sources = [
diff --git a/modules/audio_processing/aecm/BUILD.gn b/modules/audio_processing/aecm/BUILD.gn
index 80f2901049..a77f04aba5 100644
--- a/modules/audio_processing/aecm/BUILD.gn
+++ b/modules/audio_processing/aecm/BUILD.gn
@@ -29,14 +29,14 @@ rtc_library("aecm_core") {
if (rtc_build_with_neon) {
sources += [ "aecm_core_neon.cc" ]
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags += [ "-mfpu=neon" ]
}
}
- if (current_cpu == "mipsel") {
+ if (target_cpu == "mipsel") {
sources += [ "aecm_core_mips.cc" ]
} else {
sources += [ "aecm_core_c.cc" ]
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index 508f901b08..75bef1450f 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -83,7 +83,7 @@ rtc_library("legacy_agc") {
]
if (rtc_build_with_neon) {
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 3003a585bd..d709eb3699 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -18,7 +18,7 @@ rtc_library("rnn_vad") {
]
defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
+ if (rtc_build_with_neon && target_cpu != "arm64") {
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn
index d818e23f3c..8c2e9dba84 100644
--- a/modules/audio_processing/ns/BUILD.gn
+++ b/modules/audio_processing/ns/BUILD.gn
@@ -43,7 +43,7 @@ rtc_static_library("ns") {
]
defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
+ if (rtc_build_with_neon && target_cpu != "arm64") {
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index 4aa9186527..060d4e8200 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -10,7 +10,7 @@ import("//build/config/linux/gtk/gtk.gni")
import("//build/config/ui.gni")
import("../../webrtc.gni")
-use_desktop_capture_differ_sse2 = current_cpu == "x86" || current_cpu == "x64"
+use_desktop_capture_differ_sse2 = target_cpu == "x86" || target_cpu == "x64"
config("x11_config") {
if (rtc_use_x11_extensions) {
diff --git a/webrtc.gni b/webrtc.gni
index 6ae1b2329c..e23c9a1cc4 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -167,13 +167,13 @@ declare_args() {
# Selects fixed-point code where possible.
rtc_prefer_fixed_point = false
- if (current_cpu == "arm" || current_cpu == "arm64") {
+ if (target_cpu == "arm" || target_cpu == "arm64") {
rtc_prefer_fixed_point = true
}
# Determines whether NEON code will be built.
rtc_build_with_neon =
- (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64"
+ (target_cpu == "arm" && arm_use_neon) || target_cpu == "arm64"
# Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on
# all platforms except Android and iOS. Because FFmpeg can be built
--
2.34.1

539
third_party/libwebrtc/moz-patch-stack/0049.patch поставляемый
Просмотреть файл

@ -1,300 +1,297 @@
From: Nico Grunbaum <na-g@nostrum.com>
Date: Wed, 14 Jul 2021 22:26:00 +0000
Subject: Bug 1654112 - deconflate the target and host architectures in
libwebrtc build files; r=mjf
From: "Byron Campen [:bwc]" <docfaraday@gmail.com>
Date: Fri, 19 Feb 2021 15:56:00 -0600
Subject: Bug 1654112 - Get RTCP BYE and RTP timeout handling working again
(from Bug 1595479) r=mjf,dminor
Differential Revision: https://phabricator.services.mozilla.com/D119707
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/58f47eacaf10d12e21dff7362743b6f4cdd1696b
Differential Revision: https://phabricator.services.mozilla.com/D106145
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d0b311007c033e83824f5f6996a70ab9e870f31f
---
BUILD.gn | 6 +++---
common_audio/BUILD.gn | 16 ++++++++--------
common_audio/third_party/ooura/BUILD.gn | 6 +++---
common_audio/third_party/spl_sqrt_floor/BUILD.gn | 4 ++--
modules/audio_processing/aec3/BUILD.gn | 6 +++---
modules/audio_processing/aecm/BUILD.gn | 4 ++--
modules/audio_processing/agc/BUILD.gn | 2 +-
modules/audio_processing/agc2/rnn_vad/BUILD.gn | 2 +-
modules/audio_processing/ns/BUILD.gn | 2 +-
modules/desktop_capture/BUILD.gn | 2 +-
webrtc.gni | 4 ++--
11 files changed, 27 insertions(+), 27 deletions(-)
audio/audio_receive_stream.cc | 5 ++++-
audio/channel_receive.cc | 13 +++++++++----
audio/channel_receive.h | 3 ++-
call/audio_receive_stream.h | 3 +++
call/video_receive_stream.cc | 2 ++
call/video_receive_stream.h | 3 +++
modules/rtp_rtcp/include/rtp_rtcp_defines.h | 8 ++++++++
modules/rtp_rtcp/source/rtcp_receiver.cc | 18 ++++++++++++++++--
modules/rtp_rtcp/source/rtcp_receiver.h | 1 +
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 3 +++
video/rtp_video_stream_receiver2.cc | 7 +++++--
11 files changed, 56 insertions(+), 10 deletions(-)
diff --git a/BUILD.gn b/BUILD.gn
index 02e9d1d4a0..4e2a1f1e70 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -407,12 +407,12 @@ config("common_config") {
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 0bb1168384..7063f40186 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -47,6 +47,8 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const {
}
}
ss << ']';
+ ss << ", rtcp_event_observer: "
+ << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr");
ss << '}';
return ss.str();
}
@@ -81,7 +83,8 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
config.enable_non_sender_rtt, config.decoder_factory,
config.codec_pair_id, std::move(config.frame_decryptor),
- config.crypto_options, std::move(config.frame_transformer));
+ config.crypto_options, std::move(config.frame_transformer),
+ config.rtp.rtcp_event_observer);
}
} // namespace
- if (current_cpu == "arm64") {
+ if (target_cpu == "arm64") {
defines += [ "WEBRTC_ARCH_ARM64" ]
defines += [ "WEBRTC_HAS_NEON" ]
}
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index b95d98c20c..50bc94fe1f 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -102,7 +102,8 @@ class ChannelReceive : public ChannelReceiveInterface,
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer);
~ChannelReceive() override;
- if (current_cpu == "arm") {
+ if (target_cpu == "arm") {
defines += [ "WEBRTC_ARCH_ARM" ]
if (arm_version >= 7) {
defines += [ "WEBRTC_ARCH_ARM_V7" ]
@@ -422,7 +422,7 @@ config("common_config") {
}
}
void SetSink(AudioSinkInterface* sink) override;
@@ -541,7 +542,8 @@ ChannelReceive::ChannelReceive(
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer)
: worker_thread_(TaskQueueBase::Current()),
event_log_(rtc_event_log),
rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
@@ -586,6 +588,7 @@ ChannelReceive::ChannelReceive(
configuration.local_media_ssrc = local_ssrc;
configuration.rtcp_packet_type_counter_observer = this;
configuration.non_sender_rtt_measurement = enable_non_sender_rtt;
+ configuration.rtcp_event_observer = rtcp_event_observer;
- if (current_cpu == "mipsel") {
+ if (target_cpu == "mipsel") {
defines += [ "MIPS32_LE" ]
if (mips_float_abi == "hard") {
defines += [ "MIPS_FPU_LE" ]
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index a45214f754..79d9321bbd 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -66,7 +66,7 @@ rtc_library("common_audio") {
deps += [ ":common_audio_neon" ]
}
- if (current_cpu == "x86" || current_cpu == "x64") {
+ if (target_cpu == "x86" || target_cpu == "x64") {
deps += [ ":common_audio_sse2" ]
deps += [ ":common_audio_avx2" ]
}
@@ -88,7 +88,7 @@ rtc_source_set("mock_common_audio") {
rtc_source_set("common_audio_c_arm_asm") {
sources = []
deps = []
- if (current_cpu == "arm") {
+ if (target_cpu == "arm") {
sources += [ "signal_processing/complex_bit_reverse_arm.S" ]
if (arm_version >= 7) {
@@ -152,7 +152,7 @@ rtc_library("common_audio_c") {
"vad/webrtc_vad.c",
]
- if (current_cpu == "mipsel") {
+ if (target_cpu == "mipsel") {
sources += [
"signal_processing/complex_bit_reverse_mips.c",
"signal_processing/complex_fft_mips.c",
@@ -170,7 +170,7 @@ rtc_library("common_audio_c") {
sources += [ "signal_processing/complex_fft.c" ]
}
- if (current_cpu != "arm" && current_cpu != "mipsel") {
+ if (target_cpu != "arm" && target_cpu != "mipsel") {
sources += [
"signal_processing/complex_bit_reverse.c",
"signal_processing/filter_ar_fast_q12.c",
@@ -231,7 +231,7 @@ rtc_library("fir_filter_factory") {
"../rtc_base/system:arch",
"../system_wrappers",
]
- if (current_cpu == "x86" || current_cpu == "x64") {
+ if (target_cpu == "x86" || target_cpu == "x64") {
deps += [ ":common_audio_sse2" ]
deps += [ ":common_audio_avx2" ]
}
@@ -240,7 +240,7 @@ rtc_library("fir_filter_factory") {
}
if (frame_transformer)
InitFrameTransformerDelegate(std::move(frame_transformer));
@@ -1119,13 +1122,15 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer) {
return std::make_unique<ChannelReceive>(
clock, neteq_factory, audio_device_module, rtcp_send_transport,
rtc_event_log, local_ssrc, remote_ssrc, jitter_buffer_max_packets,
jitter_buffer_fast_playout, jitter_buffer_min_delay_ms,
enable_non_sender_rtt, decoder_factory, codec_pair_id,
- std::move(frame_decryptor), crypto_options, std::move(frame_transformer));
+ std::move(frame_decryptor), crypto_options, std::move(frame_transformer),
+ rtcp_event_observer);
}
-if (current_cpu == "x86" || current_cpu == "x64") {
+if (target_cpu == "x86" || target_cpu == "x64") {
rtc_library("common_audio_sse2") {
sources = [
"fir_filter_sse.cc",
@@ -289,7 +289,7 @@ if (rtc_build_with_neon) {
"resampler/sinc_resampler_neon.cc",
]
} // namespace voe
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index b47a4b5b97..dd3ca1af83 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -186,7 +186,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer);
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
@@ -312,7 +312,7 @@ if (rtc_build_with_neon) {
"signal_processing/min_max_operations_neon.c",
]
} // namespace voe
} // namespace webrtc
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 1228861c42..6fc93b2d9a 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -19,6 +19,7 @@
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/transport.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_parameters.h"
#include "call/receive_stream.h"
@@ -117,6 +118,8 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
diff --git a/common_audio/third_party/ooura/BUILD.gn b/common_audio/third_party/ooura/BUILD.gn
index 0cdf98e591..a0ddf777db 100644
--- a/common_audio/third_party/ooura/BUILD.gn
+++ b/common_audio/third_party/ooura/BUILD.gn
@@ -20,7 +20,7 @@ rtc_library("fft_size_128") {
]
cflags = []
// See NackConfig for description.
NackConfig nack;
+
+ RtcpEventObserver* rtcp_event_observer = nullptr;
} rtp;
- if (current_cpu == "x86" || current_cpu == "x64") {
+ if (target_cpu == "x86" || target_cpu == "x64") {
sources += [
"fft_size_128/ooura_fft_sse2.cc",
"fft_size_128/ooura_fft_tables_neon_sse2.h",
@@ -38,14 +38,14 @@ rtc_library("fft_size_128") {
deps += [ "../../../common_audio" ]
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags += [ "-mfpu=neon" ]
}
}
- if (current_cpu == "mipsel" && mips_float_abi == "hard") {
+ if (target_cpu == "mipsel" && mips_float_abi == "hard") {
sources += [ "fft_size_128/ooura_fft_mips.cc" ]
// Receive-side RTT.
diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc
index 87df97cbdd..838dfcf135 100644
--- a/call/video_receive_stream.cc
+++ b/call/video_receive_stream.cc
@@ -153,6 +153,8 @@ std::string VideoReceiveStreamInterface::Config::Rtp::ToString() const {
ss << ", ";
}
ss << ']';
+ ss << ", rtcp_event_observer: "
+ << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr");
ss << '}';
return ss.str();
}
diff --git a/common_audio/third_party/spl_sqrt_floor/BUILD.gn b/common_audio/third_party/spl_sqrt_floor/BUILD.gn
index ac862c65a8..e66ed2796e 100644
--- a/common_audio/third_party/spl_sqrt_floor/BUILD.gn
+++ b/common_audio/third_party/spl_sqrt_floor/BUILD.gn
@@ -12,11 +12,11 @@ rtc_library("spl_sqrt_floor") {
visibility = [ "../..:common_audio_c" ]
sources = [ "spl_sqrt_floor.h" ]
deps = []
- if (current_cpu == "arm") {
+ if (target_cpu == "arm") {
sources += [ "spl_sqrt_floor_arm.S" ]
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index cda8b1f6af..eeb7d14cc3 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -19,6 +19,7 @@
#include <vector>
deps += [ "../../../rtc_base/system:asm_defines" ]
- } else if (current_cpu == "mipsel") {
+ } else if (target_cpu == "mipsel") {
sources += [ "spl_sqrt_floor_mips.c" ]
} else {
sources += [ "spl_sqrt_floor.c" ]
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index 3e11a245a1..c29b893b7d 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -123,7 +123,7 @@ rtc_library("aec3") {
]
#include "api/call/transport.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
@@ -234,6 +235,8 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
// meta data is expected to be present in generic frame descriptor
// RTP header extension).
std::set<int> raw_payload_types;
+
+ RtcpEventObserver* rtcp_event_observer = nullptr;
} rtp;
defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
+ if (rtc_build_with_neon && target_cpu != "arm64") {
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
// Transport for outgoing packets (RTCP).
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 43bba3e57a..882f861d0b 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -211,6 +211,14 @@ class RtcpBandwidthObserver {
virtual ~RtcpBandwidthObserver() {}
};
+class RtcpEventObserver {
+ public:
+ virtual void OnRtcpBye() = 0;
+ virtual void OnRtcpTimeout() = 0;
+
+ virtual ~RtcpEventObserver() {}
+};
+
// NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType!
static constexpr size_t kNumMediaTypes = 5;
enum class RtpPacketMediaType : size_t {
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 68171d1c2a..69d62ead5a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -145,6 +145,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
rtp_rtcp_(owner),
registered_ssrcs_(false, config),
rtcp_bandwidth_observer_(config.bandwidth_callback),
+ rtcp_event_observer_(config.rtcp_event_observer),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
network_state_estimate_observer_(config.network_state_estimate_observer),
@@ -178,6 +179,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
rtp_rtcp_(owner),
registered_ssrcs_(true, config),
rtcp_bandwidth_observer_(config.bandwidth_callback),
+ rtcp_event_observer_(config.rtcp_event_observer),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
network_state_estimate_observer_(config.network_state_estimate_observer),
@@ -848,6 +850,10 @@ void RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) {
return;
}
@@ -162,7 +162,7 @@ rtc_library("aec3") {
"//third_party/abseil-cpp/absl/types:optional",
]
- if (current_cpu == "x86" || current_cpu == "x64") {
+ if (target_cpu == "x86" || target_cpu == "x64") {
deps += [ ":aec3_avx2" ]
}
}
@@ -253,7 +253,7 @@ rtc_source_set("fft_data") {
]
+ if (rtcp_event_observer_) {
+ rtcp_event_observer_->OnRtcpBye();
+ }
+
// Clear our lists.
rtts_.erase(bye.sender_ssrc());
EraseIf(received_report_blocks_, [&](const auto& elem) {
@@ -1265,12 +1271,20 @@ std::vector<rtcp::TmmbItem> RTCPReceiver::TmmbrReceived() {
}
-if (current_cpu == "x86" || current_cpu == "x64") {
+if (target_cpu == "x86" || target_cpu == "x64") {
rtc_library("aec3_avx2") {
configs += [ "..:apm_debug_dump" ]
sources = [
diff --git a/modules/audio_processing/aecm/BUILD.gn b/modules/audio_processing/aecm/BUILD.gn
index 80f2901049..a77f04aba5 100644
--- a/modules/audio_processing/aecm/BUILD.gn
+++ b/modules/audio_processing/aecm/BUILD.gn
@@ -29,14 +29,14 @@ rtc_library("aecm_core") {
if (rtc_build_with_neon) {
sources += [ "aecm_core_neon.cc" ]
bool RTCPReceiver::RtcpRrTimeoutLocked(Timestamp now) {
- return ResetTimestampIfExpired(now, last_received_rb_, report_interval_);
+ bool result = ResetTimestampIfExpired(now, last_received_rb_, report_interval_);
+ if (result && rtcp_event_observer_) {
+ rtcp_event_observer_->OnRtcpTimeout();
+ }
+ return result;
}
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags += [ "-mfpu=neon" ]
}
}
bool RTCPReceiver::RtcpRrSequenceNumberTimeoutLocked(Timestamp now) {
- return ResetTimestampIfExpired(now, last_increased_sequence_number_,
+ bool result = ResetTimestampIfExpired(now, last_increased_sequence_number_,
report_interval_);
+ if (result && rtcp_event_observer_) {
+ rtcp_event_observer_->OnRtcpTimeout();
+ }
+ return result;
}
- if (current_cpu == "mipsel") {
+ if (target_cpu == "mipsel") {
sources += [ "aecm_core_mips.cc" ]
} else {
sources += [ "aecm_core_c.cc" ]
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index 508f901b08..75bef1450f 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -83,7 +83,7 @@ rtc_library("legacy_agc") {
]
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index 6912912cfc..a05a69059a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -385,6 +385,7 @@ class RTCPReceiver final {
RegisteredSsrcs registered_ssrcs_;
if (rtc_build_with_neon) {
- if (current_cpu != "arm64") {
+ if (target_cpu != "arm64") {
# Enable compilation for the NEON instruction set.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 3003a585bd..d709eb3699 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -18,7 +18,7 @@ rtc_library("rnn_vad") {
]
RtcpBandwidthObserver* const rtcp_bandwidth_observer_;
+ RtcpEventObserver* const rtcp_event_observer_;
RtcpIntraFrameObserver* const rtcp_intra_frame_observer_;
RtcpLossNotificationObserver* const rtcp_loss_notification_observer_;
NetworkStateEstimateObserver* const network_state_estimate_observer_;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index c6854937cb..b988c7805d 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -73,6 +73,9 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
// stream.
RtcpBandwidthObserver* bandwidth_callback = nullptr;
defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
+ if (rtc_build_with_neon && target_cpu != "arm64") {
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn
index d818e23f3c..8c2e9dba84 100644
--- a/modules/audio_processing/ns/BUILD.gn
+++ b/modules/audio_processing/ns/BUILD.gn
@@ -43,7 +43,7 @@ rtc_static_library("ns") {
]
+ // Called when we receive a RTCP bye or timeout
+ RtcpEventObserver* rtcp_event_observer = nullptr;
+
NetworkStateEstimateObserver* network_state_estimate_observer = nullptr;
TransportFeedbackObserver* transport_feedback_callback = nullptr;
VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr;
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index eed9770d93..c7b5e7bc7c 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -83,7 +83,8 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
RtcpCnameCallback* rtcp_cname_callback,
bool non_sender_rtt_measurement,
uint32_t local_ssrc,
- RtcEventLog* rtc_event_log) {
+ RtcEventLog* rtc_event_log,
+ RtcpEventObserver* rtcp_event_observer) {
RtpRtcpInterface::Configuration configuration;
configuration.clock = clock;
configuration.audio = false;
@@ -95,6 +96,7 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
rtcp_packet_type_counter_observer;
configuration.rtcp_cname_callback = rtcp_cname_callback;
configuration.local_media_ssrc = local_ssrc;
+ configuration.rtcp_event_observer = rtcp_event_observer;
configuration.non_sender_rtt_measurement = non_sender_rtt_measurement;
configuration.event_log = rtc_event_log;
defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
+ if (rtc_build_with_neon && target_cpu != "arm64") {
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index 4aa9186527..060d4e8200 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -10,7 +10,7 @@ import("//build/config/linux/gtk/gtk.gni")
import("//build/config/ui.gni")
import("../../webrtc.gni")
-use_desktop_capture_differ_sse2 = current_cpu == "x86" || current_cpu == "x64"
+use_desktop_capture_differ_sse2 = target_cpu == "x86" || target_cpu == "x64"
config("x11_config") {
if (rtc_use_x11_extensions) {
diff --git a/webrtc.gni b/webrtc.gni
index 6ae1b2329c..e23c9a1cc4 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -167,13 +167,13 @@ declare_args() {
# Selects fixed-point code where possible.
rtc_prefer_fixed_point = false
- if (current_cpu == "arm" || current_cpu == "arm64") {
+ if (target_cpu == "arm" || target_cpu == "arm64") {
rtc_prefer_fixed_point = true
}
# Determines whether NEON code will be built.
rtc_build_with_neon =
- (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64"
+ (target_cpu == "arm" && arm_use_neon) || target_cpu == "arm64"
# Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on
# all platforms except Android and iOS. Because FFmpeg can be built
@@ -276,7 +278,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
rtcp_cname_callback,
config_.rtp.rtcp_xr.receiver_reference_time_report,
config_.rtp.local_ssrc,
- event_log)),
+ event_log,
+ config_.rtp.rtcp_event_observer)),
nack_periodic_processor_(nack_periodic_processor),
complete_frame_callback_(complete_frame_callback),
keyframe_request_method_(config_.rtp.keyframe_method),
--
2.34.1

305
third_party/libwebrtc/moz-patch-stack/0050.patch поставляемый
Просмотреть файл

@ -1,297 +1,32 @@
From: "Byron Campen [:bwc]" <docfaraday@gmail.com>
Date: Fri, 19 Feb 2021 15:56:00 -0600
Subject: Bug 1654112 - Get RTCP BYE and RTP timeout handling working again
(from Bug 1595479) r=mjf,dminor
Date: Fri, 12 Mar 2021 08:53:00 -0600
Subject: Bug 1654112 - libwebrtc modification: Init some stats that were being
passed to us uninitialized. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D106145
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d0b311007c033e83824f5f6996a70ab9e870f31f
Differential Revision: https://phabricator.services.mozilla.com/D108673
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c385bb870413b925af48df97aa1f2b80a26e78d2
---
audio/audio_receive_stream.cc | 5 ++++-
audio/channel_receive.cc | 13 +++++++++----
audio/channel_receive.h | 3 ++-
call/audio_receive_stream.h | 3 +++
call/video_receive_stream.cc | 2 ++
call/video_receive_stream.h | 3 +++
modules/rtp_rtcp/include/rtp_rtcp_defines.h | 8 ++++++++
modules/rtp_rtcp/source/rtcp_receiver.cc | 18 ++++++++++++++++--
modules/rtp_rtcp/source/rtcp_receiver.h | 1 +
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 3 +++
video/rtp_video_stream_receiver2.cc | 7 +++++--
11 files changed, 56 insertions(+), 10 deletions(-)
call/video_receive_stream.h | 7 ++++---
1 file changed, 4 insertions(+), 3 deletions(-)
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 0bb1168384..7063f40186 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -47,6 +47,8 @@ std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const {
}
}
ss << ']';
+ ss << ", rtcp_event_observer: "
+ << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr");
ss << '}';
return ss.str();
}
@@ -81,7 +83,8 @@ std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
config.enable_non_sender_rtt, config.decoder_factory,
config.codec_pair_id, std::move(config.frame_decryptor),
- config.crypto_options, std::move(config.frame_transformer));
+ config.crypto_options, std::move(config.frame_transformer),
+ config.rtp.rtcp_event_observer);
}
} // namespace
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index b95d98c20c..50bc94fe1f 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -102,7 +102,8 @@ class ChannelReceive : public ChannelReceiveInterface,
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer);
~ChannelReceive() override;
void SetSink(AudioSinkInterface* sink) override;
@@ -541,7 +542,8 @@ ChannelReceive::ChannelReceive(
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer)
: worker_thread_(TaskQueueBase::Current()),
event_log_(rtc_event_log),
rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
@@ -586,6 +588,7 @@ ChannelReceive::ChannelReceive(
configuration.local_media_ssrc = local_ssrc;
configuration.rtcp_packet_type_counter_observer = this;
configuration.non_sender_rtt_measurement = enable_non_sender_rtt;
+ configuration.rtcp_event_observer = rtcp_event_observer;
if (frame_transformer)
InitFrameTransformerDelegate(std::move(frame_transformer));
@@ -1119,13 +1122,15 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer) {
return std::make_unique<ChannelReceive>(
clock, neteq_factory, audio_device_module, rtcp_send_transport,
rtc_event_log, local_ssrc, remote_ssrc, jitter_buffer_max_packets,
jitter_buffer_fast_playout, jitter_buffer_min_delay_ms,
enable_non_sender_rtt, decoder_factory, codec_pair_id,
- std::move(frame_decryptor), crypto_options, std::move(frame_transformer));
+ std::move(frame_decryptor), crypto_options, std::move(frame_transformer),
+ rtcp_event_observer);
}
} // namespace voe
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index b47a4b5b97..dd3ca1af83 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -186,7 +186,8 @@ std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
absl::optional<AudioCodecPairId> codec_pair_id,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
const webrtc::CryptoOptions& crypto_options,
- rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+ rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+ RtcpEventObserver* rtcp_event_observer);
} // namespace voe
} // namespace webrtc
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 1228861c42..6fc93b2d9a 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -19,6 +19,7 @@
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/transport.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_parameters.h"
#include "call/receive_stream.h"
@@ -117,6 +118,8 @@ class AudioReceiveStreamInterface : public MediaReceiveStreamInterface {
// See NackConfig for description.
NackConfig nack;
+
+ RtcpEventObserver* rtcp_event_observer = nullptr;
} rtp;
// Receive-side RTT.
diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc
index 87df97cbdd..838dfcf135 100644
--- a/call/video_receive_stream.cc
+++ b/call/video_receive_stream.cc
@@ -153,6 +153,8 @@ std::string VideoReceiveStreamInterface::Config::Rtp::ToString() const {
ss << ", ";
}
ss << ']';
+ ss << ", rtcp_event_observer: "
+ << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr");
ss << '}';
return ss.str();
}
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index cda8b1f6af..eeb7d14cc3 100644
index eeb7d14cc3..31c9bff7b0 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -19,6 +19,7 @@
#include <vector>
@@ -145,9 +145,10 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
RtpReceiveStats rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
#include "api/call/transport.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "api/crypto/crypto_options.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
@@ -234,6 +235,8 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
// meta data is expected to be present in generic frame descriptor
// RTP header extension).
std::set<int> raw_payload_types;
+
+ RtcpEventObserver* rtcp_event_observer = nullptr;
} rtp;
- uint32_t rtcp_sender_packets_sent;
- uint32_t rtcp_sender_octets_sent;
- int64_t rtcp_sender_ntp_timestamp_ms;
+ // Mozilla modification: Init these three.
+ uint32_t rtcp_sender_packets_sent = 0;
+ uint32_t rtcp_sender_octets_sent = 0;
+ int64_t rtcp_sender_ntp_timestamp_ms = 0;
// Transport for outgoing packets (RTCP).
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 43bba3e57a..882f861d0b 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -211,6 +211,14 @@ class RtcpBandwidthObserver {
virtual ~RtcpBandwidthObserver() {}
};
+class RtcpEventObserver {
+ public:
+ virtual void OnRtcpBye() = 0;
+ virtual void OnRtcpTimeout() = 0;
+
+ virtual ~RtcpEventObserver() {}
+};
+
// NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType!
static constexpr size_t kNumMediaTypes = 5;
enum class RtpPacketMediaType : size_t {
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 68171d1c2a..69d62ead5a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -145,6 +145,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
rtp_rtcp_(owner),
registered_ssrcs_(false, config),
rtcp_bandwidth_observer_(config.bandwidth_callback),
+ rtcp_event_observer_(config.rtcp_event_observer),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
network_state_estimate_observer_(config.network_state_estimate_observer),
@@ -178,6 +179,7 @@ RTCPReceiver::RTCPReceiver(const RtpRtcpInterface::Configuration& config,
rtp_rtcp_(owner),
registered_ssrcs_(true, config),
rtcp_bandwidth_observer_(config.bandwidth_callback),
+ rtcp_event_observer_(config.rtcp_event_observer),
rtcp_intra_frame_observer_(config.intra_frame_callback),
rtcp_loss_notification_observer_(config.rtcp_loss_notification_observer),
network_state_estimate_observer_(config.network_state_estimate_observer),
@@ -848,6 +850,10 @@ void RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) {
return;
}
+ if (rtcp_event_observer_) {
+ rtcp_event_observer_->OnRtcpBye();
+ }
+
// Clear our lists.
rtts_.erase(bye.sender_ssrc());
EraseIf(received_report_blocks_, [&](const auto& elem) {
@@ -1265,12 +1271,20 @@ std::vector<rtcp::TmmbItem> RTCPReceiver::TmmbrReceived() {
}
bool RTCPReceiver::RtcpRrTimeoutLocked(Timestamp now) {
- return ResetTimestampIfExpired(now, last_received_rb_, report_interval_);
+ bool result = ResetTimestampIfExpired(now, last_received_rb_, report_interval_);
+ if (result && rtcp_event_observer_) {
+ rtcp_event_observer_->OnRtcpTimeout();
+ }
+ return result;
}
bool RTCPReceiver::RtcpRrSequenceNumberTimeoutLocked(Timestamp now) {
- return ResetTimestampIfExpired(now, last_increased_sequence_number_,
+ bool result = ResetTimestampIfExpired(now, last_increased_sequence_number_,
report_interval_);
+ if (result && rtcp_event_observer_) {
+ rtcp_event_observer_->OnRtcpTimeout();
+ }
+ return result;
}
} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index 6912912cfc..a05a69059a 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -385,6 +385,7 @@ class RTCPReceiver final {
RegisteredSsrcs registered_ssrcs_;
RtcpBandwidthObserver* const rtcp_bandwidth_observer_;
+ RtcpEventObserver* const rtcp_event_observer_;
RtcpIntraFrameObserver* const rtcp_intra_frame_observer_;
RtcpLossNotificationObserver* const rtcp_loss_notification_observer_;
NetworkStateEstimateObserver* const network_state_estimate_observer_;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index c6854937cb..b988c7805d 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -73,6 +73,9 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
// stream.
RtcpBandwidthObserver* bandwidth_callback = nullptr;
+ // Called when we receive a RTCP bye or timeout
+ RtcpEventObserver* rtcp_event_observer = nullptr;
+
NetworkStateEstimateObserver* network_state_estimate_observer = nullptr;
TransportFeedbackObserver* transport_feedback_callback = nullptr;
VideoBitrateAllocationObserver* bitrate_allocation_observer = nullptr;
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index eed9770d93..c7b5e7bc7c 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -83,7 +83,8 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
RtcpCnameCallback* rtcp_cname_callback,
bool non_sender_rtt_measurement,
uint32_t local_ssrc,
- RtcEventLog* rtc_event_log) {
+ RtcEventLog* rtc_event_log,
+ RtcpEventObserver* rtcp_event_observer) {
RtpRtcpInterface::Configuration configuration;
configuration.clock = clock;
configuration.audio = false;
@@ -95,6 +96,7 @@ std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule(
rtcp_packet_type_counter_observer;
configuration.rtcp_cname_callback = rtcp_cname_callback;
configuration.local_media_ssrc = local_ssrc;
+ configuration.rtcp_event_observer = rtcp_event_observer;
configuration.non_sender_rtt_measurement = non_sender_rtt_measurement;
configuration.event_log = rtc_event_log;
@@ -276,7 +278,8 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
rtcp_cname_callback,
config_.rtp.rtcp_xr.receiver_reference_time_report,
config_.rtp.local_ssrc,
- event_log)),
+ event_log,
+ config_.rtp.rtcp_event_observer)),
nack_periodic_processor_(nack_periodic_processor),
complete_frame_callback_(complete_frame_callback),
keyframe_request_method_(config_.rtp.keyframe_method),
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
--
2.34.1

90
third_party/libwebrtc/moz-patch-stack/0051.patch поставляемый
Просмотреть файл

@ -1,32 +1,76 @@
From: "Byron Campen [:bwc]" <docfaraday@gmail.com>
Date: Fri, 12 Mar 2021 08:53:00 -0600
Subject: Bug 1654112 - libwebrtc modification: Init some stats that were being
passed to us uninitialized. r=ng
Date: Fri, 12 Mar 2021 08:55:00 -0600
Subject: Bug 1654112 - libwebrtc modification: Surface video RTCP SR stats
again. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D108673
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c385bb870413b925af48df97aa1f2b80a26e78d2
libwebrtc has stopped surfacing these, and Chromium does not support
these stats at all.
Differential Revision: https://phabricator.services.mozilla.com/D108674
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/edac9d01a9ac7594f4b22708a4690753638ca32c
---
call/video_receive_stream.h | 7 ++++---
1 file changed, 4 insertions(+), 3 deletions(-)
video/rtp_video_stream_receiver2.cc | 10 ++++++++++
video/rtp_video_stream_receiver2.h | 6 ++++++
video/video_receive_stream2.cc | 8 ++++++++
3 files changed, 24 insertions(+)
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index eeb7d14cc3..31c9bff7b0 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -145,9 +145,10 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
RtpReceiveStats rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index c7b5e7bc7c..05447ac3bd 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -1062,6 +1062,16 @@ absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
return absl::nullopt;
}
- uint32_t rtcp_sender_packets_sent;
- uint32_t rtcp_sender_octets_sent;
- int64_t rtcp_sender_ntp_timestamp_ms;
+ // Mozilla modification: Init these three.
+ uint32_t rtcp_sender_packets_sent = 0;
+ uint32_t rtcp_sender_octets_sent = 0;
+ int64_t rtcp_sender_ntp_timestamp_ms = 0;
+// Mozilla modification: VideoReceiveStream2 and friends do not surface RTCP
+// stats at all, and even on the most recent libwebrtc code there does not
+// seem to be any support for these stats right now. So, we hack this in.
+void RtpVideoStreamReceiver2::RemoteRTCPSenderInfo(
+ uint32_t* packet_count, uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const {
+ RTC_DCHECK_RUN_ON(&worker_task_checker_);
+ rtp_rtcp_->RemoteRTCPSenderInfo(packet_count, octet_count, ntp_timestamp_ms);
+}
+
void RtpVideoStreamReceiver2::ManageFrame(
std::unique_ptr<RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 48bcab157e..21f125ae2f 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -211,6 +211,12 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
absl::optional<int64_t> LastReceivedPacketMs() const;
absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
+ // Mozilla modification: VideoReceiveStream2 and friends do not surface RTCP
+ // stats at all, and even on the most recent libwebrtc code there does not
+ // seem to be any support for these stats right now. So, we hack this in.
+ void RemoteRTCPSenderInfo(uint32_t* packet_count, uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const;
+
private:
// Implements RtpVideoFrameReceiver.
void ManageFrame(std::unique_ptr<RtpFrameObject> frame) override;
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index ce96512795..be850834d6 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -597,6 +597,14 @@ VideoReceiveStreamInterface::Stats VideoReceiveStream2::GetStats() const {
if (rtx_statistician)
stats.total_bitrate_bps += rtx_statistician->BitrateReceived();
}
+
+ // Mozilla modification: VideoReceiveStream2 and friends do not surface RTCP
+ // stats at all, and even on the most recent libwebrtc code there does not
+ // seem to be any support for these stats right now. So, we hack this in.
+ rtp_video_stream_receiver_.RemoteRTCPSenderInfo(
+ &stats.rtcp_sender_packets_sent, &stats.rtcp_sender_octets_sent,
+ &stats.rtcp_sender_ntp_timestamp_ms);
+
return stats;
}
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
--
2.34.1

92
third_party/libwebrtc/moz-patch-stack/0052.patch поставляемый
Просмотреть файл

@ -1,74 +1,36 @@
From: "Byron Campen [:bwc]" <docfaraday@gmail.com>
Date: Fri, 12 Mar 2021 08:55:00 -0600
Subject: Bug 1654112 - libwebrtc modification: Surface video RTCP SR stats
again. r=ng
From: Nico Grunbaum <na-g@nostrum.com>
Date: Mon, 26 Jul 2021 22:51:00 -0700
Subject: Bug 1654112 - fix timestamp issues with RTP sources; r=mjf
libwebrtc has stopped surfacing these, and Chromium does not support
these stats at all.
Differential Revision: https://phabricator.services.mozilla.com/D108674
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/edac9d01a9ac7594f4b22708a4690753638ca32c
Differential Revision: https://phabricator.services.mozilla.com/D120930
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/de8c14e4972f717bf937b6f2fffcd08c35e21ced
---
video/rtp_video_stream_receiver2.cc | 10 ++++++++++
video/rtp_video_stream_receiver2.h | 6 ++++++
video/video_receive_stream2.cc | 8 ++++++++
3 files changed, 24 insertions(+)
modules/rtp_rtcp/source/source_tracker.cc | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index c7b5e7bc7c..05447ac3bd 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -1062,6 +1062,16 @@ absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
return absl::nullopt;
}
diff --git a/modules/rtp_rtcp/source/source_tracker.cc b/modules/rtp_rtcp/source/source_tracker.cc
index 7a5cbac77d..65f21700d0 100644
--- a/modules/rtp_rtcp/source/source_tracker.cc
+++ b/modules/rtp_rtcp/source/source_tracker.cc
@@ -36,7 +36,8 @@ void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
SourceKey key(RtpSourceType::CSRC, csrc);
SourceEntry& entry = UpdateEntry(key);
+// Mozilla modification: VideoReceiveStream2 and friends do not surface RTCP
+// stats at all, and even on the most recent libwebrtc code there does not
+// seem to be any support for these stats right now. So, we hack this in.
+void RtpVideoStreamReceiver2::RemoteRTCPSenderInfo(
+ uint32_t* packet_count, uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const {
+ RTC_DCHECK_RUN_ON(&worker_task_checker_);
+ rtp_rtcp_->RemoteRTCPSenderInfo(packet_count, octet_count, ntp_timestamp_ms);
+}
+
void RtpVideoStreamReceiver2::ManageFrame(
std::unique_ptr<RtpFrameObject> frame) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 48bcab157e..21f125ae2f 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -211,6 +211,12 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
absl::optional<int64_t> LastReceivedPacketMs() const;
absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
+ // Mozilla modification: VideoReceiveStream2 and friends do not surface RTCP
+ // stats at all, and even on the most recent libwebrtc code there does not
+ // seem to be any support for these stats right now. So, we hack this in.
+ void RemoteRTCPSenderInfo(uint32_t* packet_count, uint32_t* octet_count,
+ int64_t* ntp_timestamp_ms) const;
+
private:
// Implements RtpVideoFrameReceiver.
void ManageFrame(std::unique_ptr<RtpFrameObject> frame) override;
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index ce96512795..be850834d6 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -597,6 +597,14 @@ VideoReceiveStreamInterface::Stats VideoReceiveStream2::GetStats() const {
if (rtx_statistician)
stats.total_bitrate_bps += rtx_statistician->BitrateReceived();
- entry.timestamp_ms = now_ms;
+ const auto packet_time = packet_info.receive_time().ms();
+ entry.timestamp_ms = packet_time ? packet_time : now_ms;
entry.audio_level = packet_info.audio_level();
entry.absolute_capture_time = packet_info.absolute_capture_time();
entry.local_capture_clock_offset =
@@ -77,6 +78,10 @@ std::vector<RtpSource> SourceTracker::GetSources() const {
.local_capture_clock_offset = entry.local_capture_clock_offset});
}
+ std::sort(sources.begin(), sources.end(), [](const auto &a, const auto &b){
+ return a.timestamp_ms() > b.timestamp_ms();
+ });
+
+ // Mozilla modification: VideoReceiveStream2 and friends do not surface RTCP
+ // stats at all, and even on the most recent libwebrtc code there does not
+ // seem to be any support for these stats right now. So, we hack this in.
+ rtp_video_stream_receiver_.RemoteRTCPSenderInfo(
+ &stats.rtcp_sender_packets_sent, &stats.rtcp_sender_octets_sent,
+ &stats.rtcp_sender_ntp_timestamp_ms);
+
return stats;
return sources;
}
--

52
third_party/libwebrtc/moz-patch-stack/0053.patch поставляемый
Просмотреть файл

@ -1,37 +1,27 @@
From: Nico Grunbaum <na-g@nostrum.com>
Date: Mon, 26 Jul 2021 22:51:00 -0700
Subject: Bug 1654112 - fix timestamp issues with RTP sources; r=mjf
From: Michael Froman <mfroman@mozilla.com>
Date: Thu, 2 Sep 2021 13:22:00 -0500
Subject: Bug 1654112 - fixes for hybrid build. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D120930
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/de8c14e4972f717bf937b6f2fffcd08c35e21ced
- adds missing includes in several places
- makes dom/media/webrtc/jsapi unified-only
Differential Revision: https://phabricator.services.mozilla.com/D124499
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/1965757ee924d49c32eab76d1a3dfa77f9eadcf4
---
modules/rtp_rtcp/source/source_tracker.cc | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
audio/channel_receive.h | 1 +
1 file changed, 1 insertion(+)
diff --git a/modules/rtp_rtcp/source/source_tracker.cc b/modules/rtp_rtcp/source/source_tracker.cc
index 7a5cbac77d..65f21700d0 100644
--- a/modules/rtp_rtcp/source/source_tracker.cc
+++ b/modules/rtp_rtcp/source/source_tracker.cc
@@ -36,7 +36,8 @@ void SourceTracker::OnFrameDelivered(const RtpPacketInfos& packet_infos) {
SourceKey key(RtpSourceType::CSRC, csrc);
SourceEntry& entry = UpdateEntry(key);
- entry.timestamp_ms = now_ms;
+ const auto packet_time = packet_info.receive_time().ms();
+ entry.timestamp_ms = packet_time ? packet_time : now_ms;
entry.audio_level = packet_info.audio_level();
entry.absolute_capture_time = packet_info.absolute_capture_time();
entry.local_capture_clock_offset =
@@ -77,6 +78,10 @@ std::vector<RtpSource> SourceTracker::GetSources() const {
.local_capture_clock_offset = entry.local_capture_clock_offset});
}
+ std::sort(sources.begin(), sources.end(), [](const auto &a, const auto &b){
+ return a.timestamp_ms() > b.timestamp_ms();
+ });
+
return sources;
}
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index dd3ca1af83..1ad3be781b 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -28,6 +28,7 @@
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/source_tracker.h"
#include "system_wrappers/include/clock.h"
--
2.34.1

44
third_party/libwebrtc/moz-patch-stack/0054.patch поставляемый
Просмотреть файл

@ -1,27 +1,31 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Thu, 2 Sep 2021 13:22:00 -0500
Subject: Bug 1654112 - fixes for hybrid build. r=ng
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Fri, 20 Aug 2021 13:52:00 +0200
Subject: Bug 1654112 - Don't check the calling thread in
webrtc::AudioReceiveStream::GetSources. r=ng
- adds missing includes in several places
- makes dom/media/webrtc/jsapi unified-only
source_tracker_ is thread safe with its own internal mutex, so this call is safe
as long as the caller has a guarantee for the lifetime of the
AudioReceiveStream. This is similar to webrtc::VideoReceiveStream.
Differential Revision: https://phabricator.services.mozilla.com/D124499
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/1965757ee924d49c32eab76d1a3dfa77f9eadcf4
Upliftable.
Differential Revision: https://phabricator.services.mozilla.com/D123226
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c186df8a088e46285a15e40149182daa34cc6805
---
audio/channel_receive.h | 1 +
1 file changed, 1 insertion(+)
audio/audio_receive_stream.cc | 1 -
1 file changed, 1 deletion(-)
diff --git a/audio/channel_receive.h b/audio/channel_receive.h
index dd3ca1af83..1ad3be781b 100644
--- a/audio/channel_receive.h
+++ b/audio/channel_receive.h
@@ -28,6 +28,7 @@
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/source_tracker.h"
#include "system_wrappers/include/clock.h"
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 7063f40186..20133e6dfe 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -389,7 +389,6 @@ int AudioReceiveStreamImpl::GetBaseMinimumPlayoutDelayMs() const {
}
std::vector<RtpSource> AudioReceiveStreamImpl::GetSources() const {
- RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return source_tracker_.GetSources();
}
--
2.34.1

46
third_party/libwebrtc/moz-patch-stack/0055.patch поставляемый
Просмотреть файл

@ -1,32 +1,30 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Fri, 20 Aug 2021 13:52:00 +0200
Subject: Bug 1654112 - Don't check the calling thread in
webrtc::AudioReceiveStream::GetSources. r=ng
From: Nico Grunbaum <na-g@nostrum.com>
Date: Thu, 28 Oct 2021 18:13:00 +0000
Subject: Bug 1729367 - P6 - Restore PID recording post cherry-pick;r=mjf
source_tracker_ is thread safe with its own internal mutex, so this call is safe
as long as the caller has a guarantee for the lifetime of the
AudioReceiveStream. This is similar to webrtc::VideoReceiveStream.
This restores the code from P0, which was removed to make cherry-picking 439ffe462a66ad9fa9a251b265e4ab28c2647d25 and 449a78b1e20ea85b11f967cf3a184ee610ce21c3 easier.
Upliftable.
Differential Revision: https://phabricator.services.mozilla.com/D123226
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c186df8a088e46285a15e40149182daa34cc6805
Differential Revision: https://phabricator.services.mozilla.com/D129714
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/72a83cb2e571023cd4150bbdef5be5455ce851f4
---
audio/audio_receive_stream.cc | 1 -
1 file changed, 1 deletion(-)
modules/desktop_capture/win/window_capture_utils.cc | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 7063f40186..20133e6dfe 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -389,7 +389,6 @@ int AudioReceiveStreamImpl::GetBaseMinimumPlayoutDelayMs() const {
}
std::vector<RtpSource> AudioReceiveStreamImpl::GetSources() const {
- RTC_DCHECK_RUN_ON(&worker_thread_checker_);
return source_tracker_.GetSources();
}
diff --git a/modules/desktop_capture/win/window_capture_utils.cc b/modules/desktop_capture/win/window_capture_utils.cc
index ccfef49bc5..d58c02e17c 100644
--- a/modules/desktop_capture/win/window_capture_utils.cc
+++ b/modules/desktop_capture/win/window_capture_utils.cc
@@ -79,6 +79,10 @@ BOOL CALLBACK GetWindowListHandler(HWND hwnd, LPARAM param) {
DesktopCapturer::Source window;
window.id = reinterpret_cast<WindowId>(hwnd);
+ DWORD pid;
+ GetWindowThreadProcessId(hwnd, &pid);
+ window.pid = static_cast<pid_t>(pid);
+
// GetWindowText* are potentially blocking operations if `hwnd` is
// owned by the current process. The APIs will send messages to the window's
// message loop, and if the message loop is waiting on this operation we will
--
2.34.1

43
third_party/libwebrtc/moz-patch-stack/0056.patch поставляемый
Просмотреть файл

@ -1,30 +1,29 @@
From: Nico Grunbaum <na-g@nostrum.com>
Date: Thu, 28 Oct 2021 18:13:00 +0000
Subject: Bug 1729367 - P6 - Restore PID recording post cherry-pick;r=mjf
Subject: Bug 1729367 - P7 - restore mac PID tracking using new API;r=mjf
a=webrtc-update
This restores the code from P0, which was removed to make cherry-picking 439ffe462a66ad9fa9a251b265e4ab28c2647d25 and 449a78b1e20ea85b11f967cf3a184ee610ce21c3 easier.
Differential Revision: https://phabricator.services.mozilla.com/D129714
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/72a83cb2e571023cd4150bbdef5be5455ce851f4
Differential Revision: https://phabricator.services.mozilla.com/D129721
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/1495ca5ef535f8ad692a3a579ca42eddc14f39a8
---
modules/desktop_capture/win/window_capture_utils.cc | 4 ++++
1 file changed, 4 insertions(+)
modules/desktop_capture/window_capturer_mac.mm | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/modules/desktop_capture/win/window_capture_utils.cc b/modules/desktop_capture/win/window_capture_utils.cc
index ccfef49bc5..d58c02e17c 100644
--- a/modules/desktop_capture/win/window_capture_utils.cc
+++ b/modules/desktop_capture/win/window_capture_utils.cc
@@ -79,6 +79,10 @@ BOOL CALLBACK GetWindowListHandler(HWND hwnd, LPARAM param) {
DesktopCapturer::Source window;
window.id = reinterpret_cast<WindowId>(hwnd);
+ DWORD pid;
+ GetWindowThreadProcessId(hwnd, &pid);
+ window.pid = static_cast<pid_t>(pid);
+
// GetWindowText* are potentially blocking operations if `hwnd` is
// owned by the current process. The APIs will send messages to the window's
// message loop, and if the message loop is waiting on this operation we will
diff --git a/modules/desktop_capture/window_capturer_mac.mm b/modules/desktop_capture/window_capturer_mac.mm
index f0b413b0a6..882498bc34 100644
--- a/modules/desktop_capture/window_capturer_mac.mm
+++ b/modules/desktop_capture/window_capturer_mac.mm
@@ -167,8 +167,9 @@ void WindowCapturerMac::CaptureFrame() {
return webrtc::GetWindowList(
[sources](CFDictionaryRef window) {
WindowId window_id = GetWindowId(window);
+ int pid = GetWindowOwnerPid(window);
if (window_id != kNullWindowId) {
- sources->push_back(DesktopCapturer::Source{window_id, GetWindowTitle(window)});
+ sources->push_back(DesktopCapturer::Source{window_id, pid, GetWindowTitle(window)});
}
return true;
},
--
2.34.1

225
third_party/libwebrtc/moz-patch-stack/0057.patch поставляемый
Просмотреть файл

@ -1,29 +1,208 @@
From: Nico Grunbaum <na-g@nostrum.com>
Date: Thu, 28 Oct 2021 18:13:00 +0000
Subject: Bug 1729367 - P7 - restore mac PID tracking using new API;r=mjf
a=webrtc-update
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Tue, 2 Nov 2021 14:35:00 +0000
Subject: Bug 1729455 - Add to stats the local receive time for receiving video
Sender Reports. r=ng
Differential Revision: https://phabricator.services.mozilla.com/D129721
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/1495ca5ef535f8ad692a3a579ca42eddc14f39a8
Differential Revision: https://phabricator.services.mozilla.com/D125712
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/99267b6d193fbcb3e4c845c5e80770424d6d06e2
---
modules/desktop_capture/window_capturer_mac.mm | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
call/video_receive_stream.h | 3 ++-
modules/rtp_rtcp/source/rtcp_receiver.cc | 6 ++++--
modules/rtp_rtcp/source/rtcp_receiver.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 10 +++++-----
modules/rtp_rtcp/source/rtp_rtcp_impl.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc | 10 +++++-----
modules/rtp_rtcp/source/rtp_rtcp_impl2.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 5 +++--
video/rtp_video_stream_receiver2.cc | 5 +++--
video/rtp_video_stream_receiver2.h | 3 ++-
video/video_receive_stream2.cc | 3 ++-
11 files changed, 32 insertions(+), 22 deletions(-)
diff --git a/modules/desktop_capture/window_capturer_mac.mm b/modules/desktop_capture/window_capturer_mac.mm
index f0b413b0a6..882498bc34 100644
--- a/modules/desktop_capture/window_capturer_mac.mm
+++ b/modules/desktop_capture/window_capturer_mac.mm
@@ -167,8 +167,9 @@ void WindowCapturerMac::CaptureFrame() {
return webrtc::GetWindowList(
[sources](CFDictionaryRef window) {
WindowId window_id = GetWindowId(window);
+ int pid = GetWindowOwnerPid(window);
if (window_id != kNullWindowId) {
- sources->push_back(DesktopCapturer::Source{window_id, GetWindowTitle(window)});
+ sources->push_back(DesktopCapturer::Source{window_id, pid, GetWindowTitle(window)});
}
return true;
},
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 31c9bff7b0..25c294a2a6 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -145,10 +145,11 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
RtpReceiveStats rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
- // Mozilla modification: Init these three.
+ // Mozilla modification: Init these.
uint32_t rtcp_sender_packets_sent = 0;
uint32_t rtcp_sender_octets_sent = 0;
int64_t rtcp_sender_ntp_timestamp_ms = 0;
+ int64_t rtcp_sender_remote_ntp_timestamp_ms = 0;
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 69d62ead5a..936750c263 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -432,11 +432,13 @@ RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const {
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const {
MutexLock lock(&rtcp_receiver_lock_);
*packet_count = remote_sender_packet_count_;
*octet_count = remote_sender_octet_count_;
- *ntp_timestamp_ms = remote_sender_ntp_time_.ToMs();
+ *ntp_timestamp_ms = last_received_sr_ntp_.ToMs();
+ *remote_ntp_timestamp_ms = remote_sender_ntp_time_.ToMs();
}
std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index a05a69059a..e3f5bc765c 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -135,7 +135,8 @@ class RTCPReceiver final {
// Get received sender packet and octet counts
void RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const;
// Get rtt.
int32_t RTT(uint32_t remote_ssrc,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index bf9e2b3bf9..1c31611409 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -527,11 +527,11 @@ void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
}
// Received RTCP report.
-void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const {
- return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count,
- ntp_timestamp_ms);
+void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(
+ uint32_t* packet_count, uint32_t* octet_count, int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(
+ packet_count, octet_count, ntp_timestamp_ms, remote_ntp_timestamp_ms);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 5cf558717e..6070b67d44 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -194,7 +194,8 @@ class ABSL_DEPRECATED("") ModuleRtpRtcpImpl
void RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const override;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const override;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index 8378a76133..66d2e7a44e 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -508,11 +508,11 @@ void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
}
// Received RTCP report.
-void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const {
- return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count,
- ntp_timestamp_ms);
+void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(
+ uint32_t* packet_count, uint32_t* octet_count, int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(
+ packet_count, octet_count, ntp_timestamp_ms, remote_ntp_timestamp_ms);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl2::GetLatestReportBlockData()
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index 4ef67d4647..c43d0c34ba 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -206,7 +206,8 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
void RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const override;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const override;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index b988c7805d..cb4a0a427f 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -403,10 +403,11 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
StreamDataCounters* rtx_counters) const = 0;
- // Returns packet count, octet count, and timestamp from RTCP sender report.
+ // Returns packet count, octet count, and timestamps from RTCP sender report.
virtual void RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const = 0;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const = 0;
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the sender-source SSRC pair is unique and per-pair the
// ReportBlockData represents the latest Report Block that was received for
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 05447ac3bd..094f8f4a54 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -1067,9 +1067,10 @@ absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
// seem to be any support for these stats right now. So, we hack this in.
void RtpVideoStreamReceiver2::RemoteRTCPSenderInfo(
uint32_t* packet_count, uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const {
+ int64_t* ntp_timestamp_ms, int64_t* remote_ntp_timestamp_ms) const {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
- rtp_rtcp_->RemoteRTCPSenderInfo(packet_count, octet_count, ntp_timestamp_ms);
+ rtp_rtcp_->RemoteRTCPSenderInfo(packet_count, octet_count, ntp_timestamp_ms,
+ remote_ntp_timestamp_ms);
}
void RtpVideoStreamReceiver2::ManageFrame(
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 21f125ae2f..6bf4bf8453 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -215,7 +215,8 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
// stats at all, and even on the most recent libwebrtc code there does not
// seem to be any support for these stats right now. So, we hack this in.
void RemoteRTCPSenderInfo(uint32_t* packet_count, uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const;
private:
// Implements RtpVideoFrameReceiver.
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index be850834d6..7cbd49d322 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -603,7 +603,8 @@ VideoReceiveStreamInterface::Stats VideoReceiveStream2::GetStats() const {
// seem to be any support for these stats right now. So, we hack this in.
rtp_video_stream_receiver_.RemoteRTCPSenderInfo(
&stats.rtcp_sender_packets_sent, &stats.rtcp_sender_octets_sent,
- &stats.rtcp_sender_ntp_timestamp_ms);
+ &stats.rtcp_sender_ntp_timestamp_ms,
+ &stats.rtcp_sender_remote_ntp_timestamp_ms);
return stats;
}
--
2.34.1

214
third_party/libwebrtc/moz-patch-stack/0058.patch поставляемый
Просмотреть файл

@ -1,208 +1,26 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Tue, 2 Nov 2021 14:35:00 +0000
Subject: Bug 1729455 - Add to stats the local receive time for receiving video
Sender Reports. r=ng
Subject: Bug 1729455 - Ensure the libwebrtc system clock is not used. r=bwc
Differential Revision: https://phabricator.services.mozilla.com/D125712
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/99267b6d193fbcb3e4c845c5e80770424d6d06e2
Differential Revision: https://phabricator.services.mozilla.com/D128244
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/3e8ac168ee3db089dd892bf140df53e15d6f0918
---
call/video_receive_stream.h | 3 ++-
modules/rtp_rtcp/source/rtcp_receiver.cc | 6 ++++--
modules/rtp_rtcp/source/rtcp_receiver.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 10 +++++-----
modules/rtp_rtcp/source/rtp_rtcp_impl.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc | 10 +++++-----
modules/rtp_rtcp/source/rtp_rtcp_impl2.h | 3 ++-
modules/rtp_rtcp/source/rtp_rtcp_interface.h | 5 +++--
video/rtp_video_stream_receiver2.cc | 5 +++--
video/rtp_video_stream_receiver2.h | 3 ++-
video/video_receive_stream2.cc | 3 ++-
11 files changed, 32 insertions(+), 22 deletions(-)
rtc_base/system_time.cc | 2 ++
1 file changed, 2 insertions(+)
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 31c9bff7b0..25c294a2a6 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -145,10 +145,11 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
RtpReceiveStats rtp_stats;
RtcpPacketTypeCounter rtcp_packet_type_counts;
diff --git a/rtc_base/system_time.cc b/rtc_base/system_time.cc
index 058e6c2990..1a5e447916 100644
--- a/rtc_base/system_time.cc
+++ b/rtc_base/system_time.cc
@@ -12,6 +12,8 @@
// rtc::SystemTimeNanos() must be provided externally.
#ifndef WEBRTC_EXCLUDE_SYSTEM_TIME
- // Mozilla modification: Init these three.
+ // Mozilla modification: Init these.
uint32_t rtcp_sender_packets_sent = 0;
uint32_t rtcp_sender_octets_sent = 0;
int64_t rtcp_sender_ntp_timestamp_ms = 0;
+ int64_t rtcp_sender_remote_ntp_timestamp_ms = 0;
+#error Mozilla: Must not use the built-in libwebrtc clock
+
#include <stdint.h>
// Timing frame info: all important timestamps for a full lifetime of a
// single 'timing frame'.
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
index 69d62ead5a..936750c263 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -432,11 +432,13 @@ RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const {
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const {
MutexLock lock(&rtcp_receiver_lock_);
*packet_count = remote_sender_packet_count_;
*octet_count = remote_sender_octet_count_;
- *ntp_timestamp_ms = remote_sender_ntp_time_.ToMs();
+ *ntp_timestamp_ms = last_received_sr_ntp_.ToMs();
+ *remote_ntp_timestamp_ms = remote_sender_ntp_time_.ToMs();
}
std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
diff --git a/modules/rtp_rtcp/source/rtcp_receiver.h b/modules/rtp_rtcp/source/rtcp_receiver.h
index a05a69059a..e3f5bc765c 100644
--- a/modules/rtp_rtcp/source/rtcp_receiver.h
+++ b/modules/rtp_rtcp/source/rtcp_receiver.h
@@ -135,7 +135,8 @@ class RTCPReceiver final {
// Get received sender packet and octet counts
void RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const;
// Get rtt.
int32_t RTT(uint32_t remote_ssrc,
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index bf9e2b3bf9..1c31611409 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -527,11 +527,11 @@ void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
}
// Received RTCP report.
-void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const {
- return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count,
- ntp_timestamp_ms);
+void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo(
+ uint32_t* packet_count, uint32_t* octet_count, int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(
+ packet_count, octet_count, ntp_timestamp_ms, remote_ntp_timestamp_ms);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 5cf558717e..6070b67d44 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -194,7 +194,8 @@ class ABSL_DEPRECATED("") ModuleRtpRtcpImpl
void RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const override;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const override;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index 8378a76133..66d2e7a44e 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -508,11 +508,11 @@ void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
}
// Received RTCP report.
-void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(uint32_t* packet_count,
- uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const {
- return rtcp_receiver_.RemoteRTCPSenderInfo(packet_count, octet_count,
- ntp_timestamp_ms);
+void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(
+ uint32_t* packet_count, uint32_t* octet_count, int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const {
+ return rtcp_receiver_.RemoteRTCPSenderInfo(
+ packet_count, octet_count, ntp_timestamp_ms, remote_ntp_timestamp_ms);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl2::GetLatestReportBlockData()
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index 4ef67d4647..c43d0c34ba 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -206,7 +206,8 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface,
void RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const override;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const override;
// A snapshot of the most recent Report Block with additional data of
// interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_interface.h b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
index b988c7805d..cb4a0a427f 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_interface.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_interface.h
@@ -403,10 +403,11 @@ class RtpRtcpInterface : public RtcpFeedbackSenderInterface {
StreamDataCounters* rtx_counters) const = 0;
- // Returns packet count, octet count, and timestamp from RTCP sender report.
+ // Returns packet count, octet count, and timestamps from RTCP sender report.
virtual void RemoteRTCPSenderInfo(uint32_t* packet_count,
uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const = 0;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const = 0;
// A snapshot of Report Blocks with additional data of interest to statistics.
// Within this list, the sender-source SSRC pair is unique and per-pair the
// ReportBlockData represents the latest Report Block that was received for
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 05447ac3bd..094f8f4a54 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -1067,9 +1067,10 @@ absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
// seem to be any support for these stats right now. So, we hack this in.
void RtpVideoStreamReceiver2::RemoteRTCPSenderInfo(
uint32_t* packet_count, uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const {
+ int64_t* ntp_timestamp_ms, int64_t* remote_ntp_timestamp_ms) const {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
- rtp_rtcp_->RemoteRTCPSenderInfo(packet_count, octet_count, ntp_timestamp_ms);
+ rtp_rtcp_->RemoteRTCPSenderInfo(packet_count, octet_count, ntp_timestamp_ms,
+ remote_ntp_timestamp_ms);
}
void RtpVideoStreamReceiver2::ManageFrame(
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 21f125ae2f..6bf4bf8453 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -215,7 +215,8 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
// stats at all, and even on the most recent libwebrtc code there does not
// seem to be any support for these stats right now. So, we hack this in.
void RemoteRTCPSenderInfo(uint32_t* packet_count, uint32_t* octet_count,
- int64_t* ntp_timestamp_ms) const;
+ int64_t* ntp_timestamp_ms,
+ int64_t* remote_ntp_timestamp_ms) const;
private:
// Implements RtpVideoFrameReceiver.
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index be850834d6..7cbd49d322 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -603,7 +603,8 @@ VideoReceiveStreamInterface::Stats VideoReceiveStream2::GetStats() const {
// seem to be any support for these stats right now. So, we hack this in.
rtp_video_stream_receiver_.RemoteRTCPSenderInfo(
&stats.rtcp_sender_packets_sent, &stats.rtcp_sender_octets_sent,
- &stats.rtcp_sender_ntp_timestamp_ms);
+ &stats.rtcp_sender_ntp_timestamp_ms,
+ &stats.rtcp_sender_remote_ntp_timestamp_ms);
return stats;
}
#include <limits>
--
2.34.1

205
third_party/libwebrtc/moz-patch-stack/0059.patch поставляемый
Просмотреть файл

@ -1,26 +1,199 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Tue, 2 Nov 2021 14:35:00 +0000
Subject: Bug 1729455 - Ensure the libwebrtc system clock is not used. r=bwc
Subject: Bug 1729455 - Inject RTCStatsTimestampMakerRealtimeClock into Call
instances. r=bwc
Differential Revision: https://phabricator.services.mozilla.com/D128244
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/3e8ac168ee3db089dd892bf140df53e15d6f0918
This patch makes libwebrtc use our clock for timestamps.
It also makes sure there's no use of the libwebrtc realtime clock, other than
for relative time tracking (like timeouts), and that future libwebrtc updates
don't introduce unaudited use of it.
Differential Revision: https://phabricator.services.mozilla.com/D127714
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/0744d68b8c944e69945de4ac5c4ca71332e78ad8
---
rtc_base/system_time.cc | 2 ++
1 file changed, 2 insertions(+)
audio/channel_send.cc | 2 +-
call/call.cc | 2 ++
call/call_factory.cc | 4 ++++
call/degraded_call.cc | 2 ++
modules/audio_coding/acm2/audio_coding_module.cc | 2 +-
modules/rtp_rtcp/include/flexfec_receiver.h | 2 ++
modules/rtp_rtcp/source/flexfec_receiver.cc | 2 ++
rtc_base/task_utils/repeating_task.h | 4 ++--
system_wrappers/include/clock.h | 2 +-
system_wrappers/source/clock.cc | 2 +-
10 files changed, 18 insertions(+), 6 deletions(-)
diff --git a/rtc_base/system_time.cc b/rtc_base/system_time.cc
index 058e6c2990..1a5e447916 100644
--- a/rtc_base/system_time.cc
+++ b/rtc_base/system_time.cc
@@ -12,6 +12,8 @@
// rtc::SystemTimeNanos() must be provided externally.
#ifndef WEBRTC_EXCLUDE_SYSTEM_TIME
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index bf8e81fda4..5f5f24e825 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -502,7 +502,7 @@ ChannelSend::ChannelSend(
RtpRtcpInterface::Configuration configuration;
configuration.bandwidth_callback = rtcp_observer_.get();
configuration.transport_feedback_callback = feedback_observer_;
- configuration.clock = (clock ? clock : Clock::GetRealTimeClock());
+ configuration.clock = clock;
configuration.audio = true;
configuration.outgoing_transport = rtp_transport;
+#error Mozilla: Must not use the built-in libwebrtc clock
+
#include <stdint.h>
diff --git a/call/call.cc b/call/call.cc
index 218505cdea..c32705585f 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -494,12 +494,14 @@ std::string Call::Stats::ToString(int64_t time_ms) const {
return ss.str();
}
#include <limits>
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
Call* Call::Create(const Call::Config& config) {
Clock* clock = Clock::GetRealTimeClock();
return Create(config, clock,
RtpTransportControllerSendFactory().Create(
config.ExtractTransportConfig(), clock));
}
+ */
Call* Call::Create(const Call::Config& config,
Clock* clock,
diff --git a/call/call_factory.cc b/call/call_factory.cc
index 380e80ce12..253f8cd7de 100644
--- a/call/call_factory.cc
+++ b/call/call_factory.cc
@@ -95,6 +95,9 @@ Call* CallFactory::CreateCall(const Call::Config& config) {
RtpTransportConfig transportConfig = config.ExtractTransportConfig();
+ RTC_CHECK(false);
+ return nullptr;
+ /* Mozilla: Avoid this since it could use GetRealTimeClock().
Call* call =
Call::Create(config, Clock::GetRealTimeClock(),
config.rtp_transport_controller_send_factory->Create(
@@ -107,6 +110,7 @@ Call* CallFactory::CreateCall(const Call::Config& config) {
}
return call;
+ */
}
std::unique_ptr<CallFactoryInterface> CreateCallFactory() {
diff --git a/call/degraded_call.cc b/call/degraded_call.cc
index 50349c1086..00b121bbdc 100644
--- a/call/degraded_call.cc
+++ b/call/degraded_call.cc
@@ -129,6 +129,7 @@ bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtcp(
return true;
}
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
DegradedCall::DegradedCall(
std::unique_ptr<Call> call,
const std::vector<TimeScopedNetworkConfig>& send_configs,
@@ -165,6 +166,7 @@ DegradedCall::DegradedCall(
}
}
}
+*/
DegradedCall::~DegradedCall() {
RTC_DCHECK_RUN_ON(call_->worker_thread());
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 4367ab08fa..2c186273b6 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -620,7 +620,7 @@ int AudioCodingModuleImpl::GetTargetBitrate() const {
AudioCodingModule::Config::Config(
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
: neteq_config(),
- clock(Clock::GetRealTimeClock()),
+ clock(Clock::GetRealTimeClockRaw()),
decoder_factory(decoder_factory) {
// Post-decode VAD is disabled by default in NetEq, however, Audio
// Conference Mixer relies on VAD decisions and fails without them.
diff --git a/modules/rtp_rtcp/include/flexfec_receiver.h b/modules/rtp_rtcp/include/flexfec_receiver.h
index 3cf4c3845e..29d9e72786 100644
--- a/modules/rtp_rtcp/include/flexfec_receiver.h
+++ b/modules/rtp_rtcp/include/flexfec_receiver.h
@@ -29,9 +29,11 @@ class Clock;
class FlexfecReceiver {
public:
+ /* Mozilla: Avoid this since it could use GetRealTimeClock().
FlexfecReceiver(uint32_t ssrc,
uint32_t protected_media_ssrc,
RecoveredPacketReceiver* recovered_packet_receiver);
+ */
FlexfecReceiver(Clock* clock,
uint32_t ssrc,
uint32_t protected_media_ssrc,
diff --git a/modules/rtp_rtcp/source/flexfec_receiver.cc b/modules/rtp_rtcp/source/flexfec_receiver.cc
index 3f345cd6d2..bd67de1f83 100644
--- a/modules/rtp_rtcp/source/flexfec_receiver.cc
+++ b/modules/rtp_rtcp/source/flexfec_receiver.cc
@@ -29,6 +29,7 @@ constexpr int kPacketLogIntervalMs = 10000;
} // namespace
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
FlexfecReceiver::FlexfecReceiver(
uint32_t ssrc,
uint32_t protected_media_ssrc,
@@ -37,6 +38,7 @@ FlexfecReceiver::FlexfecReceiver(
ssrc,
protected_media_ssrc,
recovered_packet_receiver) {}
+ */
FlexfecReceiver::FlexfecReceiver(
Clock* clock,
diff --git a/rtc_base/task_utils/repeating_task.h b/rtc_base/task_utils/repeating_task.h
index e5ea3d8174..c06bac2247 100644
--- a/rtc_base/task_utils/repeating_task.h
+++ b/rtc_base/task_utils/repeating_task.h
@@ -56,7 +56,7 @@ class RepeatingTaskHandle {
absl::AnyInvocable<TimeDelta()> closure,
TaskQueueBase::DelayPrecision precision =
TaskQueueBase::DelayPrecision::kLow,
- Clock* clock = Clock::GetRealTimeClock());
+ Clock* clock = Clock::GetRealTimeClockRaw());
// DelayedStart is equivalent to Start except that the first invocation of the
// closure will be delayed by the given amount.
@@ -66,7 +66,7 @@ class RepeatingTaskHandle {
absl::AnyInvocable<TimeDelta()> closure,
TaskQueueBase::DelayPrecision precision =
TaskQueueBase::DelayPrecision::kLow,
- Clock* clock = Clock::GetRealTimeClock());
+ Clock* clock = Clock::GetRealTimeClockRaw());
// Stops future invocations of the repeating task closure. Can only be called
// from the TaskQueue where the task is running. The closure is guaranteed to
diff --git a/system_wrappers/include/clock.h b/system_wrappers/include/clock.h
index 60296070cc..214b34c970 100644
--- a/system_wrappers/include/clock.h
+++ b/system_wrappers/include/clock.h
@@ -49,7 +49,7 @@ class RTC_EXPORT Clock {
}
// Returns an instance of the real-time system clock implementation.
- static Clock* GetRealTimeClock();
+ static Clock* GetRealTimeClockRaw();
};
class SimulatedClock : public Clock {
diff --git a/system_wrappers/source/clock.cc b/system_wrappers/source/clock.cc
index 88c99d6a68..f7460b831c 100644
--- a/system_wrappers/source/clock.cc
+++ b/system_wrappers/source/clock.cc
@@ -57,7 +57,7 @@ class RealTimeClock : public Clock {
}
};
-Clock* Clock::GetRealTimeClock() {
+Clock* Clock::GetRealTimeClockRaw() {
static Clock* const clock = new RealTimeClock();
return clock;
}
--
2.34.1

266
third_party/libwebrtc/moz-patch-stack/0060.patch поставляемый
Просмотреть файл

@ -1,199 +1,91 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Tue, 2 Nov 2021 14:35:00 +0000
Subject: Bug 1729455 - Inject RTCStatsTimestampMakerRealtimeClock into Call
instances. r=bwc
From: Landry Breuil <landry@openbsd.org>
Date: Wed, 22 Dec 2021 00:09:00 +0000
Subject: Bug 1654448 - P2 - readd partial support for BSD to webrtc
build;r=mjf
This patch makes libwebrtc use our clock for timestamps.
It also makes sure there's no use of the libwebrtc realtime clock, other than
for relative time tracking (like timeouts), and that future libwebrtc updates
don't introduce unaudited use of it.
only OpenBSD/amd64 is supported for now
Differential Revision: https://phabricator.services.mozilla.com/D127714
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/0744d68b8c944e69945de4ac5c4ca71332e78ad8
Depends on D134432
Differential Revision: https://phabricator.services.mozilla.com/D134433
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/0300b32b7de70fb8976dc82d7d3bb3adb9685857
---
audio/channel_send.cc | 2 +-
call/call.cc | 2 ++
call/call_factory.cc | 4 ++++
call/degraded_call.cc | 2 ++
modules/audio_coding/acm2/audio_coding_module.cc | 2 +-
modules/rtp_rtcp/include/flexfec_receiver.h | 2 ++
modules/rtp_rtcp/source/flexfec_receiver.cc | 2 ++
rtc_base/task_utils/repeating_task.h | 4 ++--
system_wrappers/include/clock.h | 2 +-
system_wrappers/source/clock.cc | 2 +-
10 files changed, 18 insertions(+), 6 deletions(-)
BUILD.gn | 3 +++
modules/video_capture/BUILD.gn | 2 +-
modules/video_capture/linux/device_info_v4l2.h | 2 ++
rtc_base/platform_thread_types.cc | 4 +++-
webrtc.gni | 2 +-
5 files changed, 10 insertions(+), 3 deletions(-)
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index bf8e81fda4..5f5f24e825 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -502,7 +502,7 @@ ChannelSend::ChannelSend(
RtpRtcpInterface::Configuration configuration;
configuration.bandwidth_callback = rtcp_observer_.get();
configuration.transport_feedback_callback = feedback_observer_;
- configuration.clock = (clock ? clock : Clock::GetRealTimeClock());
+ configuration.clock = clock;
configuration.audio = true;
configuration.outgoing_transport = rtp_transport;
diff --git a/call/call.cc b/call/call.cc
index 218505cdea..c32705585f 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -494,12 +494,14 @@ std::string Call::Stats::ToString(int64_t time_ms) const {
return ss.str();
}
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
Call* Call::Create(const Call::Config& config) {
Clock* clock = Clock::GetRealTimeClock();
return Create(config, clock,
RtpTransportControllerSendFactory().Create(
config.ExtractTransportConfig(), clock));
}
+ */
Call* Call::Create(const Call::Config& config,
Clock* clock,
diff --git a/call/call_factory.cc b/call/call_factory.cc
index 380e80ce12..253f8cd7de 100644
--- a/call/call_factory.cc
+++ b/call/call_factory.cc
@@ -95,6 +95,9 @@ Call* CallFactory::CreateCall(const Call::Config& config) {
RtpTransportConfig transportConfig = config.ExtractTransportConfig();
+ RTC_CHECK(false);
+ return nullptr;
+ /* Mozilla: Avoid this since it could use GetRealTimeClock().
Call* call =
Call::Create(config, Clock::GetRealTimeClock(),
config.rtp_transport_controller_send_factory->Create(
@@ -107,6 +110,7 @@ Call* CallFactory::CreateCall(const Call::Config& config) {
diff --git a/BUILD.gn b/BUILD.gn
index 4e2a1f1e70..03880dbb09 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -215,6 +215,9 @@ config("common_inherited_config") {
if (is_linux || is_chromeos) {
defines += [ "WEBRTC_LINUX" ]
}
return call;
+ */
}
std::unique_ptr<CallFactoryInterface> CreateCallFactory() {
diff --git a/call/degraded_call.cc b/call/degraded_call.cc
index 50349c1086..00b121bbdc 100644
--- a/call/degraded_call.cc
+++ b/call/degraded_call.cc
@@ -129,6 +129,7 @@ bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtcp(
return true;
}
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
DegradedCall::DegradedCall(
std::unique_ptr<Call> call,
const std::vector<TimeScopedNetworkConfig>& send_configs,
@@ -165,6 +166,7 @@ DegradedCall::DegradedCall(
}
+ if (is_bsd) {
+ defines += [ "WEBRTC_BSD" ]
+ }
if (is_mac) {
defines += [ "WEBRTC_MAC" ]
}
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index 95548906c4..4a5bf62433 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -63,7 +63,7 @@ if (!build_with_chromium) {
"../../system_wrappers",
]
- if (is_linux || is_chromeos) {
+ if (is_linux || is_bsd || is_chromeos) {
sources = [
"linux/device_info_linux.cc",
"linux/device_info_v4l2.cc",
diff --git a/modules/video_capture/linux/device_info_v4l2.h b/modules/video_capture/linux/device_info_v4l2.h
index e3c2395f49..119cb07ab8 100644
--- a/modules/video_capture/linux/device_info_v4l2.h
+++ b/modules/video_capture/linux/device_info_v4l2.h
@@ -16,7 +16,9 @@
#include "modules/video_capture/device_info_impl.h"
#include "rtc_base/platform_thread.h"
+#ifdef WEBRTC_LINUX
#include <sys/inotify.h>
+#endif
struct v4l2_capability;
diff --git a/rtc_base/platform_thread_types.cc b/rtc_base/platform_thread_types.cc
index d64ea689bb..c3c6955a7b 100644
--- a/rtc_base/platform_thread_types.cc
+++ b/rtc_base/platform_thread_types.cc
@@ -50,7 +50,9 @@ PlatformThreadId CurrentThreadId() {
return static_cast<PlatformThreadId>(pthread_self());
#else
// Default implementation for nacl and solaris.
- return reinterpret_cast<PlatformThreadId>(pthread_self());
+ // WEBRTC_BSD: pthread_t is a pointer, so cannot be casted to pid_t
+ // (aka int32_t) on 64-bit archs. Required on OpenBSD.
+ return reinterpret_cast<long>(pthread_self());
#endif
#endif // defined(WEBRTC_POSIX)
}
+*/
diff --git a/webrtc.gni b/webrtc.gni
index e23c9a1cc4..1b21d329b2 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -336,7 +336,7 @@ rtc_opus_dir = "//third_party/opus"
DegradedCall::~DegradedCall() {
RTC_DCHECK_RUN_ON(call_->worker_thread());
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 4367ab08fa..2c186273b6 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -620,7 +620,7 @@ int AudioCodingModuleImpl::GetTargetBitrate() const {
AudioCodingModule::Config::Config(
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
: neteq_config(),
- clock(Clock::GetRealTimeClock()),
+ clock(Clock::GetRealTimeClockRaw()),
decoder_factory(decoder_factory) {
// Post-decode VAD is disabled by default in NetEq, however, Audio
// Conference Mixer relies on VAD decisions and fails without them.
diff --git a/modules/rtp_rtcp/include/flexfec_receiver.h b/modules/rtp_rtcp/include/flexfec_receiver.h
index 3cf4c3845e..29d9e72786 100644
--- a/modules/rtp_rtcp/include/flexfec_receiver.h
+++ b/modules/rtp_rtcp/include/flexfec_receiver.h
@@ -29,9 +29,11 @@ class Clock;
# Desktop capturer is supported only on Windows, OSX and Linux.
rtc_desktop_capture_supported =
- (is_win && current_os != "winuwp") || is_mac ||
+ (is_win && current_os != "winuwp") || is_mac || is_bsd ||
((is_linux || is_chromeos) && (rtc_use_x11_extensions || rtc_use_pipewire))
class FlexfecReceiver {
public:
+ /* Mozilla: Avoid this since it could use GetRealTimeClock().
FlexfecReceiver(uint32_t ssrc,
uint32_t protected_media_ssrc,
RecoveredPacketReceiver* recovered_packet_receiver);
+ */
FlexfecReceiver(Clock* clock,
uint32_t ssrc,
uint32_t protected_media_ssrc,
diff --git a/modules/rtp_rtcp/source/flexfec_receiver.cc b/modules/rtp_rtcp/source/flexfec_receiver.cc
index 3f345cd6d2..bd67de1f83 100644
--- a/modules/rtp_rtcp/source/flexfec_receiver.cc
+++ b/modules/rtp_rtcp/source/flexfec_receiver.cc
@@ -29,6 +29,7 @@ constexpr int kPacketLogIntervalMs = 10000;
} // namespace
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
FlexfecReceiver::FlexfecReceiver(
uint32_t ssrc,
uint32_t protected_media_ssrc,
@@ -37,6 +38,7 @@ FlexfecReceiver::FlexfecReceiver(
ssrc,
protected_media_ssrc,
recovered_packet_receiver) {}
+ */
FlexfecReceiver::FlexfecReceiver(
Clock* clock,
diff --git a/rtc_base/task_utils/repeating_task.h b/rtc_base/task_utils/repeating_task.h
index e5ea3d8174..c06bac2247 100644
--- a/rtc_base/task_utils/repeating_task.h
+++ b/rtc_base/task_utils/repeating_task.h
@@ -56,7 +56,7 @@ class RepeatingTaskHandle {
absl::AnyInvocable<TimeDelta()> closure,
TaskQueueBase::DelayPrecision precision =
TaskQueueBase::DelayPrecision::kLow,
- Clock* clock = Clock::GetRealTimeClock());
+ Clock* clock = Clock::GetRealTimeClockRaw());
// DelayedStart is equivalent to Start except that the first invocation of the
// closure will be delayed by the given amount.
@@ -66,7 +66,7 @@ class RepeatingTaskHandle {
absl::AnyInvocable<TimeDelta()> closure,
TaskQueueBase::DelayPrecision precision =
TaskQueueBase::DelayPrecision::kLow,
- Clock* clock = Clock::GetRealTimeClock());
+ Clock* clock = Clock::GetRealTimeClockRaw());
// Stops future invocations of the repeating task closure. Can only be called
// from the TaskQueue where the task is running. The closure is guaranteed to
diff --git a/system_wrappers/include/clock.h b/system_wrappers/include/clock.h
index 60296070cc..214b34c970 100644
--- a/system_wrappers/include/clock.h
+++ b/system_wrappers/include/clock.h
@@ -49,7 +49,7 @@ class RTC_EXPORT Clock {
}
// Returns an instance of the real-time system clock implementation.
- static Clock* GetRealTimeClock();
+ static Clock* GetRealTimeClockRaw();
};
class SimulatedClock : public Clock {
diff --git a/system_wrappers/source/clock.cc b/system_wrappers/source/clock.cc
index 88c99d6a68..f7460b831c 100644
--- a/system_wrappers/source/clock.cc
+++ b/system_wrappers/source/clock.cc
@@ -57,7 +57,7 @@ class RealTimeClock : public Clock {
}
};
-Clock* Clock::GetRealTimeClock() {
+Clock* Clock::GetRealTimeClockRaw() {
static Clock* const clock = new RealTimeClock();
return clock;
}
###############################################################################
--
2.34.1

228
third_party/libwebrtc/moz-patch-stack/0061.patch поставляемый
Просмотреть файл

@ -1,91 +1,163 @@
From: Landry Breuil <landry@openbsd.org>
Date: Wed, 22 Dec 2021 00:09:00 +0000
Subject: Bug 1654448 - P2 - readd partial support for BSD to webrtc
build;r=mjf
From: Michael Froman <mjfroman@mac.com>
Date: Mon, 4 Apr 2022 12:25:26 -0500
Subject: Bug 1766646 - (fix) breakout Call::Stats and SharedModuleThread into
seperate files
only OpenBSD/amd64 is supported for now
Depends on D134432
Differential Revision: https://phabricator.services.mozilla.com/D134433
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/0300b32b7de70fb8976dc82d7d3bb3adb9685857
---
BUILD.gn | 3 +++
modules/video_capture/BUILD.gn | 2 +-
modules/video_capture/linux/device_info_v4l2.h | 2 ++
rtc_base/platform_thread_types.cc | 4 +++-
webrtc.gni | 2 +-
5 files changed, 10 insertions(+), 3 deletions(-)
call/BUILD.gn | 6 ++++++
call/call.cc | 13 -------------
call/call.h | 13 ++-----------
call/call_basic_stats.cc | 20 ++++++++++++++++++++
call/call_basic_stats.h | 21 +++++++++++++++++++++
video/video_send_stream.h | 1 -
6 files changed, 49 insertions(+), 25 deletions(-)
create mode 100644 call/call_basic_stats.cc
create mode 100644 call/call_basic_stats.h
diff --git a/BUILD.gn b/BUILD.gn
index 4e2a1f1e70..03880dbb09 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -215,6 +215,9 @@ config("common_inherited_config") {
if (is_linux || is_chromeos) {
defines += [ "WEBRTC_LINUX" ]
}
+ if (is_bsd) {
+ defines += [ "WEBRTC_BSD" ]
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 0e52e8fb3f..26618aee80 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -33,6 +33,12 @@ rtc_library("call_interfaces") {
"syncable.cc",
"syncable.h",
]
+ if (build_with_mozilla) {
+ sources += [
+ "call_basic_stats.cc",
+ "call_basic_stats.h",
+ ]
+ }
if (is_mac) {
defines += [ "WEBRTC_MAC" ]
}
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index 95548906c4..4a5bf62433 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -63,7 +63,7 @@ if (!build_with_chromium) {
"../../system_wrappers",
]
- if (is_linux || is_chromeos) {
+ if (is_linux || is_bsd || is_chromeos) {
sources = [
"linux/device_info_linux.cc",
"linux/device_info_v4l2.cc",
diff --git a/modules/video_capture/linux/device_info_v4l2.h b/modules/video_capture/linux/device_info_v4l2.h
index e3c2395f49..119cb07ab8 100644
--- a/modules/video_capture/linux/device_info_v4l2.h
+++ b/modules/video_capture/linux/device_info_v4l2.h
@@ -16,7 +16,9 @@
#include "modules/video_capture/device_info_impl.h"
deps = [
":audio_sender_interface",
diff --git a/call/call.cc b/call/call.cc
index c32705585f..1fd49cede9 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -481,19 +481,6 @@ class Call final : public webrtc::Call,
};
} // namespace internal
#include "rtc_base/platform_thread.h"
+#ifdef WEBRTC_LINUX
#include <sys/inotify.h>
+#endif
-std::string Call::Stats::ToString(int64_t time_ms) const {
- char buf[1024];
- rtc::SimpleStringBuilder ss(buf);
- ss << "Call stats: " << time_ms << ", {";
- ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
- ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
- ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
- ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
- ss << "rtt_ms: " << rtt_ms;
- ss << '}';
- return ss.str();
-}
-
/* Mozilla: Avoid this since it could use GetRealTimeClock().
Call* Call::Create(const Call::Config& config) {
Clock* clock = Clock::GetRealTimeClock();
diff --git a/call/call.h b/call/call.h
index 366978392e..42daa95a6c 100644
--- a/call/call.h
+++ b/call/call.h
@@ -21,6 +21,7 @@
#include "api/task_queue/task_queue_base.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
+#include "call/call_basic_stats.h"
#include "call/call_config.h"
#include "call/flexfec_receive_stream.h"
#include "call/packet_receiver.h"
@@ -30,7 +31,6 @@
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
-#include "rtc_base/ref_count.h"
struct v4l2_capability;
namespace webrtc {
diff --git a/rtc_base/platform_thread_types.cc b/rtc_base/platform_thread_types.cc
index d64ea689bb..c3c6955a7b 100644
--- a/rtc_base/platform_thread_types.cc
+++ b/rtc_base/platform_thread_types.cc
@@ -50,7 +50,9 @@ PlatformThreadId CurrentThreadId() {
return static_cast<PlatformThreadId>(pthread_self());
#else
// Default implementation for nacl and solaris.
- return reinterpret_cast<PlatformThreadId>(pthread_self());
+ // WEBRTC_BSD: pthread_t is a pointer, so cannot be casted to pid_t
+ // (aka int32_t) on 64-bit archs. Required on OpenBSD.
+ return reinterpret_cast<long>(pthread_self());
#endif
#endif // defined(WEBRTC_POSIX)
}
diff --git a/webrtc.gni b/webrtc.gni
index e23c9a1cc4..1b21d329b2 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -336,7 +336,7 @@ rtc_opus_dir = "//third_party/opus"
@@ -47,16 +47,7 @@ namespace webrtc {
class Call {
public:
using Config = CallConfig;
-
- struct Stats {
- std::string ToString(int64_t time_ms) const;
-
- int send_bandwidth_bps = 0; // Estimated available send bandwidth.
- int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
- int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
- int64_t pacer_delay_ms = 0;
- int64_t rtt_ms = -1;
- };
+ using Stats = CallBasicStats;
# Desktop capturer is supported only on Windows, OSX and Linux.
rtc_desktop_capture_supported =
- (is_win && current_os != "winuwp") || is_mac ||
+ (is_win && current_os != "winuwp") || is_mac || is_bsd ||
((is_linux || is_chromeos) && (rtc_use_x11_extensions || rtc_use_pipewire))
static Call* Create(const Call::Config& config);
static Call* Create(const Call::Config& config,
diff --git a/call/call_basic_stats.cc b/call/call_basic_stats.cc
new file mode 100644
index 0000000000..74333a663b
--- /dev/null
+++ b/call/call_basic_stats.cc
@@ -0,0 +1,20 @@
+#include "call/call_basic_stats.h"
+
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+std::string CallBasicStats::ToString(int64_t time_ms) const {
+ char buf[1024];
+ rtc::SimpleStringBuilder ss(buf);
+ ss << "Call stats: " << time_ms << ", {";
+ ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
+ ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
+ ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
+ ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
+ ss << "rtt_ms: " << rtt_ms;
+ ss << '}';
+ return ss.str();
+}
+
+} // namespace webrtc
diff --git a/call/call_basic_stats.h b/call/call_basic_stats.h
new file mode 100644
index 0000000000..98febe9405
--- /dev/null
+++ b/call/call_basic_stats.h
@@ -0,0 +1,21 @@
+#ifndef CALL_CALL_BASIC_STATS_H_
+#define CALL_CALL_BASIC_STATS_H_
+
+#include <string>
+
+namespace webrtc {
+
+// named to avoid conflicts with video/call_stats.h
+struct CallBasicStats {
+ std::string ToString(int64_t time_ms) const;
+
+ int send_bandwidth_bps = 0; // Estimated available send bandwidth.
+ int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
+ int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
+ int64_t pacer_delay_ms = 0;
+ int64_t rtt_ms = -1;
+};
+
+} // namespace webrtc
+
+#endif // CALL_CALL_BASIC_STATS_H_
diff --git a/video/video_send_stream.h b/video/video_send_stream.h
index a7ce112b21..404873fd39 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -37,7 +37,6 @@ namespace test {
class VideoSendStreamPeer;
} // namespace test
###############################################################################
-class CallStats;
class IvfFileWriter;
class RateLimiter;
class RtpRtcp;
--
2.34.1

181
third_party/libwebrtc/moz-patch-stack/0062.patch поставляемый
Просмотреть файл

@ -1,163 +1,32 @@
From: Michael Froman <mjfroman@mac.com>
Date: Mon, 4 Apr 2022 12:25:26 -0500
Subject: Bug 1766646 - (fix) breakout Call::Stats and SharedModuleThread into
seperate files
Date: Fri, 8 Apr 2022 11:36:36 -0500
Subject: Bug 1766646 - (fix-b556b08668) avoid InlinedVector method that can
throw exception
---
call/BUILD.gn | 6 ++++++
call/call.cc | 13 -------------
call/call.h | 13 ++-----------
call/call_basic_stats.cc | 20 ++++++++++++++++++++
call/call_basic_stats.h | 21 +++++++++++++++++++++
video/video_send_stream.h | 1 -
6 files changed, 49 insertions(+), 25 deletions(-)
create mode 100644 call/call_basic_stats.cc
create mode 100644 call/call_basic_stats.h
api/video_codecs/video_encoder.cc | 8 ++++++++
1 file changed, 8 insertions(+)
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 0e52e8fb3f..26618aee80 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -33,6 +33,12 @@ rtc_library("call_interfaces") {
"syncable.cc",
"syncable.h",
]
+ if (build_with_mozilla) {
+ sources += [
+ "call_basic_stats.cc",
+ "call_basic_stats.h",
+ ]
+ }
deps = [
":audio_sender_interface",
diff --git a/call/call.cc b/call/call.cc
index c32705585f..1fd49cede9 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -481,19 +481,6 @@ class Call final : public webrtc::Call,
};
} // namespace internal
-std::string Call::Stats::ToString(int64_t time_ms) const {
- char buf[1024];
- rtc::SimpleStringBuilder ss(buf);
- ss << "Call stats: " << time_ms << ", {";
- ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
- ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
- ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
- ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
- ss << "rtt_ms: " << rtt_ms;
- ss << '}';
- return ss.str();
-}
-
/* Mozilla: Avoid this since it could use GetRealTimeClock().
Call* Call::Create(const Call::Config& config) {
Clock* clock = Clock::GetRealTimeClock();
diff --git a/call/call.h b/call/call.h
index 366978392e..42daa95a6c 100644
--- a/call/call.h
+++ b/call/call.h
@@ -21,6 +21,7 @@
#include "api/task_queue/task_queue_base.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
+#include "call/call_basic_stats.h"
#include "call/call_config.h"
#include "call/flexfec_receive_stream.h"
#include "call/packet_receiver.h"
@@ -30,7 +31,6 @@
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "rtc_base/network_route.h"
-#include "rtc_base/ref_count.h"
namespace webrtc {
@@ -47,16 +47,7 @@ namespace webrtc {
class Call {
public:
using Config = CallConfig;
-
- struct Stats {
- std::string ToString(int64_t time_ms) const;
-
- int send_bandwidth_bps = 0; // Estimated available send bandwidth.
- int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
- int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
- int64_t pacer_delay_ms = 0;
- int64_t rtt_ms = -1;
- };
+ using Stats = CallBasicStats;
static Call* Create(const Call::Config& config);
static Call* Create(const Call::Config& config,
diff --git a/call/call_basic_stats.cc b/call/call_basic_stats.cc
new file mode 100644
index 0000000000..74333a663b
--- /dev/null
+++ b/call/call_basic_stats.cc
@@ -0,0 +1,20 @@
+#include "call/call_basic_stats.h"
+
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+std::string CallBasicStats::ToString(int64_t time_ms) const {
+ char buf[1024];
+ rtc::SimpleStringBuilder ss(buf);
+ ss << "Call stats: " << time_ms << ", {";
+ ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
+ ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
+ ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
+ ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
+ ss << "rtt_ms: " << rtt_ms;
+ ss << '}';
+ return ss.str();
+}
+
+} // namespace webrtc
diff --git a/call/call_basic_stats.h b/call/call_basic_stats.h
new file mode 100644
index 0000000000..98febe9405
--- /dev/null
+++ b/call/call_basic_stats.h
@@ -0,0 +1,21 @@
+#ifndef CALL_CALL_BASIC_STATS_H_
+#define CALL_CALL_BASIC_STATS_H_
+
+#include <string>
+
+namespace webrtc {
+
+// named to avoid conflicts with video/call_stats.h
+struct CallBasicStats {
+ std::string ToString(int64_t time_ms) const;
+
+ int send_bandwidth_bps = 0; // Estimated available send bandwidth.
+ int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
+ int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
+ int64_t pacer_delay_ms = 0;
+ int64_t rtt_ms = -1;
+};
+
+} // namespace webrtc
+
+#endif // CALL_CALL_BASIC_STATS_H_
diff --git a/video/video_send_stream.h b/video/video_send_stream.h
index a7ce112b21..404873fd39 100644
--- a/video/video_send_stream.h
+++ b/video/video_send_stream.h
@@ -37,7 +37,6 @@ namespace test {
class VideoSendStreamPeer;
} // namespace test
-class CallStats;
class IvfFileWriter;
class RateLimiter;
class RtpRtcp;
diff --git a/api/video_codecs/video_encoder.cc b/api/video_codecs/video_encoder.cc
index b85b9328cf..deb4fdc637 100644
--- a/api/video_codecs/video_encoder.cc
+++ b/api/video_codecs/video_encoder.cc
@@ -179,7 +179,15 @@ std::string VideoEncoder::EncoderInfo::ToString() const {
for (size_t i = 0; i < preferred_pixel_formats.size(); ++i) {
if (i > 0)
oss << ", ";
+#if defined(WEBRTC_MOZILLA_BUILD)
+ // This could assert, as opposed to throw using the form in the
+ // else, but since we're in a for loop that uses .size() we can
+ // be fairly sure that this is safe without doing a further
+ // check to make sure 'i' is in-range.
+ oss << VideoFrameBufferTypeToString(preferred_pixel_formats[i]);
+#else
oss << VideoFrameBufferTypeToString(preferred_pixel_formats.at(i));
+#endif
}
oss << "]";
if (is_qp_trusted.has_value()) {
--
2.34.1

97
third_party/libwebrtc/moz-patch-stack/0063.patch поставляемый
Просмотреть файл

@ -1,32 +1,81 @@
From: Michael Froman <mjfroman@mac.com>
Date: Fri, 8 Apr 2022 11:36:36 -0500
Subject: Bug 1766646 - (fix-b556b08668) avoid InlinedVector method that can
throw exception
Date: Thu, 28 Apr 2022 10:53:43 -0500
Subject: Bug 1766646 - (fix-a0bb2ef2dc) add back VideoType enum values
---
api/video_codecs/video_encoder.cc | 8 ++++++++
1 file changed, 8 insertions(+)
common_video/libyuv/include/webrtc_libyuv.h | 4 ++++
common_video/libyuv/webrtc_libyuv.cc | 11 +++++++++++
2 files changed, 15 insertions(+)
diff --git a/api/video_codecs/video_encoder.cc b/api/video_codecs/video_encoder.cc
index b85b9328cf..deb4fdc637 100644
--- a/api/video_codecs/video_encoder.cc
+++ b/api/video_codecs/video_encoder.cc
@@ -179,7 +179,15 @@ std::string VideoEncoder::EncoderInfo::ToString() const {
for (size_t i = 0; i < preferred_pixel_formats.size(); ++i) {
if (i > 0)
oss << ", ";
+#if defined(WEBRTC_MOZILLA_BUILD)
+ // This could assert, as opposed to throw using the form in the
+ // else, but since we're in a for loop that uses .size() we can
+ // be fairly sure that this is safe without doing a further
+ // check to make sure 'i' is in-range.
+ oss << VideoFrameBufferTypeToString(preferred_pixel_formats[i]);
+#else
oss << VideoFrameBufferTypeToString(preferred_pixel_formats.at(i));
+#endif
diff --git a/common_video/libyuv/include/webrtc_libyuv.h b/common_video/libyuv/include/webrtc_libyuv.h
index 08a035a8d7..6d9071bcd5 100644
--- a/common_video/libyuv/include/webrtc_libyuv.h
+++ b/common_video/libyuv/include/webrtc_libyuv.h
@@ -32,12 +32,16 @@ enum class VideoType {
kI420,
kIYUV,
kRGB24,
+ kABGR,
kARGB,
+ kARGB4444,
kRGB565,
+ kARGB1555,
kYUY2,
kYV12,
kUYVY,
kMJPEG,
+ kNV21,
kBGRA,
kNV12,
};
diff --git a/common_video/libyuv/webrtc_libyuv.cc b/common_video/libyuv/webrtc_libyuv.cc
index 14e2d22612..8998af191d 100644
--- a/common_video/libyuv/webrtc_libyuv.cc
+++ b/common_video/libyuv/webrtc_libyuv.cc
@@ -25,6 +25,7 @@ size_t CalcBufferSize(VideoType type, int width, int height) {
size_t buffer_size = 0;
switch (type) {
case VideoType::kI420:
+ case VideoType::kNV21:
case VideoType::kIYUV:
case VideoType::kYV12:
case VideoType::kNV12: {
@@ -33,7 +34,9 @@ size_t CalcBufferSize(VideoType type, int width, int height) {
buffer_size = width * height + half_width * half_height * 2;
break;
}
+ case VideoType::kARGB4444:
case VideoType::kRGB565:
+ case VideoType::kARGB1555:
case VideoType::kYUY2:
case VideoType::kUYVY:
buffer_size = width * height * 2;
@@ -94,6 +97,8 @@ int ConvertVideoType(VideoType video_type) {
return libyuv::FOURCC_YV12;
case VideoType::kRGB24:
return libyuv::FOURCC_24BG;
+ case VideoType::kABGR:
+ return libyuv::FOURCC_ABGR;
case VideoType::kRGB565:
return libyuv::FOURCC_RGBP;
case VideoType::kYUY2:
@@ -102,10 +107,16 @@ int ConvertVideoType(VideoType video_type) {
return libyuv::FOURCC_UYVY;
case VideoType::kMJPEG:
return libyuv::FOURCC_MJPG;
+ case VideoType::kNV21:
+ return libyuv::FOURCC_NV21;
case VideoType::kARGB:
return libyuv::FOURCC_ARGB;
case VideoType::kBGRA:
return libyuv::FOURCC_BGRA;
+ case VideoType::kARGB4444:
+ return libyuv::FOURCC_R444;
+ case VideoType::kARGB1555:
+ return libyuv::FOURCC_RGBO;
case VideoType::kNV12:
return libyuv::FOURCC_NV12;
}
oss << "]";
if (is_qp_trusted.has_value()) {
--
2.34.1

141
third_party/libwebrtc/moz-patch-stack/0064.patch поставляемый
Просмотреть файл

@ -1,81 +1,76 @@
From: Michael Froman <mjfroman@mac.com>
Date: Thu, 28 Apr 2022 10:53:43 -0500
Subject: Bug 1766646 - (fix-a0bb2ef2dc) add back VideoType enum values
Date: Thu, 19 May 2022 15:32:32 -0500
Subject: Bug 1772380 - Build 1766646 - (fix-c89fdd716c) fixes for the
refactored PlatformThread API;r?mjf
---
common_video/libyuv/include/webrtc_libyuv.h | 4 ++++
common_video/libyuv/webrtc_libyuv.cc | 11 +++++++++++
2 files changed, 15 insertions(+)
.../video_capture/linux/device_info_v4l2.cc | 20 ++++++-------------
.../video_capture/linux/device_info_v4l2.h | 3 +--
2 files changed, 7 insertions(+), 16 deletions(-)
diff --git a/common_video/libyuv/include/webrtc_libyuv.h b/common_video/libyuv/include/webrtc_libyuv.h
index 08a035a8d7..6d9071bcd5 100644
--- a/common_video/libyuv/include/webrtc_libyuv.h
+++ b/common_video/libyuv/include/webrtc_libyuv.h
@@ -32,12 +32,16 @@ enum class VideoType {
kI420,
kIYUV,
kRGB24,
+ kABGR,
kARGB,
+ kARGB4444,
kRGB565,
+ kARGB1555,
kYUY2,
kYV12,
kUYVY,
kMJPEG,
+ kNV21,
kBGRA,
kNV12,
};
diff --git a/common_video/libyuv/webrtc_libyuv.cc b/common_video/libyuv/webrtc_libyuv.cc
index 14e2d22612..8998af191d 100644
--- a/common_video/libyuv/webrtc_libyuv.cc
+++ b/common_video/libyuv/webrtc_libyuv.cc
@@ -25,6 +25,7 @@ size_t CalcBufferSize(VideoType type, int width, int height) {
size_t buffer_size = 0;
switch (type) {
case VideoType::kI420:
+ case VideoType::kNV21:
case VideoType::kIYUV:
case VideoType::kYV12:
case VideoType::kNV12: {
@@ -33,7 +34,9 @@ size_t CalcBufferSize(VideoType type, int width, int height) {
buffer_size = width * height + half_width * half_height * 2;
break;
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index e8abcdda78..7651dd6651 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -133,11 +133,6 @@ int DeviceInfoV4l2::ProcessInotifyEvents()
return 0;
}
-void DeviceInfoV4l2::InotifyEventThread(void* obj)
-{
- static_cast<DeviceInfoLinux*> (obj)->InotifyProcess();
-}
-
void DeviceInfoV4l2::InotifyProcess()
{
_fd_v4l = inotify_init();
@@ -163,16 +158,14 @@ void DeviceInfoV4l2::InotifyProcess()
DeviceInfoV4l2::DeviceInfoV4l2() : DeviceInfoImpl()
#ifdef WEBRTC_LINUX
- , _inotifyEventThread(new rtc::PlatformThread(
- InotifyEventThread, this, "InotifyEventThread"))
, _isShutdown(false)
#endif
{
#ifdef WEBRTC_LINUX
- if (_inotifyEventThread)
- {
- _inotifyEventThread->Start();
- }
+ _inotifyEventThread = rtc::PlatformThread::SpawnJoinable(
+ [this] {
+ InotifyProcess();
+ }, "InotifyEventThread");
#endif
}
@@ -184,9 +177,8 @@ DeviceInfoV4l2::~DeviceInfoV4l2() {
#ifdef WEBRTC_LINUX
_isShutdown = true;
- if (_inotifyEventThread) {
- _inotifyEventThread->Stop();
- _inotifyEventThread = nullptr;
+ if (!_inotifyEventThread.empty()) {
+ _inotifyEventThread.Finalize();
}
+ case VideoType::kARGB4444:
case VideoType::kRGB565:
+ case VideoType::kARGB1555:
case VideoType::kYUY2:
case VideoType::kUYVY:
buffer_size = width * height * 2;
@@ -94,6 +97,8 @@ int ConvertVideoType(VideoType video_type) {
return libyuv::FOURCC_YV12;
case VideoType::kRGB24:
return libyuv::FOURCC_24BG;
+ case VideoType::kABGR:
+ return libyuv::FOURCC_ABGR;
case VideoType::kRGB565:
return libyuv::FOURCC_RGBP;
case VideoType::kYUY2:
@@ -102,10 +107,16 @@ int ConvertVideoType(VideoType video_type) {
return libyuv::FOURCC_UYVY;
case VideoType::kMJPEG:
return libyuv::FOURCC_MJPG;
+ case VideoType::kNV21:
+ return libyuv::FOURCC_NV21;
case VideoType::kARGB:
return libyuv::FOURCC_ARGB;
case VideoType::kBGRA:
return libyuv::FOURCC_BGRA;
+ case VideoType::kARGB4444:
+ return libyuv::FOURCC_R444;
+ case VideoType::kARGB1555:
+ return libyuv::FOURCC_RGBO;
case VideoType::kNV12:
return libyuv::FOURCC_NV12;
}
#endif
}
diff --git a/modules/video_capture/linux/device_info_v4l2.h b/modules/video_capture/linux/device_info_v4l2.h
index 119cb07ab8..0bec3eb765 100644
--- a/modules/video_capture/linux/device_info_v4l2.h
+++ b/modules/video_capture/linux/device_info_v4l2.h
@@ -60,8 +60,7 @@ class DeviceInfoV4l2 : public DeviceInfoImpl {
int EventCheck(int fd);
int HandleEvents(int fd);
int ProcessInotifyEvents();
- std::unique_ptr<rtc::PlatformThread> _inotifyEventThread;
- static void InotifyEventThread(void*);
+ rtc::PlatformThread _inotifyEventThread;
void InotifyProcess();
int _fd_v4l, _fd_dev, _wd_v4l, _wd_dev; /* accessed on InotifyEventThread thread */
std::atomic<bool> _isShutdown;
--
2.34.1

222
third_party/libwebrtc/moz-patch-stack/0065.patch поставляемый
Просмотреть файл

@ -1,76 +1,174 @@
From: Michael Froman <mjfroman@mac.com>
Date: Thu, 19 May 2022 15:32:32 -0500
Subject: Bug 1772380 - Build 1766646 - (fix-c89fdd716c) fixes for the
refactored PlatformThread API;r?mjf
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 18 Jan 2021 11:07:00 +0100
Subject: Bug 1766646 - (fix-ae0d117d51) ifdef our Csrc impl vs upstream's
impl, see Bug 1771332.
---
.../video_capture/linux/device_info_v4l2.cc | 20 ++++++-------------
.../video_capture/linux/device_info_v4l2.h | 3 +--
2 files changed, 7 insertions(+), 16 deletions(-)
modules/rtp_rtcp/source/rtp_header_extensions.cc | 4 ++++
modules/rtp_rtcp/source/rtp_header_extensions.h | 4 ++++
modules/rtp_rtcp/source/rtp_packet.cc | 4 ++++
modules/rtp_rtcp/source/rtp_sender.cc | 4 ++++
test/fuzzers/rtp_packet_fuzzer.cc | 4 ++++
5 files changed, 20 insertions(+)
diff --git a/modules/video_capture/linux/device_info_v4l2.cc b/modules/video_capture/linux/device_info_v4l2.cc
index e8abcdda78..7651dd6651 100644
--- a/modules/video_capture/linux/device_info_v4l2.cc
+++ b/modules/video_capture/linux/device_info_v4l2.cc
@@ -133,11 +133,6 @@ int DeviceInfoV4l2::ProcessInotifyEvents()
return 0;
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.cc b/modules/rtp_rtcp/source/rtp_header_extensions.cc
index a57d9e7f62..de29fd2075 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -185,6 +185,7 @@ bool AudioLevel::Write(rtc::ArrayView<uint8_t> data,
return true;
}
-void DeviceInfoV4l2::InotifyEventThread(void* obj)
-{
- static_cast<DeviceInfoLinux*> (obj)->InotifyProcess();
-}
-
void DeviceInfoV4l2::InotifyProcess()
{
_fd_v4l = inotify_init();
@@ -163,16 +158,14 @@ void DeviceInfoV4l2::InotifyProcess()
+#if !defined(WEBRTC_MOZILLA_BUILD)
// An RTP Header Extension for Mixer-to-Client Audio Level Indication
//
// https://tools.ietf.org/html/rfc6465
@@ -237,6 +238,7 @@ bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
}
return true;
}
+#endif
DeviceInfoV4l2::DeviceInfoV4l2() : DeviceInfoImpl()
#ifdef WEBRTC_LINUX
- , _inotifyEventThread(new rtc::PlatformThread(
- InotifyEventThread, this, "InotifyEventThread"))
, _isShutdown(false)
#endif
{
#ifdef WEBRTC_LINUX
- if (_inotifyEventThread)
- {
- _inotifyEventThread->Start();
- }
+ _inotifyEventThread = rtc::PlatformThread::SpawnJoinable(
+ [this] {
+ InotifyProcess();
+ }, "InotifyEventThread");
#endif
// From RFC 5450: Transmission Time Offsets in RTP Streams.
//
@@ -446,6 +448,7 @@ bool PlayoutDelayLimits::Write(rtc::ArrayView<uint8_t> data,
return true;
}
@@ -184,9 +177,8 @@ DeviceInfoV4l2::~DeviceInfoV4l2() {
#ifdef WEBRTC_LINUX
_isShutdown = true;
+#if defined(WEBRTC_MOZILLA_BUILD)
// CSRCAudioLevel
// Sample Audio Level Encoding Using the One-Byte Header Format
// Note that the range of len is 1 to 15 which is encoded as 0 to 14
@@ -484,6 +487,7 @@ bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
// This extension if used must have at least one audio level
return csrcAudioLevels.numAudioLevels;
}
+#endif
- if (_inotifyEventThread) {
- _inotifyEventThread->Stop();
- _inotifyEventThread = nullptr;
+ if (!_inotifyEventThread.empty()) {
+ _inotifyEventThread.Finalize();
// Video Content Type.
//
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.h b/modules/rtp_rtcp/source/rtp_header_extensions.h
index 89c73955a2..4b4984bf6d 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -88,6 +88,7 @@ class AudioLevel {
uint8_t audio_level);
};
+#if !defined(WEBRTC_MOZILLA_BUILD)
class CsrcAudioLevel {
public:
static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
@@ -102,6 +103,7 @@ class CsrcAudioLevel {
static bool Write(rtc::ArrayView<uint8_t> data,
rtc::ArrayView<const uint8_t> csrc_audio_levels);
};
+#endif
class TransmissionOffset {
public:
@@ -292,6 +294,7 @@ class ColorSpaceExtension {
static size_t WriteLuminance(uint8_t* data, float f, int denominator);
};
+#if defined(WEBRTC_MOZILLA_BUILD)
class CsrcAudioLevel {
public:
static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
@@ -306,6 +309,7 @@ class CsrcAudioLevel {
static size_t ValueSize(const CsrcAudioLevelList& csrcAudioLevels);
static bool Write(rtc::ArrayView<uint8_t> data, const CsrcAudioLevelList& csrcAudioLevels);
};
+#endif
// Base extension class for RTP header extensions which are strings.
// Subclasses must defined kId and kUri static constexpr members.
diff --git a/modules/rtp_rtcp/source/rtp_packet.cc b/modules/rtp_rtcp/source/rtp_packet.cc
index 9495841984..fd2f5c5ae8 100644
--- a/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/modules/rtp_rtcp/source/rtp_packet.cc
@@ -187,7 +187,9 @@ void RtpPacket::ZeroMutableExtensions() {
break;
}
case RTPExtensionType::kRtpExtensionAudioLevel:
+#if !defined(WEBRTC_MOZILLA_BUILD)
case RTPExtensionType::kRtpExtensionCsrcAudioLevel:
+#endif
case RTPExtensionType::kRtpExtensionAbsoluteCaptureTime:
case RTPExtensionType::kRtpExtensionColorSpace:
case RTPExtensionType::kRtpExtensionGenericFrameDescriptor:
@@ -205,10 +207,12 @@ void RtpPacket::ZeroMutableExtensions() {
// Non-mutable extension. Don't change it.
break;
}
+#if defined(WEBRTC_MOZILLA_BUILD)
case RTPExtensionType::kRtpExtensionCsrcAudioLevel: {
// TODO: This is a Mozilla addition, we need to add a handler for this.
RTC_CHECK(false);
}
+#endif
}
#endif
}
}
diff --git a/modules/video_capture/linux/device_info_v4l2.h b/modules/video_capture/linux/device_info_v4l2.h
index 119cb07ab8..0bec3eb765 100644
--- a/modules/video_capture/linux/device_info_v4l2.h
+++ b/modules/video_capture/linux/device_info_v4l2.h
@@ -60,8 +60,7 @@ class DeviceInfoV4l2 : public DeviceInfoImpl {
int EventCheck(int fd);
int HandleEvents(int fd);
int ProcessInotifyEvents();
- std::unique_ptr<rtc::PlatformThread> _inotifyEventThread;
- static void InotifyEventThread(void*);
+ rtc::PlatformThread _inotifyEventThread;
void InotifyProcess();
int _fd_v4l, _fd_dev, _wd_v4l, _wd_dev; /* accessed on InotifyEventThread thread */
std::atomic<bool> _isShutdown;
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 0ed7243d0a..336a117f4e 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -108,7 +108,9 @@ bool IsNonVolatile(RTPExtensionType type) {
switch (type) {
case kRtpExtensionTransmissionTimeOffset:
case kRtpExtensionAudioLevel:
+#if !defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel:
+#endif
case kRtpExtensionAbsoluteSendTime:
case kRtpExtensionTransportSequenceNumber:
case kRtpExtensionTransportSequenceNumber02:
@@ -132,10 +134,12 @@ bool IsNonVolatile(RTPExtensionType type) {
case kRtpExtensionNumberOfExtensions:
RTC_DCHECK_NOTREACHED();
return false;
+#if defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel:
// TODO: Mozilla implement for CsrcAudioLevel
RTC_CHECK(false);
return false;
+#endif
}
RTC_CHECK_NOTREACHED();
}
diff --git a/test/fuzzers/rtp_packet_fuzzer.cc b/test/fuzzers/rtp_packet_fuzzer.cc
index 0e10a8fa3a..5d117529bb 100644
--- a/test/fuzzers/rtp_packet_fuzzer.cc
+++ b/test/fuzzers/rtp_packet_fuzzer.cc
@@ -77,11 +77,13 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
uint8_t audio_level;
packet.GetExtension<AudioLevel>(&voice_activity, &audio_level);
break;
+#if !defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel: {
std::vector<uint8_t> audio_levels;
packet.GetExtension<CsrcAudioLevel>(&audio_levels);
break;
}
+#endif
case kRtpExtensionAbsoluteSendTime:
uint32_t sendtime;
packet.GetExtension<AbsoluteSendTime>(&sendtime);
@@ -164,11 +166,13 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
// This extension requires state to read and so complicated that
// deserves own fuzzer.
break;
+#if defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel: {
CsrcAudioLevelList levels;
packet.GetExtension<CsrcAudioLevel>(&levels);
break;
}
+#endif
}
}
--
2.34.1

185
third_party/libwebrtc/moz-patch-stack/0066.patch поставляемый
Просмотреть файл

@ -1,174 +1,25 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 18 Jan 2021 11:07:00 +0100
Subject: Bug 1766646 - (fix-ae0d117d51) ifdef our Csrc impl vs upstream's
impl, see Bug 1771332.
From: Michael Froman <mjfroman@mac.com>
Date: Wed, 1 Jun 2022 12:47:00 -0500
Subject: Bug 1766646 - (fix-f137b75a4d) specify default constructor on
config.emplace(...)
---
modules/rtp_rtcp/source/rtp_header_extensions.cc | 4 ++++
modules/rtp_rtcp/source/rtp_header_extensions.h | 4 ++++
modules/rtp_rtcp/source/rtp_packet.cc | 4 ++++
modules/rtp_rtcp/source/rtp_sender.cc | 4 ++++
test/fuzzers/rtp_packet_fuzzer.cc | 4 ++++
5 files changed, 20 insertions(+)
modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.cc b/modules/rtp_rtcp/source/rtp_header_extensions.cc
index a57d9e7f62..de29fd2075 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.cc
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.cc
@@ -185,6 +185,7 @@ bool AudioLevel::Write(rtc::ArrayView<uint8_t> data,
return true;
}
+#if !defined(WEBRTC_MOZILLA_BUILD)
// An RTP Header Extension for Mixer-to-Client Audio Level Indication
//
// https://tools.ietf.org/html/rfc6465
@@ -237,6 +238,7 @@ bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
index b4d3ae8c1f..b6efdeee9e 100644
--- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
+++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
@@ -457,7 +457,7 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig(
if (!enabled.Get()) {
return config;
}
return true;
}
+#endif
// From RFC 5450: Transmission Time Offsets in RTP Streams.
//
@@ -446,6 +448,7 @@ bool PlayoutDelayLimits::Write(rtc::ArrayView<uint8_t> data,
return true;
}
+#if defined(WEBRTC_MOZILLA_BUILD)
// CSRCAudioLevel
// Sample Audio Level Encoding Using the One-Byte Header Format
// Note that the range of len is 1 to 15 which is encoded as 0 to 14
@@ -484,6 +487,7 @@ bool CsrcAudioLevel::Write(rtc::ArrayView<uint8_t> data,
// This extension if used must have at least one audio level
return csrcAudioLevels.numAudioLevels;
}
+#endif
// Video Content Type.
//
diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.h b/modules/rtp_rtcp/source/rtp_header_extensions.h
index 89c73955a2..4b4984bf6d 100644
--- a/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -88,6 +88,7 @@ class AudioLevel {
uint8_t audio_level);
};
+#if !defined(WEBRTC_MOZILLA_BUILD)
class CsrcAudioLevel {
public:
static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
@@ -102,6 +103,7 @@ class CsrcAudioLevel {
static bool Write(rtc::ArrayView<uint8_t> data,
rtc::ArrayView<const uint8_t> csrc_audio_levels);
};
+#endif
class TransmissionOffset {
public:
@@ -292,6 +294,7 @@ class ColorSpaceExtension {
static size_t WriteLuminance(uint8_t* data, float f, int denominator);
};
+#if defined(WEBRTC_MOZILLA_BUILD)
class CsrcAudioLevel {
public:
static constexpr RTPExtensionType kId = kRtpExtensionCsrcAudioLevel;
@@ -306,6 +309,7 @@ class CsrcAudioLevel {
static size_t ValueSize(const CsrcAudioLevelList& csrcAudioLevels);
static bool Write(rtc::ArrayView<uint8_t> data, const CsrcAudioLevelList& csrcAudioLevels);
};
+#endif
// Base extension class for RTP header extensions which are strings.
// Subclasses must defined kId and kUri static constexpr members.
diff --git a/modules/rtp_rtcp/source/rtp_packet.cc b/modules/rtp_rtcp/source/rtp_packet.cc
index 9495841984..fd2f5c5ae8 100644
--- a/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/modules/rtp_rtcp/source/rtp_packet.cc
@@ -187,7 +187,9 @@ void RtpPacket::ZeroMutableExtensions() {
break;
}
case RTPExtensionType::kRtpExtensionAudioLevel:
+#if !defined(WEBRTC_MOZILLA_BUILD)
case RTPExtensionType::kRtpExtensionCsrcAudioLevel:
+#endif
case RTPExtensionType::kRtpExtensionAbsoluteCaptureTime:
case RTPExtensionType::kRtpExtensionColorSpace:
case RTPExtensionType::kRtpExtensionGenericFrameDescriptor:
@@ -205,10 +207,12 @@ void RtpPacket::ZeroMutableExtensions() {
// Non-mutable extension. Don't change it.
break;
}
+#if defined(WEBRTC_MOZILLA_BUILD)
case RTPExtensionType::kRtpExtensionCsrcAudioLevel: {
// TODO: This is a Mozilla addition, we need to add a handler for this.
RTC_CHECK(false);
}
+#endif
}
}
}
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 0ed7243d0a..336a117f4e 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -108,7 +108,9 @@ bool IsNonVolatile(RTPExtensionType type) {
switch (type) {
case kRtpExtensionTransmissionTimeOffset:
case kRtpExtensionAudioLevel:
+#if !defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel:
+#endif
case kRtpExtensionAbsoluteSendTime:
case kRtpExtensionTransportSequenceNumber:
case kRtpExtensionTransportSequenceNumber02:
@@ -132,10 +134,12 @@ bool IsNonVolatile(RTPExtensionType type) {
case kRtpExtensionNumberOfExtensions:
RTC_DCHECK_NOTREACHED();
return false;
+#if defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel:
// TODO: Mozilla implement for CsrcAudioLevel
RTC_CHECK(false);
return false;
+#endif
}
RTC_CHECK_NOTREACHED();
}
diff --git a/test/fuzzers/rtp_packet_fuzzer.cc b/test/fuzzers/rtp_packet_fuzzer.cc
index 0e10a8fa3a..5d117529bb 100644
--- a/test/fuzzers/rtp_packet_fuzzer.cc
+++ b/test/fuzzers/rtp_packet_fuzzer.cc
@@ -77,11 +77,13 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
uint8_t audio_level;
packet.GetExtension<AudioLevel>(&voice_activity, &audio_level);
break;
+#if !defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel: {
std::vector<uint8_t> audio_levels;
packet.GetExtension<CsrcAudioLevel>(&audio_levels);
break;
}
+#endif
case kRtpExtensionAbsoluteSendTime:
uint32_t sendtime;
packet.GetExtension<AbsoluteSendTime>(&sendtime);
@@ -164,11 +166,13 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
// This extension requires state to read and so complicated that
// deserves own fuzzer.
break;
+#if defined(WEBRTC_MOZILLA_BUILD)
case kRtpExtensionCsrcAudioLevel: {
CsrcAudioLevelList levels;
packet.GetExtension<CsrcAudioLevel>(&levels);
break;
}
+#endif
}
}
- config.emplace();
+ config.emplace(Config());
config->bandwidth_rampup_upper_bound_factor =
bandwidth_rampup_upper_bound_factor.Get();
config->rampup_acceleration_max_factor = rampup_acceleration_max_factor.Get();
--
2.34.1

38
third_party/libwebrtc/moz-patch-stack/0067.patch поставляемый
Просмотреть файл

@ -1,25 +1,27 @@
From: Michael Froman <mjfroman@mac.com>
Date: Wed, 1 Jun 2022 12:47:00 -0500
Subject: Bug 1766646 - (fix-f137b75a4d) specify default constructor on
config.emplace(...)
Date: Tue, 21 Jun 2022 11:11:09 -0500
Subject: Bug 1773223 - Generate webrtc moz.builds for all platforms at once.
r=mjf,firefox-build-system-reviewers,ahochheiden
---
modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
build_overrides/build.gni | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
index b4d3ae8c1f..b6efdeee9e 100644
--- a/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
+++ b/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
@@ -457,7 +457,7 @@ absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig(
if (!enabled.Get()) {
return config;
}
- config.emplace();
+ config.emplace(Config());
config->bandwidth_rampup_upper_bound_factor =
bandwidth_rampup_upper_bound_factor.Get();
config->rampup_acceleration_max_factor = rampup_acceleration_max_factor.Get();
diff --git a/build_overrides/build.gni b/build_overrides/build.gni
index 137b6a40b2..e662c94ece 100644
--- a/build_overrides/build.gni
+++ b/build_overrides/build.gni
@@ -45,6 +45,10 @@ if (host_os == "mac" || host_os == "linux") {
use_system_xcode = _result == 0
}
+use_system_xcode = false
+xcode_version = "10.15"
+mac_xcode_version = "default"
+
declare_args() {
# WebRTC doesn't depend on //base from production code but only for testing
# purposes. In any case, it doesn't depend on //third_party/perfetto which
--
2.34.1

39
third_party/libwebrtc/moz-patch-stack/0068.patch поставляемый
Просмотреть файл

@ -1,27 +1,26 @@
From: Michael Froman <mjfroman@mac.com>
Date: Tue, 21 Jun 2022 11:11:09 -0500
Subject: Bug 1773223 - Generate webrtc moz.builds for all platforms at once.
r=mjf,firefox-build-system-reviewers,ahochheiden
Date: Tue, 21 Jun 2022 11:17:46 -0500
Subject: Bug 1772380 - to upstream - ref count this in lambda capture
---
build_overrides/build.gni | 4 ++++
1 file changed, 4 insertions(+)
modules/video_capture/linux/video_capture_v4l2.cc | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/build_overrides/build.gni b/build_overrides/build.gni
index 137b6a40b2..e662c94ece 100644
--- a/build_overrides/build.gni
+++ b/build_overrides/build.gni
@@ -45,6 +45,10 @@ if (host_os == "mac" || host_os == "linux") {
use_system_xcode = _result == 0
}
+use_system_xcode = false
+xcode_version = "10.15"
+mac_xcode_version = "default"
+
declare_args() {
# WebRTC doesn't depend on //base from production code but only for testing
# purposes. In any case, it doesn't depend on //third_party/perfetto which
diff --git a/modules/video_capture/linux/video_capture_v4l2.cc b/modules/video_capture/linux/video_capture_v4l2.cc
index b527a331e4..c7dcb722bc 100644
--- a/modules/video_capture/linux/video_capture_v4l2.cc
+++ b/modules/video_capture/linux/video_capture_v4l2.cc
@@ -250,8 +250,8 @@ int32_t VideoCaptureModuleV4L2::StartCapture(
if (_captureThread.empty()) {
quit_ = false;
_captureThread = rtc::PlatformThread::SpawnJoinable(
- [this] {
- while (CaptureProcess()) {
+ [self = rtc::scoped_refptr(this)] {
+ while (self->CaptureProcess()) {
}
},
"CaptureThread",
--
2.34.1

35
third_party/libwebrtc/moz-patch-stack/0069.patch поставляемый
Просмотреть файл

@ -1,26 +1,23 @@
From: Michael Froman <mjfroman@mac.com>
Date: Tue, 21 Jun 2022 11:17:46 -0500
Subject: Bug 1772380 - to upstream - ref count this in lambda capture
Date: Wed, 13 Jul 2022 12:50:36 -0500
Subject: Bug 1766646 - (fix-4d12174ca5) add missing include to fix build
---
modules/video_capture/linux/video_capture_v4l2.cc | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
modules/audio_processing/aec3/multi_channel_content_detector.h | 1 +
1 file changed, 1 insertion(+)
diff --git a/modules/video_capture/linux/video_capture_v4l2.cc b/modules/video_capture/linux/video_capture_v4l2.cc
index b527a331e4..c7dcb722bc 100644
--- a/modules/video_capture/linux/video_capture_v4l2.cc
+++ b/modules/video_capture/linux/video_capture_v4l2.cc
@@ -250,8 +250,8 @@ int32_t VideoCaptureModuleV4L2::StartCapture(
if (_captureThread.empty()) {
quit_ = false;
_captureThread = rtc::PlatformThread::SpawnJoinable(
- [this] {
- while (CaptureProcess()) {
+ [self = rtc::scoped_refptr(this)] {
+ while (self->CaptureProcess()) {
}
},
"CaptureThread",
diff --git a/modules/audio_processing/aec3/multi_channel_content_detector.h b/modules/audio_processing/aec3/multi_channel_content_detector.h
index be8717f3af..1742c5fc17 100644
--- a/modules/audio_processing/aec3/multi_channel_content_detector.h
+++ b/modules/audio_processing/aec3/multi_channel_content_detector.h
@@ -11,6 +11,7 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC3_MULTI_CHANNEL_CONTENT_DETECTOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_MULTI_CHANNEL_CONTENT_DETECTOR_H_
+#include <memory>
#include <stddef.h>
#include <memory>
--
2.34.1

60
third_party/libwebrtc/moz-patch-stack/0070.patch поставляемый
Просмотреть файл

@ -1,23 +1,51 @@
From: Michael Froman <mjfroman@mac.com>
Date: Wed, 13 Jul 2022 12:50:36 -0500
Subject: Bug 1766646 - (fix-4d12174ca5) add missing include to fix build
Date: Wed, 3 Aug 2022 20:21:25 -0500
Subject: Bug 1780582 - work around generating VideoFrameBufferType;r=mjf
---
modules/audio_processing/aec3/multi_channel_content_detector.h | 1 +
1 file changed, 1 insertion(+)
.../api/org/webrtc/VideoFrameBufferType.java | 33 +++++++++++++++++++
1 file changed, 33 insertions(+)
create mode 100644 sdk/android/api/org/webrtc/VideoFrameBufferType.java
diff --git a/modules/audio_processing/aec3/multi_channel_content_detector.h b/modules/audio_processing/aec3/multi_channel_content_detector.h
index be8717f3af..1742c5fc17 100644
--- a/modules/audio_processing/aec3/multi_channel_content_detector.h
+++ b/modules/audio_processing/aec3/multi_channel_content_detector.h
@@ -11,6 +11,7 @@
#ifndef MODULES_AUDIO_PROCESSING_AEC3_MULTI_CHANNEL_CONTENT_DETECTOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_MULTI_CHANNEL_CONTENT_DETECTOR_H_
+#include <memory>
#include <stddef.h>
#include <memory>
diff --git a/sdk/android/api/org/webrtc/VideoFrameBufferType.java b/sdk/android/api/org/webrtc/VideoFrameBufferType.java
new file mode 100644
index 0000000000..7b05b88cba
--- /dev/null
+++ b/sdk/android/api/org/webrtc/VideoFrameBufferType.java
@@ -0,0 +1,33 @@
+
+// Copyright 2022 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// This file is autogenerated by
+// java_cpp_enum.py
+// From
+// ../../api/video/video_frame_buffer.h
+
+package org.webrtc;
+
+import androidx.annotation.IntDef;
+
+import java.lang.annotation.Retention;
+import java.lang.annotation.RetentionPolicy;
+
+@IntDef({
+ VideoFrameBufferType.NATIVE, VideoFrameBufferType.I420, VideoFrameBufferType.I420A,
+ VideoFrameBufferType.I422, VideoFrameBufferType.I444, VideoFrameBufferType.I010,
+ VideoFrameBufferType.I210, VideoFrameBufferType.NV12
+})
+@Retention(RetentionPolicy.SOURCE)
+public @interface VideoFrameBufferType {
+ int NATIVE = 0;
+ int I420 = 1;
+ int I420A = 2;
+ int I422 = 3;
+ int I444 = 4;
+ int I010 = 5;
+ int I210 = 6;
+ int NV12 = 7;
+}
--
2.34.1

95
third_party/libwebrtc/moz-patch-stack/0071.patch поставляемый
Просмотреть файл

@ -1,51 +1,56 @@
From: Michael Froman <mjfroman@mac.com>
Date: Wed, 3 Aug 2022 20:21:25 -0500
Subject: Bug 1780582 - work around generating VideoFrameBufferType;r=mjf
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 5 Sep 2022 13:56:00 +0000
Subject: Bug 1786502 - Lock access to DeviceInfo devicechange callbacks.
r=webrtc-reviewers,jib
Differential Revision: https://phabricator.services.mozilla.com/D155365
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e826dfadfe1264c59d9b13e3c17d6f75a40f5c33
---
.../api/org/webrtc/VideoFrameBufferType.java | 33 +++++++++++++++++++
1 file changed, 33 insertions(+)
create mode 100644 sdk/android/api/org/webrtc/VideoFrameBufferType.java
modules/video_capture/video_capture.h | 8 +++++++-
1 file changed, 7 insertions(+), 1 deletion(-)
diff --git a/sdk/android/api/org/webrtc/VideoFrameBufferType.java b/sdk/android/api/org/webrtc/VideoFrameBufferType.java
new file mode 100644
index 0000000000..7b05b88cba
--- /dev/null
+++ b/sdk/android/api/org/webrtc/VideoFrameBufferType.java
@@ -0,0 +1,33 @@
+
+// Copyright 2022 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// This file is autogenerated by
+// java_cpp_enum.py
+// From
+// ../../api/video/video_frame_buffer.h
+
+package org.webrtc;
+
+import androidx.annotation.IntDef;
+
+import java.lang.annotation.Retention;
+import java.lang.annotation.RetentionPolicy;
+
+@IntDef({
+ VideoFrameBufferType.NATIVE, VideoFrameBufferType.I420, VideoFrameBufferType.I420A,
+ VideoFrameBufferType.I422, VideoFrameBufferType.I444, VideoFrameBufferType.I010,
+ VideoFrameBufferType.I210, VideoFrameBufferType.NV12
+})
+@Retention(RetentionPolicy.SOURCE)
+public @interface VideoFrameBufferType {
+ int NATIVE = 0;
+ int I420 = 1;
+ int I420A = 2;
+ int I422 = 3;
+ int I444 = 4;
+ int I010 = 5;
+ int I210 = 6;
+ int NV12 = 7;
+}
diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
index 258bc7f810..ad1b341b62 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -16,6 +16,8 @@
#include "modules/desktop_capture/desktop_capture_types.h"
#include "modules/video_capture/raw_video_sink_interface.h"
#include "modules/video_capture/video_capture_defines.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
#include <set>
#if defined(ANDROID)
@@ -44,15 +46,18 @@ class VideoCaptureModule : public rtc::RefCountInterface {
virtual uint32_t NumberOfDevices() = 0;
virtual int32_t Refresh() = 0;
virtual void DeviceChange() {
+ MutexLock lock(&_inputCallbacksMutex);
for (auto inputCallBack : _inputCallBacks) {
inputCallBack->OnDeviceChange();
}
}
virtual void RegisterVideoInputFeedBack(VideoInputFeedBack* callBack) {
+ MutexLock lock(&_inputCallbacksMutex);
_inputCallBacks.insert(callBack);
}
virtual void DeRegisterVideoInputFeedBack(VideoInputFeedBack* callBack) {
+ MutexLock lock(&_inputCallbacksMutex);
auto it = _inputCallBacks.find(callBack);
if (it != _inputCallBacks.end()) {
_inputCallBacks.erase(it);
@@ -106,7 +111,8 @@ class VideoCaptureModule : public rtc::RefCountInterface {
virtual ~DeviceInfo() {}
private:
- std::set<VideoInputFeedBack*> _inputCallBacks;
+ Mutex _inputCallbacksMutex;
+ std::set<VideoInputFeedBack*> _inputCallBacks RTC_GUARDED_BY(_inputCallbacksMutex);
};
// Register capture data callback
--
2.34.1

69
third_party/libwebrtc/moz-patch-stack/0072.patch поставляемый
Просмотреть файл

@ -1,56 +1,27 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 5 Sep 2022 13:56:00 +0000
Subject: Bug 1786502 - Lock access to DeviceInfo devicechange callbacks.
r=webrtc-reviewers,jib
From: Michael Froman <mfroman@mozilla.com>
Date: Wed, 28 Sep 2022 14:19:00 -0500
Subject: Bug 1790097 - (fix-e66b83f8ad) restore deleted include file for
std::towupper
Differential Revision: https://phabricator.services.mozilla.com/D155365
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e826dfadfe1264c59d9b13e3c17d6f75a40f5c33
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/2789837da50ca0d6bce9a06bd036bdd049128f20
---
modules/video_capture/video_capture.h | 8 +++++++-
1 file changed, 7 insertions(+), 1 deletion(-)
.../desktop_capture/win/full_screen_win_application_handler.cc | 3 +++
1 file changed, 3 insertions(+)
diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
index 258bc7f810..ad1b341b62 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -16,6 +16,8 @@
#include "modules/desktop_capture/desktop_capture_types.h"
#include "modules/video_capture/raw_video_sink_interface.h"
#include "modules/video_capture/video_capture_defines.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
#include <set>
diff --git a/modules/desktop_capture/win/full_screen_win_application_handler.cc b/modules/desktop_capture/win/full_screen_win_application_handler.cc
index f7047dba06..c6143ef785 100644
--- a/modules/desktop_capture/win/full_screen_win_application_handler.cc
+++ b/modules/desktop_capture/win/full_screen_win_application_handler.cc
@@ -11,6 +11,9 @@
#include "modules/desktop_capture/win/full_screen_win_application_handler.h"
#if defined(ANDROID)
@@ -44,15 +46,18 @@ class VideoCaptureModule : public rtc::RefCountInterface {
virtual uint32_t NumberOfDevices() = 0;
virtual int32_t Refresh() = 0;
virtual void DeviceChange() {
+ MutexLock lock(&_inputCallbacksMutex);
for (auto inputCallBack : _inputCallBacks) {
inputCallBack->OnDeviceChange();
}
}
virtual void RegisterVideoInputFeedBack(VideoInputFeedBack* callBack) {
+ MutexLock lock(&_inputCallbacksMutex);
_inputCallBacks.insert(callBack);
}
virtual void DeRegisterVideoInputFeedBack(VideoInputFeedBack* callBack) {
+ MutexLock lock(&_inputCallbacksMutex);
auto it = _inputCallBacks.find(callBack);
if (it != _inputCallBacks.end()) {
_inputCallBacks.erase(it);
@@ -106,7 +111,8 @@ class VideoCaptureModule : public rtc::RefCountInterface {
virtual ~DeviceInfo() {}
private:
- std::set<VideoInputFeedBack*> _inputCallBacks;
+ Mutex _inputCallbacksMutex;
+ std::set<VideoInputFeedBack*> _inputCallBacks RTC_GUARDED_BY(_inputCallbacksMutex);
};
// Register capture data callback
#include <algorithm>
+#if defined(WEBRTC_MOZILLA_BUILD)
+#include <cwctype>
+#endif
#include <memory>
#include <string>
#include <vector>
--
2.34.1

63
third_party/libwebrtc/moz-patch-stack/0073.patch поставляемый
Просмотреть файл

@ -1,27 +1,52 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Wed, 28 Sep 2022 14:19:00 -0500
Subject: Bug 1790097 - (fix-e66b83f8ad) restore deleted include file for
std::towupper
Date: Mon, 24 Oct 2022 13:00:00 -0500
Subject: Bug 1797161 - pt1 - tweak BUILD.gn around task_queue_win usage. r?ng!
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/2789837da50ca0d6bce9a06bd036bdd049128f20
Add assurance that we will not build task_queue_win.cc to avoid
possible win32k API usage.
Differential Revision: https://phabricator.services.mozilla.com/D160115
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f097eb8cbd8b7686ce306a46a4db691194fd39c1
---
.../desktop_capture/win/full_screen_win_application_handler.cc | 3 +++
1 file changed, 3 insertions(+)
api/task_queue/BUILD.gn | 5 +++++
rtc_base/BUILD.gn | 4 ++++
2 files changed, 9 insertions(+)
diff --git a/modules/desktop_capture/win/full_screen_win_application_handler.cc b/modules/desktop_capture/win/full_screen_win_application_handler.cc
index f7047dba06..c6143ef785 100644
--- a/modules/desktop_capture/win/full_screen_win_application_handler.cc
+++ b/modules/desktop_capture/win/full_screen_win_application_handler.cc
@@ -11,6 +11,9 @@
#include "modules/desktop_capture/win/full_screen_win_application_handler.h"
diff --git a/api/task_queue/BUILD.gn b/api/task_queue/BUILD.gn
index c9b4a5d0ec..1c342cb57e 100644
--- a/api/task_queue/BUILD.gn
+++ b/api/task_queue/BUILD.gn
@@ -30,6 +30,11 @@ rtc_library("task_queue") {
]
}
#include <algorithm>
+#if defined(WEBRTC_MOZILLA_BUILD)
+#include <cwctype>
+#endif
#include <memory>
#include <string>
#include <vector>
+# Mozilla - we want to ensure that rtc_include_tests is set to false
+# to guarantee that default_task_queue_factory is not used so we
+# know that remaining win32k code in task_queue_win.cc is not built.
+# See Bug 1797161 for more info.
+assert(!rtc_include_tests, "Mozilla - verify rtc_include_tests is off")
if (rtc_include_tests) {
rtc_library("task_queue_test") {
visibility = [ "*" ]
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 7e162cecbb..3cd0bfff06 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -686,10 +686,14 @@ if (is_mac || is_ios) {
if (is_win) {
rtc_library("rtc_task_queue_win") {
visibility = [ "../api/task_queue:default_task_queue_factory" ]
+# See Bug 1797161 for more info. Remove from build until win32k
+# usage is removed.
+if (!build_with_mozilla) {
sources = [
"task_queue_win.cc",
"task_queue_win.h",
]
+}
deps = [
":checks",
":logging",
--
2.34.1

70
third_party/libwebrtc/moz-patch-stack/0074.patch поставляемый
Просмотреть файл

@ -1,52 +1,34 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Mon, 24 Oct 2022 13:00:00 -0500
Subject: Bug 1797161 - pt1 - tweak BUILD.gn around task_queue_win usage. r?ng!
Date: Mon, 24 Oct 2022 14:03:00 -0500
Subject: Bug 1797161 - pt3 - add static_assert to ensure we don't include
task_queue_win.cc in Mozilla builds. r?ng!
Add assurance that we will not build task_queue_win.cc to avoid
possible win32k API usage.
Differential Revision: https://phabricator.services.mozilla.com/D160115
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f097eb8cbd8b7686ce306a46a4db691194fd39c1
Differential Revision: https://phabricator.services.mozilla.com/D160117
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/50b15e036924203147e34ec20e2689fe4a847645
---
api/task_queue/BUILD.gn | 5 +++++
rtc_base/BUILD.gn | 4 ++++
2 files changed, 9 insertions(+)
rtc_base/task_queue_win.cc | 9 +++++++++
1 file changed, 9 insertions(+)
diff --git a/api/task_queue/BUILD.gn b/api/task_queue/BUILD.gn
index c9b4a5d0ec..1c342cb57e 100644
--- a/api/task_queue/BUILD.gn
+++ b/api/task_queue/BUILD.gn
@@ -30,6 +30,11 @@ rtc_library("task_queue") {
]
}
diff --git a/rtc_base/task_queue_win.cc b/rtc_base/task_queue_win.cc
index 9ea7fc60ae..6da3094548 100644
--- a/rtc_base/task_queue_win.cc
+++ b/rtc_base/task_queue_win.cc
@@ -8,6 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+# Mozilla - we want to ensure that rtc_include_tests is set to false
+# to guarantee that default_task_queue_factory is not used so we
+# know that remaining win32k code in task_queue_win.cc is not built.
+# See Bug 1797161 for more info.
+assert(!rtc_include_tests, "Mozilla - verify rtc_include_tests is off")
if (rtc_include_tests) {
rtc_library("task_queue_test") {
visibility = [ "*" ]
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 7e162cecbb..3cd0bfff06 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -686,10 +686,14 @@ if (is_mac || is_ios) {
if (is_win) {
rtc_library("rtc_task_queue_win") {
visibility = [ "../api/task_queue:default_task_queue_factory" ]
+# See Bug 1797161 for more info. Remove from build until win32k
+# usage is removed.
+if (!build_with_mozilla) {
sources = [
"task_queue_win.cc",
"task_queue_win.h",
]
+}
deps = [
":checks",
":logging",
+// Mozilla - this file should not be included in Mozilla builds until
+// win32k API usage is removed. This was once done in Bug 1395259, but
+// the upstreaming attempt stalled. Until win32k usage is officially
+// removed upstream, we have reverted to upstream's version of the file
+// (to reduce or elminate merge conflicts), and a static assert is
+// placed here to ensure this file isn't accidentally included in the
+// Mozilla build.
+static_assert(false, "This file should not be built, see Bug 1797161.");
+
#include "rtc_base/task_queue_win.h"
// clang-format off
--
2.34.1

97
third_party/libwebrtc/moz-patch-stack/0075.patch поставляемый
Просмотреть файл

@ -1,34 +1,81 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Mon, 24 Oct 2022 14:03:00 -0500
Subject: Bug 1797161 - pt3 - add static_assert to ensure we don't include
task_queue_win.cc in Mozilla builds. r?ng!
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 12 Dec 2022 15:47:00 +0000
Subject: Bug 1451394 - Expose mac camera capture backend in .gn and switch it
to gecko libyuv. r=webrtc-reviewers,mjf
Differential Revision: https://phabricator.services.mozilla.com/D160117
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/50b15e036924203147e34ec20e2689fe4a847645
Differential Revision: https://phabricator.services.mozilla.com/D163682
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b0658888969395dca938597783c8a377b9bea209
---
rtc_base/task_queue_win.cc | 9 +++++++++
1 file changed, 9 insertions(+)
BUILD.gn | 4 ++++
sdk/BUILD.gn | 6 ++++++
2 files changed, 10 insertions(+)
diff --git a/rtc_base/task_queue_win.cc b/rtc_base/task_queue_win.cc
index 9ea7fc60ae..6da3094548 100644
--- a/rtc_base/task_queue_win.cc
+++ b/rtc_base/task_queue_win.cc
@@ -8,6 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
diff --git a/BUILD.gn b/BUILD.gn
index 03880dbb09..873aec3850 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -559,6 +559,10 @@ if (!build_with_chromium) {
]
}
+// Mozilla - this file should not be included in Mozilla builds until
+// win32k API usage is removed. This was once done in Bug 1395259, but
+// the upstreaming attempt stalled. Until win32k usage is officially
+// removed upstream, we have reverted to upstream's version of the file
+// (to reduce or elminate merge conflicts), and a static assert is
+// placed here to ensure this file isn't accidentally included in the
+// Mozilla build.
+static_assert(false, "This file should not be built, see Bug 1797161.");
+ if (build_with_mozilla && is_mac) {
+ deps += [ "sdk:videocapture_objc" ]
+ }
+
#include "rtc_base/task_queue_win.h"
if (rtc_enable_protobuf) {
deps += [ "logging:rtc_event_log_proto" ]
}
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index a361656a59..65d1b10124 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -449,6 +449,7 @@ if (is_ios || is_mac) {
]
}
// clang-format off
+ if (!build_with_mozilla) {
rtc_library("videosource_objc") {
sources = [
"objc/api/peerconnection/RTCVideoSource+Private.h",
@@ -478,6 +479,7 @@ if (is_ios || is_mac) {
":used_from_extension",
]
}
+ }
rtc_library("videoframebuffer_objc") {
visibility = [ "*" ]
@@ -510,6 +512,7 @@ if (is_ios || is_mac) {
]
}
+ if (!build_with_mozilla) {
rtc_library("opengl_objc") {
sources = [
"objc/components/renderer/opengl/RTCDefaultShader.h",
@@ -662,6 +665,7 @@ if (is_ios || is_mac) {
":videoframebuffer_objc",
]
}
+ }
rtc_library("videocapture_objc") {
visibility = [ "*" ]
@@ -690,6 +694,7 @@ if (is_ios || is_mac) {
]
}
+ if (!build_with_mozilla) {
rtc_library("videocodec_objc") {
visibility = [ "*" ]
configs += [ "..:no_global_constructors" ]
@@ -1747,5 +1752,6 @@ if (is_ios || is_mac) {
"VideoToolbox.framework",
]
}
+ }
}
}
--
2.34.1

92
third_party/libwebrtc/moz-patch-stack/0076.patch поставляемый
Просмотреть файл

@ -1,81 +1,31 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 12 Dec 2022 15:47:00 +0000
Subject: Bug 1451394 - Expose mac camera capture backend in .gn and switch it
to gecko libyuv. r=webrtc-reviewers,mjf
Subject: Bug 1451394 - Record video frame captures with PerformanceRecorder in
the new mac camera backend. r=padenot
Differential Revision: https://phabricator.services.mozilla.com/D163682
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b0658888969395dca938597783c8a377b9bea209
Also includes:
Bug 1806605 - Pass TrackingId instead of nsCString to CaptureStage.
Differential Revision: https://phabricator.services.mozilla.com/D163687
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a7362238c9e6fbe0d28200f6b41fc40a0c9a2158
---
BUILD.gn | 4 ++++
sdk/BUILD.gn | 6 ++++++
2 files changed, 10 insertions(+)
modules/video_capture/video_capture.h | 3 +++
1 file changed, 3 insertions(+)
diff --git a/BUILD.gn b/BUILD.gn
index 03880dbb09..873aec3850 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -559,6 +559,10 @@ if (!build_with_chromium) {
]
}
diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
index ad1b341b62..7e181c538e 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -158,6 +158,9 @@ class VideoCaptureModule : public rtc::RefCountInterface {
// Return whether the rotation is applied or left pending.
virtual bool GetApplyRotation() = 0;
+ if (build_with_mozilla && is_mac) {
+ deps += [ "sdk:videocapture_objc" ]
+ }
+ // Mozilla: TrackingId setter for use in profiler markers.
+ virtual void SetTrackingId(uint32_t aTrackingIdProcId) {}
+
if (rtc_enable_protobuf) {
deps += [ "logging:rtc_event_log_proto" ]
}
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index a361656a59..65d1b10124 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -449,6 +449,7 @@ if (is_ios || is_mac) {
]
}
+ if (!build_with_mozilla) {
rtc_library("videosource_objc") {
sources = [
"objc/api/peerconnection/RTCVideoSource+Private.h",
@@ -478,6 +479,7 @@ if (is_ios || is_mac) {
":used_from_extension",
]
}
+ }
rtc_library("videoframebuffer_objc") {
visibility = [ "*" ]
@@ -510,6 +512,7 @@ if (is_ios || is_mac) {
]
}
+ if (!build_with_mozilla) {
rtc_library("opengl_objc") {
sources = [
"objc/components/renderer/opengl/RTCDefaultShader.h",
@@ -662,6 +665,7 @@ if (is_ios || is_mac) {
":videoframebuffer_objc",
]
}
+ }
rtc_library("videocapture_objc") {
visibility = [ "*" ]
@@ -690,6 +694,7 @@ if (is_ios || is_mac) {
]
}
+ if (!build_with_mozilla) {
rtc_library("videocodec_objc") {
visibility = [ "*" ]
configs += [ "..:no_global_constructors" ]
@@ -1747,5 +1752,6 @@ if (is_ios || is_mac) {
"VideoToolbox.framework",
]
}
+ }
}
}
protected:
~VideoCaptureModule() override {}
};
--
2.34.1

353
third_party/libwebrtc/moz-patch-stack/0077.patch поставляемый
Просмотреть файл

@ -1,31 +1,346 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 12 Dec 2022 15:47:00 +0000
Subject: Bug 1451394 - Record video frame captures with PerformanceRecorder in
the new mac camera backend. r=padenot
Date: Tue, 23 Nov 2021 14:11:00 +0000
Subject: Bug 1742181 - libwebrtc: Implement packetsDiscarded bookkeeping for
received video. r=ng
Depends on D131707
Differential Revision: https://phabricator.services.mozilla.com/D131708
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d0196a45a1f449874fc2a759e85e403c45c25575
Also includes:
Bug 1806605 - Pass TrackingId instead of nsCString to CaptureStage.
Differential Revision: https://phabricator.services.mozilla.com/D163687
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a7362238c9e6fbe0d28200f6b41fc40a0c9a2158
Bug 1804288 - (fix-de7ae5755b) reimplement Bug 1742181 - libwebrtc: Implement packetsDiscarded bookkeeping for received video. r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D163959
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/ee566d1bfb654d36e5d58dce637fb0580b989ac1
---
modules/video_capture/video_capture.h | 3 +++
1 file changed, 3 insertions(+)
api/video/frame_buffer.cc | 25 ++++++++++++++++---
api/video/frame_buffer.h | 4 +++
call/video_receive_stream.h | 2 ++
.../include/video_coding_defines.h | 2 ++
modules/video_coding/packet_buffer.cc | 10 +++++---
modules/video_coding/packet_buffer.h | 5 +++-
video/receive_statistics_proxy2.cc | 5 ++++
video/receive_statistics_proxy2.h | 1 +
video/rtp_video_stream_receiver2.cc | 5 +++-
video/rtp_video_stream_receiver2.h | 2 ++
video/video_receive_stream2.cc | 1 +
video/video_stream_buffer_controller.cc | 12 +++++++++
video/video_stream_buffer_controller.h | 3 +++
13 files changed, 69 insertions(+), 8 deletions(-)
diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
index ad1b341b62..7e181c538e 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -158,6 +158,9 @@ class VideoCaptureModule : public rtc::RefCountInterface {
// Return whether the rotation is applied or left pending.
virtual bool GetApplyRotation() = 0;
diff --git a/api/video/frame_buffer.cc b/api/video/frame_buffer.cc
index 4cdf2212a6..8267b8e6cb 100644
--- a/api/video/frame_buffer.cc
+++ b/api/video/frame_buffer.cc
@@ -140,14 +140,29 @@ void FrameBuffer::DropNextDecodableTemporalUnit() {
}
+ // Mozilla: TrackingId setter for use in profiler markers.
+ virtual void SetTrackingId(uint32_t aTrackingIdProcId) {}
auto end_it = std::next(next_decodable_temporal_unit_->last_frame);
- num_dropped_frames_ += std::count_if(
- frames_.begin(), end_it,
- [](const auto& f) { return f.second.encoded_frame != nullptr; });
+
protected:
~VideoCaptureModule() override {}
+ UpdateDroppedFramesAndDiscardedPackets(frames_.begin(), end_it);
frames_.erase(frames_.begin(), end_it);
FindNextAndLastDecodableTemporalUnit();
}
+void FrameBuffer::UpdateDroppedFramesAndDiscardedPackets(FrameIterator begin_it,
+ FrameIterator end_it) {
+ unsigned int num_discarded_packets = 0;
+ unsigned int num_dropped_frames =
+ std::count_if(begin_it, end_it, [&](const auto& f) {
+ if (f.second.encoded_frame) {
+ const auto& packetInfos = f.second.encoded_frame->PacketInfos();
+ num_discarded_packets += packetInfos.size();
+ }
+ return f.second.encoded_frame != nullptr;
+ });
+
+ num_dropped_frames_ += num_dropped_frames;
+ num_discarded_packets_ += num_discarded_packets;
+}
+
absl::optional<int64_t> FrameBuffer::LastContinuousFrameId() const {
return last_continuous_frame_id_;
}
@@ -167,6 +182,9 @@ int FrameBuffer::GetTotalNumberOfContinuousTemporalUnits() const {
int FrameBuffer::GetTotalNumberOfDroppedFrames() const {
return num_dropped_frames_;
}
+int FrameBuffer::GetTotalNumberOfDiscardedPackets() const {
+ return num_discarded_packets_;
+}
size_t FrameBuffer::CurrentSize() const {
return frames_.size();
@@ -269,6 +287,7 @@ void FrameBuffer::FindNextAndLastDecodableTemporalUnit() {
}
void FrameBuffer::Clear() {
+ UpdateDroppedFramesAndDiscardedPackets(frames_.begin(), frames_.end());
frames_.clear();
next_decodable_temporal_unit_.reset();
decodable_temporal_units_info_.reset();
diff --git a/api/video/frame_buffer.h b/api/video/frame_buffer.h
index 94edf64d5a..81fd12da58 100644
--- a/api/video/frame_buffer.h
+++ b/api/video/frame_buffer.h
@@ -66,6 +66,7 @@ class FrameBuffer {
int GetTotalNumberOfContinuousTemporalUnits() const;
int GetTotalNumberOfDroppedFrames() const;
+ int GetTotalNumberOfDiscardedPackets() const;
size_t CurrentSize() const;
private:
@@ -87,6 +88,8 @@ class FrameBuffer {
void PropagateContinuity(const FrameIterator& frame_it);
void FindNextAndLastDecodableTemporalUnit();
void Clear();
+ void UpdateDroppedFramesAndDiscardedPackets(FrameIterator begin_it,
+ FrameIterator end_it);
const bool legacy_frame_id_jump_behavior_;
const size_t max_size_;
@@ -99,6 +102,7 @@ class FrameBuffer {
int num_continuous_temporal_units_ = 0;
int num_dropped_frames_ = 0;
+ int num_discarded_packets_ = 0;
};
} // namespace webrtc
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 25c294a2a6..1ab4a2a85b 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -106,6 +106,8 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
// https://www.w3.org/TR/webrtc-stats/#dom-rtcvideoreceiverstats-framesdropped
uint32_t frames_dropped = 0;
uint32_t frames_decoded = 0;
+ // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsdiscarded
+ uint64_t packets_discarded = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime
TimeDelta total_decode_time = TimeDelta::Zero();
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
diff --git a/modules/video_coding/include/video_coding_defines.h b/modules/video_coding/include/video_coding_defines.h
index 8f70e0298d..bf98d5e668 100644
--- a/modules/video_coding/include/video_coding_defines.h
+++ b/modules/video_coding/include/video_coding_defines.h
@@ -76,6 +76,8 @@ class VCMReceiveStatisticsCallback {
virtual void OnDroppedFrames(uint32_t frames_dropped) = 0;
+ virtual void OnDiscardedPackets(uint32_t packets_discarded) = 0;
+
virtual void OnFrameBufferTimingsUpdated(int max_decode_ms,
int current_delay_ms,
int target_delay_ms,
diff --git a/modules/video_coding/packet_buffer.cc b/modules/video_coding/packet_buffer.cc
index 3dcfc48213..04f02fce97 100644
--- a/modules/video_coding/packet_buffer.cc
+++ b/modules/video_coding/packet_buffer.cc
@@ -115,25 +115,27 @@ PacketBuffer::InsertResult PacketBuffer::InsertPacket(
return result;
}
-void PacketBuffer::ClearTo(uint16_t seq_num) {
+uint32_t PacketBuffer::ClearTo(uint16_t seq_num) {
// We have already cleared past this sequence number, no need to do anything.
if (is_cleared_to_first_seq_num_ &&
AheadOf<uint16_t>(first_seq_num_, seq_num)) {
- return;
+ return 0;
}
// If the packet buffer was cleared between a frame was created and returned.
if (!first_packet_received_)
- return;
+ return 0;
// Avoid iterating over the buffer more than once by capping the number of
// iterations to the `size_` of the buffer.
++seq_num;
+ uint32_t num_cleared_packets = 0;
size_t diff = ForwardDiff<uint16_t>(first_seq_num_, seq_num);
size_t iterations = std::min(diff, buffer_.size());
for (size_t i = 0; i < iterations; ++i) {
auto& stored = buffer_[first_seq_num_ % buffer_.size()];
if (stored != nullptr && AheadOf<uint16_t>(seq_num, stored->seq_num)) {
+ ++num_cleared_packets;
stored = nullptr;
}
++first_seq_num_;
@@ -149,6 +151,8 @@ void PacketBuffer::ClearTo(uint16_t seq_num) {
received_padding_.erase(received_padding_.begin(),
received_padding_.lower_bound(seq_num));
+
+ return num_cleared_packets;
}
void PacketBuffer::Clear() {
diff --git a/modules/video_coding/packet_buffer.h b/modules/video_coding/packet_buffer.h
index 53e08c95a1..47b2ffe199 100644
--- a/modules/video_coding/packet_buffer.h
+++ b/modules/video_coding/packet_buffer.h
@@ -78,7 +78,10 @@ class PacketBuffer {
ABSL_MUST_USE_RESULT InsertResult
InsertPacket(std::unique_ptr<Packet> packet);
ABSL_MUST_USE_RESULT InsertResult InsertPadding(uint16_t seq_num);
- void ClearTo(uint16_t seq_num);
+
+ // Clear all packets older than |seq_num|. Returns the number of packets
+ // cleared.
+ uint32_t ClearTo(uint16_t seq_num);
void Clear();
void ForceSpsPpsIdrIsH264Keyframe();
diff --git a/video/receive_statistics_proxy2.cc b/video/receive_statistics_proxy2.cc
index 4f208a1d5e..020e4bb0ae 100644
--- a/video/receive_statistics_proxy2.cc
+++ b/video/receive_statistics_proxy2.cc
@@ -959,6 +959,11 @@ void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
}));
}
+void ReceiveStatisticsProxy::OnDiscardedPackets(uint32_t packets_discarded) {
+ RTC_DCHECK_RUN_ON(&main_thread_);
+ stats_.packets_discarded += packets_discarded;
+}
+
void ReceiveStatisticsProxy::OnPreDecode(VideoCodecType codec_type, int qp) {
RTC_DCHECK_RUN_ON(&main_thread_);
last_codec_type_ = codec_type;
diff --git a/video/receive_statistics_proxy2.h b/video/receive_statistics_proxy2.h
index 1a2bb77fa6..20139b45e5 100644
--- a/video/receive_statistics_proxy2.h
+++ b/video/receive_statistics_proxy2.h
@@ -90,6 +90,7 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
size_t size_bytes,
VideoContentType content_type) override;
void OnDroppedFrames(uint32_t frames_dropped) override;
+ void OnDiscardedPackets(uint32_t packets_discarded) override;
void OnFrameBufferTimingsUpdated(int max_decode_ms,
int current_delay_ms,
int target_delay_ms,
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 094f8f4a54..46998b6d7c 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -244,6 +244,7 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RtcpCnameCallback* rtcp_cname_callback,
NackPeriodicProcessor* nack_periodic_processor,
+ VCMReceiveStatisticsCallback* vcm_receive_statistics,
OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
@@ -293,6 +294,7 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
&rtcp_feedback_buffer_,
&rtcp_feedback_buffer_,
field_trials_)),
+ vcm_receive_statistics_(vcm_receive_statistics),
packet_buffer_(kPacketBufferStartSize,
PacketBufferMaxSize(field_trials_)),
reference_finder_(std::make_unique<RtpFrameReferenceFinder>()),
@@ -1219,7 +1221,8 @@ void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
int64_t unwrapped_rtp_seq_num = rtp_seq_num_unwrapper_.Unwrap(seq_num);
packet_infos_.erase(packet_infos_.begin(),
packet_infos_.upper_bound(unwrapped_rtp_seq_num));
- packet_buffer_.ClearTo(seq_num);
+ uint32_t num_packets_cleared = packet_buffer_.ClearTo(seq_num);
+ vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
reference_finder_->ClearTo(seq_num);
}
}
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 6bf4bf8453..931525a054 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -91,6 +91,7 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RtcpCnameCallback* rtcp_cname_callback,
NackPeriodicProcessor* nack_periodic_processor,
+ VCMReceiveStatisticsCallback* vcm_receive_statistics,
// The KeyFrameRequestSender is optional; if not provided, key frame
// requests are sent via the internal RtpRtcp module.
OnCompleteFrameCallback* complete_frame_callback,
@@ -368,6 +369,7 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
std::unique_ptr<LossNotificationController> loss_notification_controller_
RTC_GUARDED_BY(packet_sequence_checker_);
+ VCMReceiveStatisticsCallback* const vcm_receive_statistics_;
video_coding::PacketBuffer packet_buffer_
RTC_GUARDED_BY(packet_sequence_checker_);
UniqueTimestampCounter frame_counter_
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 7cbd49d322..beb894e139 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -211,6 +211,7 @@ VideoReceiveStream2::VideoReceiveStream2(
&stats_proxy_,
&stats_proxy_,
nack_periodic_processor,
+ &stats_proxy_,
this, // OnCompleteFrameCallback
std::move(config_.frame_decryptor),
std::move(config_.frame_transformer),
diff --git a/video/video_stream_buffer_controller.cc b/video/video_stream_buffer_controller.cc
index f7d3acdaf6..7e44eff39a 100644
--- a/video/video_stream_buffer_controller.cc
+++ b/video/video_stream_buffer_controller.cc
@@ -247,6 +247,7 @@ void VideoStreamBufferController::OnFrameReady(
// Update stats.
UpdateDroppedFrames();
+ UpdateDiscardedPackets();
UpdateJitterDelay();
UpdateTimingFrameInfo();
@@ -312,6 +313,17 @@ void VideoStreamBufferController::UpdateDroppedFrames()
buffer_->GetTotalNumberOfDroppedFrames();
}
+void VideoStreamBufferController::UpdateDiscardedPackets()
+ RTC_RUN_ON(&worker_sequence_checker_) {
+ const int discarded_packets = buffer_->GetTotalNumberOfDiscardedPackets() -
+ packets_discarded_before_last_new_frame_;
+ if (discarded_packets > 0) {
+ stats_proxy_->OnDiscardedPackets(discarded_packets);
+ }
+ packets_discarded_before_last_new_frame_ =
+ buffer_->GetTotalNumberOfDiscardedPackets();
+}
+
void VideoStreamBufferController::UpdateJitterDelay() {
auto timings = timing_->GetTimings();
if (timings.num_decoded_frames) {
diff --git a/video/video_stream_buffer_controller.h b/video/video_stream_buffer_controller.h
index ed79b0fa1f..7638c91471 100644
--- a/video/video_stream_buffer_controller.h
+++ b/video/video_stream_buffer_controller.h
@@ -67,6 +67,7 @@ class VideoStreamBufferController {
void OnTimeout(TimeDelta delay);
void FrameReadyForDecode(uint32_t rtp_timestamp, Timestamp render_time);
void UpdateDroppedFrames() RTC_RUN_ON(&worker_sequence_checker_);
+ void UpdateDiscardedPackets() RTC_RUN_ON(&worker_sequence_checker_);
void UpdateJitterDelay();
void UpdateTimingFrameInfo();
bool IsTooManyFramesQueued() const RTC_RUN_ON(&worker_sequence_checker_);
@@ -94,6 +95,8 @@ class VideoStreamBufferController {
RTC_GUARDED_BY(&worker_sequence_checker_);
int frames_dropped_before_last_new_frame_
RTC_GUARDED_BY(&worker_sequence_checker_) = 0;
+ int packets_discarded_before_last_new_frame_
+ RTC_GUARDED_BY(&worker_sequence_checker_) = 0;
VCMVideoProtection protection_mode_
RTC_GUARDED_BY(&worker_sequence_checker_) = kProtectionNack;
--
2.34.1

353
third_party/libwebrtc/moz-patch-stack/0078.patch поставляемый
Просмотреть файл

@ -1,346 +1,49 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Tue, 23 Nov 2021 14:11:00 +0000
Subject: Bug 1742181 - libwebrtc: Implement packetsDiscarded bookkeeping for
received video. r=ng
Date: Thu, 6 Jan 2022 00:16:00 +0000
Subject: Bug 1748478 - Propagate calculated discarded packets to stats. r=bwc
Depends on D131707
Differential Revision: https://phabricator.services.mozilla.com/D131708
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d0196a45a1f449874fc2a759e85e403c45c25575
Also includes:
Bug 1804288 - (fix-de7ae5755b) reimplement Bug 1742181 - libwebrtc: Implement packetsDiscarded bookkeeping for received video. r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D163959
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/ee566d1bfb654d36e5d58dce637fb0580b989ac1
Differential Revision: https://phabricator.services.mozilla.com/D135061
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/56fbf0469e25fa0d589c51ca112ce534a7c0ab91
---
api/video/frame_buffer.cc | 25 ++++++++++++++++---
api/video/frame_buffer.h | 4 +++
call/video_receive_stream.h | 2 ++
.../include/video_coding_defines.h | 2 ++
modules/video_coding/packet_buffer.cc | 10 +++++---
modules/video_coding/packet_buffer.h | 5 +++-
video/receive_statistics_proxy2.cc | 5 ++++
video/receive_statistics_proxy2.h | 1 +
video/rtp_video_stream_receiver2.cc | 5 +++-
video/rtp_video_stream_receiver2.h | 2 ++
video/video_receive_stream2.cc | 1 +
video/video_stream_buffer_controller.cc | 12 +++++++++
video/video_stream_buffer_controller.h | 3 +++
13 files changed, 69 insertions(+), 8 deletions(-)
video/receive_statistics_proxy2.cc | 9 +++++++--
video/rtp_video_stream_receiver2.cc | 4 +++-
2 files changed, 10 insertions(+), 3 deletions(-)
diff --git a/api/video/frame_buffer.cc b/api/video/frame_buffer.cc
index 4cdf2212a6..8267b8e6cb 100644
--- a/api/video/frame_buffer.cc
+++ b/api/video/frame_buffer.cc
@@ -140,14 +140,29 @@ void FrameBuffer::DropNextDecodableTemporalUnit() {
}
auto end_it = std::next(next_decodable_temporal_unit_->last_frame);
- num_dropped_frames_ += std::count_if(
- frames_.begin(), end_it,
- [](const auto& f) { return f.second.encoded_frame != nullptr; });
+
+ UpdateDroppedFramesAndDiscardedPackets(frames_.begin(), end_it);
frames_.erase(frames_.begin(), end_it);
FindNextAndLastDecodableTemporalUnit();
}
+void FrameBuffer::UpdateDroppedFramesAndDiscardedPackets(FrameIterator begin_it,
+ FrameIterator end_it) {
+ unsigned int num_discarded_packets = 0;
+ unsigned int num_dropped_frames =
+ std::count_if(begin_it, end_it, [&](const auto& f) {
+ if (f.second.encoded_frame) {
+ const auto& packetInfos = f.second.encoded_frame->PacketInfos();
+ num_discarded_packets += packetInfos.size();
+ }
+ return f.second.encoded_frame != nullptr;
+ });
+
+ num_dropped_frames_ += num_dropped_frames;
+ num_discarded_packets_ += num_discarded_packets;
+}
+
absl::optional<int64_t> FrameBuffer::LastContinuousFrameId() const {
return last_continuous_frame_id_;
}
@@ -167,6 +182,9 @@ int FrameBuffer::GetTotalNumberOfContinuousTemporalUnits() const {
int FrameBuffer::GetTotalNumberOfDroppedFrames() const {
return num_dropped_frames_;
}
+int FrameBuffer::GetTotalNumberOfDiscardedPackets() const {
+ return num_discarded_packets_;
+}
size_t FrameBuffer::CurrentSize() const {
return frames_.size();
@@ -269,6 +287,7 @@ void FrameBuffer::FindNextAndLastDecodableTemporalUnit() {
}
void FrameBuffer::Clear() {
+ UpdateDroppedFramesAndDiscardedPackets(frames_.begin(), frames_.end());
frames_.clear();
next_decodable_temporal_unit_.reset();
decodable_temporal_units_info_.reset();
diff --git a/api/video/frame_buffer.h b/api/video/frame_buffer.h
index 94edf64d5a..81fd12da58 100644
--- a/api/video/frame_buffer.h
+++ b/api/video/frame_buffer.h
@@ -66,6 +66,7 @@ class FrameBuffer {
int GetTotalNumberOfContinuousTemporalUnits() const;
int GetTotalNumberOfDroppedFrames() const;
+ int GetTotalNumberOfDiscardedPackets() const;
size_t CurrentSize() const;
private:
@@ -87,6 +88,8 @@ class FrameBuffer {
void PropagateContinuity(const FrameIterator& frame_it);
void FindNextAndLastDecodableTemporalUnit();
void Clear();
+ void UpdateDroppedFramesAndDiscardedPackets(FrameIterator begin_it,
+ FrameIterator end_it);
const bool legacy_frame_id_jump_behavior_;
const size_t max_size_;
@@ -99,6 +102,7 @@ class FrameBuffer {
int num_continuous_temporal_units_ = 0;
int num_dropped_frames_ = 0;
+ int num_discarded_packets_ = 0;
};
} // namespace webrtc
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 25c294a2a6..1ab4a2a85b 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -106,6 +106,8 @@ class VideoReceiveStreamInterface : public MediaReceiveStreamInterface {
// https://www.w3.org/TR/webrtc-stats/#dom-rtcvideoreceiverstats-framesdropped
uint32_t frames_dropped = 0;
uint32_t frames_decoded = 0;
+ // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetsdiscarded
+ uint64_t packets_discarded = 0;
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime
TimeDelta total_decode_time = TimeDelta::Zero();
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
diff --git a/modules/video_coding/include/video_coding_defines.h b/modules/video_coding/include/video_coding_defines.h
index 8f70e0298d..bf98d5e668 100644
--- a/modules/video_coding/include/video_coding_defines.h
+++ b/modules/video_coding/include/video_coding_defines.h
@@ -76,6 +76,8 @@ class VCMReceiveStatisticsCallback {
virtual void OnDroppedFrames(uint32_t frames_dropped) = 0;
+ virtual void OnDiscardedPackets(uint32_t packets_discarded) = 0;
+
virtual void OnFrameBufferTimingsUpdated(int max_decode_ms,
int current_delay_ms,
int target_delay_ms,
diff --git a/modules/video_coding/packet_buffer.cc b/modules/video_coding/packet_buffer.cc
index 3dcfc48213..04f02fce97 100644
--- a/modules/video_coding/packet_buffer.cc
+++ b/modules/video_coding/packet_buffer.cc
@@ -115,25 +115,27 @@ PacketBuffer::InsertResult PacketBuffer::InsertPacket(
return result;
}
-void PacketBuffer::ClearTo(uint16_t seq_num) {
+uint32_t PacketBuffer::ClearTo(uint16_t seq_num) {
// We have already cleared past this sequence number, no need to do anything.
if (is_cleared_to_first_seq_num_ &&
AheadOf<uint16_t>(first_seq_num_, seq_num)) {
- return;
+ return 0;
}
// If the packet buffer was cleared between a frame was created and returned.
if (!first_packet_received_)
- return;
+ return 0;
// Avoid iterating over the buffer more than once by capping the number of
// iterations to the `size_` of the buffer.
++seq_num;
+ uint32_t num_cleared_packets = 0;
size_t diff = ForwardDiff<uint16_t>(first_seq_num_, seq_num);
size_t iterations = std::min(diff, buffer_.size());
for (size_t i = 0; i < iterations; ++i) {
auto& stored = buffer_[first_seq_num_ % buffer_.size()];
if (stored != nullptr && AheadOf<uint16_t>(seq_num, stored->seq_num)) {
+ ++num_cleared_packets;
stored = nullptr;
}
++first_seq_num_;
@@ -149,6 +151,8 @@ void PacketBuffer::ClearTo(uint16_t seq_num) {
received_padding_.erase(received_padding_.begin(),
received_padding_.lower_bound(seq_num));
+
+ return num_cleared_packets;
}
void PacketBuffer::Clear() {
diff --git a/modules/video_coding/packet_buffer.h b/modules/video_coding/packet_buffer.h
index 53e08c95a1..47b2ffe199 100644
--- a/modules/video_coding/packet_buffer.h
+++ b/modules/video_coding/packet_buffer.h
@@ -78,7 +78,10 @@ class PacketBuffer {
ABSL_MUST_USE_RESULT InsertResult
InsertPacket(std::unique_ptr<Packet> packet);
ABSL_MUST_USE_RESULT InsertResult InsertPadding(uint16_t seq_num);
- void ClearTo(uint16_t seq_num);
+
+ // Clear all packets older than |seq_num|. Returns the number of packets
+ // cleared.
+ uint32_t ClearTo(uint16_t seq_num);
void Clear();
void ForceSpsPpsIdrIsH264Keyframe();
diff --git a/video/receive_statistics_proxy2.cc b/video/receive_statistics_proxy2.cc
index 4f208a1d5e..020e4bb0ae 100644
index 020e4bb0ae..f5011c46ef 100644
--- a/video/receive_statistics_proxy2.cc
+++ b/video/receive_statistics_proxy2.cc
@@ -959,6 +959,11 @@ void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
}));
@@ -960,8 +960,13 @@ void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
}
void ReceiveStatisticsProxy::OnDiscardedPackets(uint32_t packets_discarded) {
- RTC_DCHECK_RUN_ON(&main_thread_);
- stats_.packets_discarded += packets_discarded;
+ // Can be called on either the decode queue or the worker thread
+ // See FrameBuffer2 for more details.
+ worker_thread_->PostTask(
+ SafeTask(task_safety_.flag(), [packets_discarded, this]() {
+ RTC_DCHECK_RUN_ON(&main_thread_);
+ stats_.packets_discarded += packets_discarded;
+ }));
}
+void ReceiveStatisticsProxy::OnDiscardedPackets(uint32_t packets_discarded) {
+ RTC_DCHECK_RUN_ON(&main_thread_);
+ stats_.packets_discarded += packets_discarded;
+}
+
void ReceiveStatisticsProxy::OnPreDecode(VideoCodecType codec_type, int qp) {
RTC_DCHECK_RUN_ON(&main_thread_);
last_codec_type_ = codec_type;
diff --git a/video/receive_statistics_proxy2.h b/video/receive_statistics_proxy2.h
index 1a2bb77fa6..20139b45e5 100644
--- a/video/receive_statistics_proxy2.h
+++ b/video/receive_statistics_proxy2.h
@@ -90,6 +90,7 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
size_t size_bytes,
VideoContentType content_type) override;
void OnDroppedFrames(uint32_t frames_dropped) override;
+ void OnDiscardedPackets(uint32_t packets_discarded) override;
void OnFrameBufferTimingsUpdated(int max_decode_ms,
int current_delay_ms,
int target_delay_ms,
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 094f8f4a54..46998b6d7c 100644
index 46998b6d7c..a5d5f637e5 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -244,6 +244,7 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RtcpCnameCallback* rtcp_cname_callback,
NackPeriodicProcessor* nack_periodic_processor,
+ VCMReceiveStatisticsCallback* vcm_receive_statistics,
OnCompleteFrameCallback* complete_frame_callback,
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
@@ -293,6 +294,7 @@ RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
&rtcp_feedback_buffer_,
&rtcp_feedback_buffer_,
field_trials_)),
+ vcm_receive_statistics_(vcm_receive_statistics),
packet_buffer_(kPacketBufferStartSize,
PacketBufferMaxSize(field_trials_)),
reference_finder_(std::make_unique<RtpFrameReferenceFinder>()),
@@ -1219,7 +1221,8 @@ void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
int64_t unwrapped_rtp_seq_num = rtp_seq_num_unwrapper_.Unwrap(seq_num);
@@ -1222,7 +1222,9 @@ void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
packet_infos_.erase(packet_infos_.begin(),
packet_infos_.upper_bound(unwrapped_rtp_seq_num));
- packet_buffer_.ClearTo(seq_num);
+ uint32_t num_packets_cleared = packet_buffer_.ClearTo(seq_num);
+ vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
uint32_t num_packets_cleared = packet_buffer_.ClearTo(seq_num);
- vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
+ if (num_packets_cleared > 0) {
+ vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
+ }
reference_finder_->ClearTo(seq_num);
}
}
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
index 6bf4bf8453..931525a054 100644
--- a/video/rtp_video_stream_receiver2.h
+++ b/video/rtp_video_stream_receiver2.h
@@ -91,6 +91,7 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
RtcpCnameCallback* rtcp_cname_callback,
NackPeriodicProcessor* nack_periodic_processor,
+ VCMReceiveStatisticsCallback* vcm_receive_statistics,
// The KeyFrameRequestSender is optional; if not provided, key frame
// requests are sent via the internal RtpRtcp module.
OnCompleteFrameCallback* complete_frame_callback,
@@ -368,6 +369,7 @@ class RtpVideoStreamReceiver2 : public LossNotificationSender,
std::unique_ptr<LossNotificationController> loss_notification_controller_
RTC_GUARDED_BY(packet_sequence_checker_);
+ VCMReceiveStatisticsCallback* const vcm_receive_statistics_;
video_coding::PacketBuffer packet_buffer_
RTC_GUARDED_BY(packet_sequence_checker_);
UniqueTimestampCounter frame_counter_
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index 7cbd49d322..beb894e139 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -211,6 +211,7 @@ VideoReceiveStream2::VideoReceiveStream2(
&stats_proxy_,
&stats_proxy_,
nack_periodic_processor,
+ &stats_proxy_,
this, // OnCompleteFrameCallback
std::move(config_.frame_decryptor),
std::move(config_.frame_transformer),
diff --git a/video/video_stream_buffer_controller.cc b/video/video_stream_buffer_controller.cc
index f7d3acdaf6..7e44eff39a 100644
--- a/video/video_stream_buffer_controller.cc
+++ b/video/video_stream_buffer_controller.cc
@@ -247,6 +247,7 @@ void VideoStreamBufferController::OnFrameReady(
// Update stats.
UpdateDroppedFrames();
+ UpdateDiscardedPackets();
UpdateJitterDelay();
UpdateTimingFrameInfo();
@@ -312,6 +313,17 @@ void VideoStreamBufferController::UpdateDroppedFrames()
buffer_->GetTotalNumberOfDroppedFrames();
}
+void VideoStreamBufferController::UpdateDiscardedPackets()
+ RTC_RUN_ON(&worker_sequence_checker_) {
+ const int discarded_packets = buffer_->GetTotalNumberOfDiscardedPackets() -
+ packets_discarded_before_last_new_frame_;
+ if (discarded_packets > 0) {
+ stats_proxy_->OnDiscardedPackets(discarded_packets);
+ }
+ packets_discarded_before_last_new_frame_ =
+ buffer_->GetTotalNumberOfDiscardedPackets();
+}
+
void VideoStreamBufferController::UpdateJitterDelay() {
auto timings = timing_->GetTimings();
if (timings.num_decoded_frames) {
diff --git a/video/video_stream_buffer_controller.h b/video/video_stream_buffer_controller.h
index ed79b0fa1f..7638c91471 100644
--- a/video/video_stream_buffer_controller.h
+++ b/video/video_stream_buffer_controller.h
@@ -67,6 +67,7 @@ class VideoStreamBufferController {
void OnTimeout(TimeDelta delay);
void FrameReadyForDecode(uint32_t rtp_timestamp, Timestamp render_time);
void UpdateDroppedFrames() RTC_RUN_ON(&worker_sequence_checker_);
+ void UpdateDiscardedPackets() RTC_RUN_ON(&worker_sequence_checker_);
void UpdateJitterDelay();
void UpdateTimingFrameInfo();
bool IsTooManyFramesQueued() const RTC_RUN_ON(&worker_sequence_checker_);
@@ -94,6 +95,8 @@ class VideoStreamBufferController {
RTC_GUARDED_BY(&worker_sequence_checker_);
int frames_dropped_before_last_new_frame_
RTC_GUARDED_BY(&worker_sequence_checker_) = 0;
+ int packets_discarded_before_last_new_frame_
+ RTC_GUARDED_BY(&worker_sequence_checker_) = 0;
VCMVideoProtection protection_mode_
RTC_GUARDED_BY(&worker_sequence_checker_) = kProtectionNack;
--
2.34.1

302
third_party/libwebrtc/moz-patch-stack/0079.patch поставляемый
Просмотреть файл

@ -1,49 +1,295 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Thu, 6 Jan 2022 00:16:00 +0000
Subject: Bug 1748478 - Propagate calculated discarded packets to stats. r=bwc
Subject: Bug 1748458 - Add TRACE_EVENTs for dropped frames and packets for
received video. r=bwc
Differential Revision: https://phabricator.services.mozilla.com/D135061
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/56fbf0469e25fa0d589c51ca112ce534a7c0ab91
This lets us see in the profiler how many received frames and packets we decide
to drop and the reasons why.
Differential Revision: https://phabricator.services.mozilla.com/D135062
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/08e252da94c4752eccfd845eef13d8517953cc6a
Also includes:
Bug 1804288 - (fix-de7ae5755b) reimplement Bug 1748458 - Add TRACE_EVENTs for dropped frames and packets for received video. r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D163960
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8e9a326a99cd5eaa6e447ff57c01ad9d79a09744
---
video/receive_statistics_proxy2.cc | 9 +++++++--
video/rtp_video_stream_receiver2.cc | 4 +++-
2 files changed, 10 insertions(+), 3 deletions(-)
api/video/frame_buffer.cc | 33 +++++++++++++++++++++++++
modules/video_coding/frame_buffer2.cc | 25 +++++++++++++++++++
video/receive_statistics_proxy2.cc | 11 +++++++++
video/rtp_video_stream_receiver2.cc | 4 +++
video/video_stream_buffer_controller.cc | 7 ++++++
5 files changed, 80 insertions(+)
diff --git a/api/video/frame_buffer.cc b/api/video/frame_buffer.cc
index 8267b8e6cb..f5d93f5f76 100644
--- a/api/video/frame_buffer.cc
+++ b/api/video/frame_buffer.cc
@@ -16,6 +16,7 @@
#include "absl/container/inlined_vector.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
@@ -68,7 +69,12 @@ FrameBuffer::FrameBuffer(int max_size,
decoded_frame_history_(max_decode_history) {}
bool FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
+ const uint32_t ssrc =
+ frame->PacketInfos().empty() ? 0 : frame->PacketInfos()[0].ssrc();
if (!ValidReferences(*frame)) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Invalid references)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING) << "Frame " << frame->Id()
<< " has invalid references, dropping frame.";
return false;
@@ -78,23 +84,35 @@ bool FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
if (legacy_frame_id_jump_behavior_ && frame->is_keyframe() &&
AheadOf(frame->Timestamp(),
*decoded_frame_history_.GetLastDecodedFrameTimestamp())) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (OOO + PicId jump)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING)
<< "Keyframe " << frame->Id()
<< " has newer timestamp but older picture id, clearing buffer.";
Clear();
} else {
// Already decoded past this frame.
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Out of order)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
return false;
}
}
if (frames_.size() == max_size_) {
if (frame->is_keyframe()) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (KF + Full buffer)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING) << "Keyframe " << frame->Id()
<< " inserted into full buffer, clearing buffer.";
Clear();
} else {
// No space for this frame.
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Full buffer)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
return false;
}
}
@@ -149,16 +167,31 @@ void FrameBuffer::DropNextDecodableTemporalUnit() {
void FrameBuffer::UpdateDroppedFramesAndDiscardedPackets(FrameIterator begin_it,
FrameIterator end_it) {
+ uint32_t dropped_ssrc = 0;
+ int64_t dropped_frame_id = 0;
unsigned int num_discarded_packets = 0;
unsigned int num_dropped_frames =
std::count_if(begin_it, end_it, [&](const auto& f) {
if (f.second.encoded_frame) {
const auto& packetInfos = f.second.encoded_frame->PacketInfos();
+ dropped_frame_id = f.first;
+ if (!packetInfos.empty()) {
+ dropped_ssrc = packetInfos[0].ssrc();
+ }
num_discarded_packets += packetInfos.size();
}
return f.second.encoded_frame != nullptr;
});
+ if (num_dropped_frames > 0) {
+ TRACE_EVENT2("webrtc", "FrameBuffer Dropping Old Frames", "remote_ssrc",
+ dropped_ssrc, "frame_id", dropped_frame_id);
+ }
+ if (num_discarded_packets > 0) {
+ TRACE_EVENT2("webrtc", "FrameBuffer Discarding Old Packets", "remote_ssrc",
+ dropped_ssrc, "frame_id", dropped_frame_id);
+ }
+
num_dropped_frames_ += num_dropped_frames;
num_discarded_packets_ += num_discarded_packets;
}
diff --git a/modules/video_coding/frame_buffer2.cc b/modules/video_coding/frame_buffer2.cc
index b289663eec..a70b143a29 100644
--- a/modules/video_coding/frame_buffer2.cc
+++ b/modules/video_coding/frame_buffer2.cc
@@ -371,9 +371,14 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
MutexLock lock(&mutex_);
+ const auto& pis = frame->PacketInfos();
int64_t last_continuous_frame_id = last_continuous_frame_.value_or(-1);
if (!ValidReferences(*frame)) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Invalid references)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "Frame " << frame->Id()
<< " has invalid frame references, dropping frame.";
return last_continuous_frame_id;
@@ -381,11 +386,19 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
if (frames_.size() >= kMaxFramesBuffered) {
if (frame->is_keyframe()) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (KF + Full buffer)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "Inserting keyframe " << frame->Id()
<< " but buffer is full, clearing"
" buffer and inserting the frame.";
ClearFramesAndHistory();
} else {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Full buffer)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "Frame " << frame->Id()
<< " could not be inserted due to the frame "
"buffer being full, dropping frame.";
@@ -404,11 +417,19 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
// reconfiguration or some other reason. Even though this is not according
// to spec we can still continue to decode from this frame if it is a
// keyframe.
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (OOO + PicId jump)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING)
<< "A jump in frame id was detected, clearing buffer.";
ClearFramesAndHistory();
last_continuous_frame_id = -1;
} else {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Out of order)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "Frame " << frame->Id() << " inserted after frame "
<< *last_decoded_frame
<< " was handed off for decoding, dropping frame.";
@@ -421,6 +442,10 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
// when the frame id make large jumps mid stream.
if (!frames_.empty() && frame->Id() < frames_.begin()->first &&
frames_.rbegin()->first < frame->Id()) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (PicId big-jump)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "A jump in frame id was detected, clearing buffer.";
ClearFramesAndHistory();
last_continuous_frame_id = -1;
diff --git a/video/receive_statistics_proxy2.cc b/video/receive_statistics_proxy2.cc
index 020e4bb0ae..f5011c46ef 100644
index f5011c46ef..508c36eaaf 100644
--- a/video/receive_statistics_proxy2.cc
+++ b/video/receive_statistics_proxy2.cc
@@ -960,8 +960,13 @@ void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
}
@@ -20,6 +20,7 @@
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
#include "video/video_receive_stream2.h"
@@ -921,6 +922,9 @@ void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
VideoContentType content_type) {
RTC_DCHECK_RUN_ON(&main_thread_);
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnCompleteFrame",
+ "remote_ssrc", remote_ssrc_, "is_keyframe", is_keyframe);
+
if (is_keyframe) {
++stats_.frame_counts.key_frames;
} else {
@@ -952,6 +956,8 @@ void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
// Can be called on either the decode queue or the worker thread
// See FrameBuffer2 for more details.
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnDroppedFrames",
+ "remote_ssrc", remote_ssrc_, "frames_dropped", frames_dropped);
worker_thread_->PostTask(
SafeTask(task_safety_.flag(), [frames_dropped, this]() {
RTC_DCHECK_RUN_ON(&main_thread_);
@@ -962,6 +968,9 @@ void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
void ReceiveStatisticsProxy::OnDiscardedPackets(uint32_t packets_discarded) {
- RTC_DCHECK_RUN_ON(&main_thread_);
- stats_.packets_discarded += packets_discarded;
+ // Can be called on either the decode queue or the worker thread
+ // See FrameBuffer2 for more details.
+ worker_thread_->PostTask(
+ SafeTask(task_safety_.flag(), [packets_discarded, this]() {
+ RTC_DCHECK_RUN_ON(&main_thread_);
+ stats_.packets_discarded += packets_discarded;
+ }));
// Can be called on either the decode queue or the worker thread
// See FrameBuffer2 for more details.
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnDiscardedPackets",
+ "remote_ssrc", remote_ssrc_, "packets_discarded",
+ packets_discarded);
worker_thread_->PostTask(
SafeTask(task_safety_.flag(), [packets_discarded, this]() {
RTC_DCHECK_RUN_ON(&main_thread_);
@@ -991,6 +1000,8 @@ void ReceiveStatisticsProxy::OnStreamInactive() {
void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms) {
RTC_DCHECK_RUN_ON(&main_thread_);
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnRttUpdate",
+ "remote_ssrc", remote_ssrc_, "avg_rtt_ms", avg_rtt_ms);
avg_rtt_ms_ = avg_rtt_ms;
}
void ReceiveStatisticsProxy::OnPreDecode(VideoCodecType codec_type, int qp) {
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 46998b6d7c..a5d5f637e5 100644
index a5d5f637e5..eb0a7787ac 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -1222,7 +1222,9 @@ void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
packet_infos_.erase(packet_infos_.begin(),
@@ -44,6 +44,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/ntp_time.h"
@@ -1223,6 +1224,9 @@ void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
packet_infos_.upper_bound(unwrapped_rtp_seq_num));
uint32_t num_packets_cleared = packet_buffer_.ClearTo(seq_num);
- vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
+ if (num_packets_cleared > 0) {
+ vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
+ }
if (num_packets_cleared > 0) {
+ TRACE_EVENT2("webrtc",
+ "RtpVideoStreamReceiver2::FrameDecoded Cleared Old Packets",
+ "remote_ssrc", config_.rtp.remote_ssrc, "seq_num", seq_num);
vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
}
reference_finder_->ClearTo(seq_num);
}
}
diff --git a/video/video_stream_buffer_controller.cc b/video/video_stream_buffer_controller.cc
index 7e44eff39a..37724a8338 100644
--- a/video/video_stream_buffer_controller.cc
+++ b/video/video_stream_buffer_controller.cc
@@ -28,6 +28,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/thread_annotations.h"
+#include "rtc_base/trace_event.h"
#include "video/frame_decode_scheduler.h"
#include "video/frame_decode_timing.h"
#include "video/task_queue_frame_decode_scheduler.h"
@@ -139,6 +140,9 @@ absl::optional<int64_t> VideoStreamBufferController::InsertFrame(
std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
FrameMetadata metadata(*frame);
+ const uint32_t ssrc =
+ frame->PacketInfos().empty() ? 0 : frame->PacketInfos()[0].ssrc();
+ const int64_t frameId = frame->Id();
int complete_units = buffer_->GetTotalNumberOfContinuousTemporalUnits();
if (buffer_->InsertFrame(std::move(frame))) {
RTC_DCHECK(metadata.receive_time) << "Frame receive time must be set!";
@@ -149,6 +153,9 @@ absl::optional<int64_t> VideoStreamBufferController::InsertFrame(
*metadata.receive_time);
}
if (complete_units < buffer_->GetTotalNumberOfContinuousTemporalUnits()) {
+ TRACE_EVENT2("webrtc",
+ "VideoStreamBufferController::InsertFrame Frame Complete",
+ "remote_ssrc", ssrc, "frame_id", frameId);
stats_proxy_->OnCompleteFrame(metadata.is_keyframe, metadata.size,
metadata.contentType);
MaybeScheduleFrameForRelease();
--
2.34.1

301
third_party/libwebrtc/moz-patch-stack/0080.patch поставляемый
Просмотреть файл

@ -1,295 +1,30 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Thu, 6 Jan 2022 00:16:00 +0000
Subject: Bug 1748458 - Add TRACE_EVENTs for dropped frames and packets for
received video. r=bwc
Subject: Bug 1748458 - Add TRACE_EVENT for keyframe requests. r=bwc
This lets us see in the profiler how many received frames and packets we decide
to drop and the reasons why.
Differential Revision: https://phabricator.services.mozilla.com/D135062
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/08e252da94c4752eccfd845eef13d8517953cc6a
Also includes:
Bug 1804288 - (fix-de7ae5755b) reimplement Bug 1748458 - Add TRACE_EVENTs for dropped frames and packets for received video. r=pehrsons
Differential Revision: https://phabricator.services.mozilla.com/D163960
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8e9a326a99cd5eaa6e447ff57c01ad9d79a09744
Differential Revision: https://phabricator.services.mozilla.com/D135113
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/5b2a7894ef1cf096d0e8977754507c0820e757fc
---
api/video/frame_buffer.cc | 33 +++++++++++++++++++++++++
modules/video_coding/frame_buffer2.cc | 25 +++++++++++++++++++
video/receive_statistics_proxy2.cc | 11 +++++++++
video/rtp_video_stream_receiver2.cc | 4 +++
video/video_stream_buffer_controller.cc | 7 ++++++
5 files changed, 80 insertions(+)
video/rtp_video_stream_receiver2.cc | 6 ++++++
1 file changed, 6 insertions(+)
diff --git a/api/video/frame_buffer.cc b/api/video/frame_buffer.cc
index 8267b8e6cb..f5d93f5f76 100644
--- a/api/video/frame_buffer.cc
+++ b/api/video/frame_buffer.cc
@@ -16,6 +16,7 @@
#include "absl/container/inlined_vector.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/trace_event.h"
namespace webrtc {
namespace {
@@ -68,7 +69,12 @@ FrameBuffer::FrameBuffer(int max_size,
decoded_frame_history_(max_decode_history) {}
bool FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
+ const uint32_t ssrc =
+ frame->PacketInfos().empty() ? 0 : frame->PacketInfos()[0].ssrc();
if (!ValidReferences(*frame)) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Invalid references)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING) << "Frame " << frame->Id()
<< " has invalid references, dropping frame.";
return false;
@@ -78,23 +84,35 @@ bool FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
if (legacy_frame_id_jump_behavior_ && frame->is_keyframe() &&
AheadOf(frame->Timestamp(),
*decoded_frame_history_.GetLastDecodedFrameTimestamp())) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (OOO + PicId jump)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING)
<< "Keyframe " << frame->Id()
<< " has newer timestamp but older picture id, clearing buffer.";
Clear();
} else {
// Already decoded past this frame.
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Out of order)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
return false;
}
}
if (frames_.size() == max_size_) {
if (frame->is_keyframe()) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (KF + Full buffer)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
RTC_DLOG(LS_WARNING) << "Keyframe " << frame->Id()
<< " inserted into full buffer, clearing buffer.";
Clear();
} else {
// No space for this frame.
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Full buffer)",
+ "remote_ssrc", ssrc, "frame_id", frame->Id());
return false;
}
}
@@ -149,16 +167,31 @@ void FrameBuffer::DropNextDecodableTemporalUnit() {
void FrameBuffer::UpdateDroppedFramesAndDiscardedPackets(FrameIterator begin_it,
FrameIterator end_it) {
+ uint32_t dropped_ssrc = 0;
+ int64_t dropped_frame_id = 0;
unsigned int num_discarded_packets = 0;
unsigned int num_dropped_frames =
std::count_if(begin_it, end_it, [&](const auto& f) {
if (f.second.encoded_frame) {
const auto& packetInfos = f.second.encoded_frame->PacketInfos();
+ dropped_frame_id = f.first;
+ if (!packetInfos.empty()) {
+ dropped_ssrc = packetInfos[0].ssrc();
+ }
num_discarded_packets += packetInfos.size();
}
return f.second.encoded_frame != nullptr;
});
+ if (num_dropped_frames > 0) {
+ TRACE_EVENT2("webrtc", "FrameBuffer Dropping Old Frames", "remote_ssrc",
+ dropped_ssrc, "frame_id", dropped_frame_id);
+ }
+ if (num_discarded_packets > 0) {
+ TRACE_EVENT2("webrtc", "FrameBuffer Discarding Old Packets", "remote_ssrc",
+ dropped_ssrc, "frame_id", dropped_frame_id);
+ }
+
num_dropped_frames_ += num_dropped_frames;
num_discarded_packets_ += num_discarded_packets;
}
diff --git a/modules/video_coding/frame_buffer2.cc b/modules/video_coding/frame_buffer2.cc
index b289663eec..a70b143a29 100644
--- a/modules/video_coding/frame_buffer2.cc
+++ b/modules/video_coding/frame_buffer2.cc
@@ -371,9 +371,14 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
MutexLock lock(&mutex_);
+ const auto& pis = frame->PacketInfos();
int64_t last_continuous_frame_id = last_continuous_frame_.value_or(-1);
if (!ValidReferences(*frame)) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Invalid references)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "Frame " << frame->Id()
<< " has invalid frame references, dropping frame.";
return last_continuous_frame_id;
@@ -381,11 +386,19 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
if (frames_.size() >= kMaxFramesBuffered) {
if (frame->is_keyframe()) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (KF + Full buffer)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "Inserting keyframe " << frame->Id()
<< " but buffer is full, clearing"
" buffer and inserting the frame.";
ClearFramesAndHistory();
} else {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Full buffer)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "Frame " << frame->Id()
<< " could not be inserted due to the frame "
"buffer being full, dropping frame.";
@@ -404,11 +417,19 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
// reconfiguration or some other reason. Even though this is not according
// to spec we can still continue to decode from this frame if it is a
// keyframe.
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (OOO + PicId jump)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING)
<< "A jump in frame id was detected, clearing buffer.";
ClearFramesAndHistory();
last_continuous_frame_id = -1;
} else {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frame dropped (Out of order)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "Frame " << frame->Id() << " inserted after frame "
<< *last_decoded_frame
<< " was handed off for decoding, dropping frame.";
@@ -421,6 +442,10 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
// when the frame id make large jumps mid stream.
if (!frames_.empty() && frame->Id() < frames_.begin()->first &&
frames_.rbegin()->first < frame->Id()) {
+ TRACE_EVENT2("webrtc",
+ "FrameBuffer::InsertFrame Frames dropped (PicId big-jump)",
+ "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id",
+ frame->Id());
RTC_LOG(LS_WARNING) << "A jump in frame id was detected, clearing buffer.";
ClearFramesAndHistory();
last_continuous_frame_id = -1;
diff --git a/video/receive_statistics_proxy2.cc b/video/receive_statistics_proxy2.cc
index f5011c46ef..508c36eaaf 100644
--- a/video/receive_statistics_proxy2.cc
+++ b/video/receive_statistics_proxy2.cc
@@ -20,6 +20,7 @@
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
#include "video/video_receive_stream2.h"
@@ -921,6 +922,9 @@ void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
VideoContentType content_type) {
RTC_DCHECK_RUN_ON(&main_thread_);
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnCompleteFrame",
+ "remote_ssrc", remote_ssrc_, "is_keyframe", is_keyframe);
+
if (is_keyframe) {
++stats_.frame_counts.key_frames;
} else {
@@ -952,6 +956,8 @@ void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
// Can be called on either the decode queue or the worker thread
// See FrameBuffer2 for more details.
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnDroppedFrames",
+ "remote_ssrc", remote_ssrc_, "frames_dropped", frames_dropped);
worker_thread_->PostTask(
SafeTask(task_safety_.flag(), [frames_dropped, this]() {
RTC_DCHECK_RUN_ON(&main_thread_);
@@ -962,6 +968,9 @@ void ReceiveStatisticsProxy::OnDroppedFrames(uint32_t frames_dropped) {
void ReceiveStatisticsProxy::OnDiscardedPackets(uint32_t packets_discarded) {
// Can be called on either the decode queue or the worker thread
// See FrameBuffer2 for more details.
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnDiscardedPackets",
+ "remote_ssrc", remote_ssrc_, "packets_discarded",
+ packets_discarded);
worker_thread_->PostTask(
SafeTask(task_safety_.flag(), [packets_discarded, this]() {
RTC_DCHECK_RUN_ON(&main_thread_);
@@ -991,6 +1000,8 @@ void ReceiveStatisticsProxy::OnStreamInactive() {
void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms) {
RTC_DCHECK_RUN_ON(&main_thread_);
+ TRACE_EVENT2("webrtc", "ReceiveStatisticsProxy::OnRttUpdate",
+ "remote_ssrc", remote_ssrc_, "avg_rtt_ms", avg_rtt_ms);
avg_rtt_ms_ = avg_rtt_ms;
}
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index a5d5f637e5..eb0a7787ac 100644
index eb0a7787ac..8055ac0e0f 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -44,6 +44,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/trace_event.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/ntp_time.h"
@@ -735,6 +735,12 @@ void RtpVideoStreamReceiver2::OnRtpPacket(const RtpPacketReceived& packet) {
@@ -1223,6 +1224,9 @@ void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
packet_infos_.upper_bound(unwrapped_rtp_seq_num));
uint32_t num_packets_cleared = packet_buffer_.ClearTo(seq_num);
if (num_packets_cleared > 0) {
+ TRACE_EVENT2("webrtc",
+ "RtpVideoStreamReceiver2::FrameDecoded Cleared Old Packets",
+ "remote_ssrc", config_.rtp.remote_ssrc, "seq_num", seq_num);
vcm_receive_statistics_->OnDiscardedPackets(num_packets_cleared);
}
reference_finder_->ClearTo(seq_num);
diff --git a/video/video_stream_buffer_controller.cc b/video/video_stream_buffer_controller.cc
index 7e44eff39a..37724a8338 100644
--- a/video/video_stream_buffer_controller.cc
+++ b/video/video_stream_buffer_controller.cc
@@ -28,6 +28,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/thread_annotations.h"
+#include "rtc_base/trace_event.h"
#include "video/frame_decode_scheduler.h"
#include "video/frame_decode_timing.h"
#include "video/task_queue_frame_decode_scheduler.h"
@@ -139,6 +140,9 @@ absl::optional<int64_t> VideoStreamBufferController::InsertFrame(
std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
FrameMetadata metadata(*frame);
+ const uint32_t ssrc =
+ frame->PacketInfos().empty() ? 0 : frame->PacketInfos()[0].ssrc();
+ const int64_t frameId = frame->Id();
int complete_units = buffer_->GetTotalNumberOfContinuousTemporalUnits();
if (buffer_->InsertFrame(std::move(frame))) {
RTC_DCHECK(metadata.receive_time) << "Frame receive time must be set!";
@@ -149,6 +153,9 @@ absl::optional<int64_t> VideoStreamBufferController::InsertFrame(
*metadata.receive_time);
}
if (complete_units < buffer_->GetTotalNumberOfContinuousTemporalUnits()) {
+ TRACE_EVENT2("webrtc",
+ "VideoStreamBufferController::InsertFrame Frame Complete",
+ "remote_ssrc", ssrc, "frame_id", frameId);
stats_proxy_->OnCompleteFrame(metadata.is_keyframe, metadata.size,
metadata.contentType);
MaybeScheduleFrameForRelease();
void RtpVideoStreamReceiver2::RequestKeyFrame() {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
+ TRACE_EVENT2("webrtc", "RtpVideoStreamReceiver2::RequestKeyFrame",
+ "remote_ssrc", config_.rtp.remote_ssrc, "method",
+ keyframe_request_method_ == KeyFrameReqMethod::kPliRtcp ? "PLI"
+ : keyframe_request_method_ == KeyFrameReqMethod::kFirRtcp ? "FIR"
+ : keyframe_request_method_ == KeyFrameReqMethod::kNone ? "None"
+ : "Other");
// TODO(bugs.webrtc.org/10336): Allow the sender to ignore key frame requests
// issued by anything other than the LossNotificationController if it (the
// sender) is relying on LNTF alone.
--
2.34.1

59
third_party/libwebrtc/moz-patch-stack/0081.patch поставляемый
Просмотреть файл

@ -1,30 +1,45 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Thu, 6 Jan 2022 00:16:00 +0000
Subject: Bug 1748458 - Add TRACE_EVENT for keyframe requests. r=bwc
Date: Wed, 11 Jan 2023 22:42:00 +0000
Subject: Bug 1800942 - Add DCHECKs to
TimestampExtrapolator::ExtrapolateLocalTime. r=mjf
Differential Revision: https://phabricator.services.mozilla.com/D135113
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/5b2a7894ef1cf096d0e8977754507c0820e757fc
Differential Revision: https://phabricator.services.mozilla.com/D166536
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c5df7f40392464ffc63f44a53ddcaab2091741e0
---
video/rtp_video_stream_receiver2.cc | 6 ++++++
1 file changed, 6 insertions(+)
modules/video_coding/timing/timestamp_extrapolator.cc | 7 +++++++
1 file changed, 7 insertions(+)
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index eb0a7787ac..8055ac0e0f 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -735,6 +735,12 @@ void RtpVideoStreamReceiver2::OnRtpPacket(const RtpPacketReceived& packet) {
diff --git a/modules/video_coding/timing/timestamp_extrapolator.cc b/modules/video_coding/timing/timestamp_extrapolator.cc
index c91aa1a362..dc62ac674a 100644
--- a/modules/video_coding/timing/timestamp_extrapolator.cc
+++ b/modules/video_coding/timing/timestamp_extrapolator.cc
@@ -125,6 +125,7 @@ void TimestampExtrapolator::Update(Timestamp now, uint32_t ts90khz) {
absl::optional<Timestamp> TimestampExtrapolator::ExtrapolateLocalTime(
uint32_t timestamp90khz) const {
int64_t unwrapped_ts90khz = unwrapper_.PeekUnwrap(timestamp90khz);
+ RTC_DCHECK_GE(unwrapped_ts90khz, 0);
void RtpVideoStreamReceiver2::RequestKeyFrame() {
RTC_DCHECK_RUN_ON(&worker_task_checker_);
+ TRACE_EVENT2("webrtc", "RtpVideoStreamReceiver2::RequestKeyFrame",
+ "remote_ssrc", config_.rtp.remote_ssrc, "method",
+ keyframe_request_method_ == KeyFrameReqMethod::kPliRtcp ? "PLI"
+ : keyframe_request_method_ == KeyFrameReqMethod::kFirRtcp ? "FIR"
+ : keyframe_request_method_ == KeyFrameReqMethod::kNone ? "None"
+ : "Other");
// TODO(bugs.webrtc.org/10336): Allow the sender to ignore key frame requests
// issued by anything other than the LossNotificationController if it (the
// sender) is relying on LNTF alone.
if (!first_unwrapped_timestamp_) {
return absl::nullopt;
@@ -132,12 +133,18 @@ absl::optional<Timestamp> TimestampExtrapolator::ExtrapolateLocalTime(
constexpr double kRtpTicksPerMs = 90;
TimeDelta diff = TimeDelta::Millis(
(unwrapped_ts90khz - *prev_unwrapped_timestamp_) / kRtpTicksPerMs);
+ if (diff.ms() < 0) {
+ RTC_DCHECK_GE(prev_.ms(), -diff.ms());
+ }
return prev_ + diff;
} else if (w_[0] < 1e-3) {
return start_;
} else {
double timestampDiff = unwrapped_ts90khz - *first_unwrapped_timestamp_;
auto diff_ms = static_cast<int64_t>((timestampDiff - w_[1]) / w_[0] + 0.5);
+ if (diff_ms < 0) {
+ RTC_DCHECK_GE(start_.ms(), -diff_ms);
+ }
return start_ + TimeDelta::Millis(diff_ms);
}
}
--
2.34.1

59
third_party/libwebrtc/moz-patch-stack/0082.patch поставляемый
Просмотреть файл

@ -1,45 +1,28 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Wed, 11 Jan 2023 22:42:00 +0000
Subject: Bug 1800942 - Add DCHECKs to
TimestampExtrapolator::ExtrapolateLocalTime. r=mjf
Date: Wed, 8 Feb 2023 08:01:00 +0000
Subject: Bug 1814692 - Don't attempt realtime scheduling rtc::PlatformThreads.
r=webrtc-reviewers,bwc
Differential Revision: https://phabricator.services.mozilla.com/D166536
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c5df7f40392464ffc63f44a53ddcaab2091741e0
Differential Revision: https://phabricator.services.mozilla.com/D169036
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9e64a965e26c8379261466e5273c3b383164b2c7
---
modules/video_coding/timing/timestamp_extrapolator.cc | 7 +++++++
1 file changed, 7 insertions(+)
rtc_base/platform_thread.cc | 3 +++
1 file changed, 3 insertions(+)
diff --git a/modules/video_coding/timing/timestamp_extrapolator.cc b/modules/video_coding/timing/timestamp_extrapolator.cc
index c91aa1a362..dc62ac674a 100644
--- a/modules/video_coding/timing/timestamp_extrapolator.cc
+++ b/modules/video_coding/timing/timestamp_extrapolator.cc
@@ -125,6 +125,7 @@ void TimestampExtrapolator::Update(Timestamp now, uint32_t ts90khz) {
absl::optional<Timestamp> TimestampExtrapolator::ExtrapolateLocalTime(
uint32_t timestamp90khz) const {
int64_t unwrapped_ts90khz = unwrapper_.PeekUnwrap(timestamp90khz);
+ RTC_DCHECK_GE(unwrapped_ts90khz, 0);
if (!first_unwrapped_timestamp_) {
return absl::nullopt;
@@ -132,12 +133,18 @@ absl::optional<Timestamp> TimestampExtrapolator::ExtrapolateLocalTime(
constexpr double kRtpTicksPerMs = 90;
TimeDelta diff = TimeDelta::Millis(
(unwrapped_ts90khz - *prev_unwrapped_timestamp_) / kRtpTicksPerMs);
+ if (diff.ms() < 0) {
+ RTC_DCHECK_GE(prev_.ms(), -diff.ms());
+ }
return prev_ + diff;
} else if (w_[0] < 1e-3) {
return start_;
} else {
double timestampDiff = unwrapped_ts90khz - *first_unwrapped_timestamp_;
auto diff_ms = static_cast<int64_t>((timestampDiff - w_[1]) / w_[0] + 0.5);
+ if (diff_ms < 0) {
+ RTC_DCHECK_GE(start_.ms(), -diff_ms);
+ }
return start_ + TimeDelta::Millis(diff_ms);
}
}
diff --git a/rtc_base/platform_thread.cc b/rtc_base/platform_thread.cc
index 71a9f1b224..bcbb784b97 100644
--- a/rtc_base/platform_thread.cc
+++ b/rtc_base/platform_thread.cc
@@ -50,6 +50,9 @@ bool SetPriority(ThreadPriority priority) {
// TODO(tommi): Switch to the same mechanism as Chromium uses for changing
// thread priorities.
return true;
+#elif defined(WEBRTC_MOZILLA_BUILD) && defined(WEBRTC_LINUX)
+ // Only realtime audio uses realtime scheduling in Firefox.
+ return true;
#else
const int policy = SCHED_FIFO;
const int min_prio = sched_get_priority_min(policy);
--
2.34.1

48
third_party/libwebrtc/moz-patch-stack/0083.patch поставляемый
Просмотреть файл

@ -1,28 +1,34 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Wed, 8 Feb 2023 08:01:00 +0000
Subject: Bug 1814692 - Don't attempt realtime scheduling rtc::PlatformThreads.
r=webrtc-reviewers,bwc
Date: Fri, 24 Feb 2023 15:01:00 +0100
Subject: Bug 1817024 - (fix-0e2cf6cc01) Skip library
create_peer_connection_quality_test_frame_generator. r?mjf!
Differential Revision: https://phabricator.services.mozilla.com/D169036
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9e64a965e26c8379261466e5273c3b383164b2c7
Differential Revision: https://phabricator.services.mozilla.com/D170887
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/fbbc1bf963fda30bca26ae6aac0c3459b8ebea6f
---
rtc_base/platform_thread.cc | 3 +++
1 file changed, 3 insertions(+)
api/BUILD.gn | 2 ++
1 file changed, 2 insertions(+)
diff --git a/rtc_base/platform_thread.cc b/rtc_base/platform_thread.cc
index 71a9f1b224..bcbb784b97 100644
--- a/rtc_base/platform_thread.cc
+++ b/rtc_base/platform_thread.cc
@@ -50,6 +50,9 @@ bool SetPriority(ThreadPriority priority) {
// TODO(tommi): Switch to the same mechanism as Chromium uses for changing
// thread priorities.
return true;
+#elif defined(WEBRTC_MOZILLA_BUILD) && defined(WEBRTC_LINUX)
+ // Only realtime audio uses realtime scheduling in Firefox.
+ return true;
#else
const int policy = SCHED_FIFO;
const int min_prio = sched_get_priority_min(policy);
diff --git a/api/BUILD.gn b/api/BUILD.gn
index ab5d6c91ce..3f313e2743 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -702,6 +702,7 @@ rtc_library("create_frame_generator") {
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+if (!build_with_mozilla) {
rtc_library("create_peer_connection_quality_test_frame_generator") {
visibility = [ "*" ]
testonly = true
@@ -718,6 +719,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_source_set("libjingle_logging_api") {
visibility = [ "*" ]
--
2.34.1

197
third_party/libwebrtc/moz-patch-stack/0084.patch поставляемый
Просмотреть файл

@ -1,34 +1,189 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Fri, 24 Feb 2023 15:01:00 +0100
Subject: Bug 1817024 - (fix-0e2cf6cc01) Skip library
create_peer_connection_quality_test_frame_generator. r?mjf!
From: Michael Froman <mfroman@mozilla.com>
Date: Wed, 8 Mar 2023 00:26:00 +0000
Subject: Bug 1820869 - avoid building unreachable files. r=ng,webrtc-reviewers
Differential Revision: https://phabricator.services.mozilla.com/D170887
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/fbbc1bf963fda30bca26ae6aac0c3459b8ebea6f
Differential Revision: https://phabricator.services.mozilla.com/D171922
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/88b3cc6bbece7c53d00e124713330f3d34d2789d
---
api/BUILD.gn | 2 ++
1 file changed, 2 insertions(+)
BUILD.gn | 9 +++++++++
call/BUILD.gn | 10 ++++++++++
media/BUILD.gn | 7 ++++++-
modules/audio_device/BUILD.gn | 11 ++++++++++-
rtc_base/BUILD.gn | 2 ++
webrtc.gni | 2 +-
6 files changed, 38 insertions(+), 3 deletions(-)
diff --git a/api/BUILD.gn b/api/BUILD.gn
index ab5d6c91ce..3f313e2743 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -702,6 +702,7 @@ rtc_library("create_frame_generator") {
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
diff --git a/BUILD.gn b/BUILD.gn
index 873aec3850..75e536e046 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -549,6 +549,15 @@ if (!build_with_chromium) {
"api/video:video_rtp_headers",
"test:rtp_test_utils",
]
+ # Added when we removed deps in other places to avoid building
+ # unreachable sources. See Bug 1820869.
+ deps += [
+ "api/video_codecs:video_codecs_api",
+ "api/video_codecs:rtc_software_fallback_wrappers",
+ "media:rtc_encoder_simulcast_proxy",
+ "modules/video_coding:webrtc_vp8",
+ "modules/video_coding:webrtc_vp9",
+ ]
} else {
deps += [
"api",
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 26618aee80..fb23b7ef39 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -352,6 +352,16 @@ rtc_library("call") {
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
+ if (build_with_mozilla) { # See Bug 1820869.
+ sources -= [
+ "call_factory.cc",
+ "degraded_call.cc",
+ ]
+ deps -= [
+ ":fake_network",
+ ":simulated_network",
+ ]
+ }
}
+if (!build_with_mozilla) {
rtc_library("create_peer_connection_quality_test_frame_generator") {
visibility = [ "*" ]
testonly = true
@@ -718,6 +719,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
rtc_source_set("receive_stream_interface") {
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 4ddc8349a8..daca67e033 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -442,7 +442,10 @@ rtc_library("rtc_internal_video_codecs") {
"../test:fake_video_codecs",
]
if (build_with_mozilla) {
- deps -= [ "../test:fake_video_codecs" ]
+ deps -= [
+ "../modules/video_coding:webrtc_multiplex", # See Bug 1820869.
+ "../test:fake_video_codecs",
+ ]
}
if (enable_libaom) {
@@ -477,6 +480,8 @@ rtc_library("rtc_internal_video_codecs") {
sources -= [
"engine/fake_video_codec_factory.cc",
"engine/fake_video_codec_factory.h",
+ "engine/internal_encoder_factory.cc", # See Bug 1820869.
+ "engine/multiplex_codec_factory.cc", # See Bug 1820869.
]
}
}
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index e35a442025..61cd531edd 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -30,6 +30,7 @@ rtc_source_set("audio_device_default") {
}
rtc_source_set("audio_device") {
+if (!build_with_mozilla) { # See Bug 1820869.
visibility = [ "*" ]
public_deps = [
":audio_device_api",
@@ -40,6 +41,7 @@ rtc_source_set("audio_device") {
":audio_device_impl",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_source_set("libjingle_logging_api") {
rtc_source_set("audio_device_api") {
visibility = [ "*" ]
@@ -58,6 +60,7 @@ rtc_source_set("audio_device_api") {
}
rtc_library("audio_device_buffer") {
+if (!build_with_mozilla) { # See Bug 1820869.
sources = [
"audio_device_buffer.cc",
"audio_device_buffer.h",
@@ -85,6 +88,7 @@ rtc_library("audio_device_buffer") {
"../../system_wrappers:metrics",
]
}
+}
rtc_library("audio_device_generic") {
sources = [
@@ -180,6 +184,7 @@ rtc_source_set("audio_device_module_from_input_and_output") {
# Contains default implementations of webrtc::AudioDeviceModule for Windows,
# Linux, Mac, iOS and Android.
rtc_library("audio_device_impl") {
+if (!build_with_mozilla) { # See Bug 1820869.
visibility = [ "*" ]
deps = [
":audio_device_api",
@@ -373,6 +378,7 @@ rtc_library("audio_device_impl") {
]
}
}
+}
if (is_mac) {
rtc_source_set("audio_device_impl_frameworks") {
@@ -390,6 +396,7 @@ if (is_mac) {
}
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_source_set("mock_audio_device") {
visibility = [ "*" ]
testonly = true
@@ -406,8 +413,10 @@ rtc_source_set("mock_audio_device") {
"../../test:test_support",
]
}
+}
-if (rtc_include_tests && !build_with_chromium) {
+# See Bug 1820869 for !build_with_mozilla.
+if (rtc_include_tests && !build_with_chromium && !build_with_mozilla) {
rtc_library("audio_device_unittests") {
testonly = true
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 3cd0bfff06..0b1e2a6208 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -283,6 +283,7 @@ rtc_library("sample_counter") {
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_library("timestamp_aligner") {
visibility = [ "*" ]
sources = [
@@ -296,6 +297,7 @@ rtc_library("timestamp_aligner") {
"system:rtc_export",
]
}
+}
rtc_library("zero_memory") {
visibility = [ "*" ]
diff --git a/webrtc.gni b/webrtc.gni
index 1b21d329b2..46a9433141 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -221,7 +221,7 @@ declare_args() {
# video codecs they depends on will not be included in libwebrtc.{a|lib}
# (they will still be included in libjingle_peerconnection_so.so and
# WebRTC.framework)
- rtc_include_builtin_video_codecs = true
+ rtc_include_builtin_video_codecs = !build_with_mozilla # See Bug 1820869.
# When set to true and in a standalone build, it will undefine UNICODE and
# _UNICODE (which are always defined globally by the Chromium Windows
--
2.34.1

201
third_party/libwebrtc/moz-patch-stack/0085.patch поставляемый
Просмотреть файл

@ -1,189 +1,34 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Wed, 8 Mar 2023 00:26:00 +0000
Subject: Bug 1820869 - avoid building unreachable files. r=ng,webrtc-reviewers
Date: Wed, 12 Apr 2023 16:03:00 +0000
Subject: Bug 1826428 - remove libwebrtc's jvm_android.cc from build
r=ng,webrtc-reviewers
Differential Revision: https://phabricator.services.mozilla.com/D171922
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/88b3cc6bbece7c53d00e124713330f3d34d2789d
Based on info from John Lin and previous try runs, we're almost
certainly not using this. Let's try removing it from the build
and landing it. If no problems emerge, we'll be able to remove
our custom changes to upstream code in jvm_android.cc.
Differential Revision: https://phabricator.services.mozilla.com/D174793
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/dca1b97525487ae57d43ced1ebdb4a2d9c9dae89
---
BUILD.gn | 9 +++++++++
call/BUILD.gn | 10 ++++++++++
media/BUILD.gn | 7 ++++++-
modules/audio_device/BUILD.gn | 11 ++++++++++-
rtc_base/BUILD.gn | 2 ++
webrtc.gni | 2 +-
6 files changed, 38 insertions(+), 3 deletions(-)
modules/utility/BUILD.gn | 4 ++++
1 file changed, 4 insertions(+)
diff --git a/BUILD.gn b/BUILD.gn
index 873aec3850..75e536e046 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -549,6 +549,15 @@ if (!build_with_chromium) {
"api/video:video_rtp_headers",
"test:rtp_test_utils",
]
+ # Added when we removed deps in other places to avoid building
+ # unreachable sources. See Bug 1820869.
+ deps += [
+ "api/video_codecs:video_codecs_api",
+ "api/video_codecs:rtc_software_fallback_wrappers",
+ "media:rtc_encoder_simulcast_proxy",
+ "modules/video_coding:webrtc_vp8",
+ "modules/video_coding:webrtc_vp9",
+ ]
} else {
deps += [
"api",
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 26618aee80..fb23b7ef39 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -352,6 +352,16 @@ rtc_library("call") {
"//third_party/abseil-cpp/absl/strings",
"//third_party/abseil-cpp/absl/types:optional",
]
+ if (build_with_mozilla) { # See Bug 1820869.
+ sources -= [
+ "call_factory.cc",
+ "degraded_call.cc",
+ ]
+ deps -= [
+ ":fake_network",
+ ":simulated_network",
+ ]
+ }
}
rtc_source_set("receive_stream_interface") {
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 4ddc8349a8..daca67e033 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -442,7 +442,10 @@ rtc_library("rtc_internal_video_codecs") {
"../test:fake_video_codecs",
]
if (build_with_mozilla) {
- deps -= [ "../test:fake_video_codecs" ]
+ deps -= [
+ "../modules/video_coding:webrtc_multiplex", # See Bug 1820869.
+ "../test:fake_video_codecs",
+ ]
}
if (enable_libaom) {
@@ -477,6 +480,8 @@ rtc_library("rtc_internal_video_codecs") {
sources -= [
"engine/fake_video_codec_factory.cc",
"engine/fake_video_codec_factory.h",
+ "engine/internal_encoder_factory.cc", # See Bug 1820869.
+ "engine/multiplex_codec_factory.cc", # See Bug 1820869.
diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
index 3fe4ca8c92..46bca17f02 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -47,6 +47,10 @@ rtc_source_set("utility") {
"../../rtc_base:platform_thread",
"../../rtc_base/system:arch",
]
}
}
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index e35a442025..61cd531edd 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -30,6 +30,7 @@ rtc_source_set("audio_device_default") {
}
rtc_source_set("audio_device") {
+if (!build_with_mozilla) { # See Bug 1820869.
visibility = [ "*" ]
public_deps = [
":audio_device_api",
@@ -40,6 +41,7 @@ rtc_source_set("audio_device") {
":audio_device_impl",
]
}
+}
rtc_source_set("audio_device_api") {
visibility = [ "*" ]
@@ -58,6 +60,7 @@ rtc_source_set("audio_device_api") {
}
rtc_library("audio_device_buffer") {
+if (!build_with_mozilla) { # See Bug 1820869.
sources = [
"audio_device_buffer.cc",
"audio_device_buffer.h",
@@ -85,6 +88,7 @@ rtc_library("audio_device_buffer") {
"../../system_wrappers:metrics",
]
}
+}
rtc_library("audio_device_generic") {
sources = [
@@ -180,6 +184,7 @@ rtc_source_set("audio_device_module_from_input_and_output") {
# Contains default implementations of webrtc::AudioDeviceModule for Windows,
# Linux, Mac, iOS and Android.
rtc_library("audio_device_impl") {
+if (!build_with_mozilla) { # See Bug 1820869.
visibility = [ "*" ]
deps = [
":audio_device_api",
@@ -373,6 +378,7 @@ rtc_library("audio_device_impl") {
]
}
}
+}
if (is_mac) {
rtc_source_set("audio_device_impl_frameworks") {
@@ -390,6 +396,7 @@ if (is_mac) {
+
+ if (build_with_mozilla) {
+ sources -= [ "source/jvm_android.cc" ]
+ }
}
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_source_set("mock_audio_device") {
visibility = [ "*" ]
testonly = true
@@ -406,8 +413,10 @@ rtc_source_set("mock_audio_device") {
"../../test:test_support",
]
}
+}
-if (rtc_include_tests && !build_with_chromium) {
+# See Bug 1820869 for !build_with_mozilla.
+if (rtc_include_tests && !build_with_chromium && !build_with_mozilla) {
rtc_library("audio_device_unittests") {
testonly = true
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 3cd0bfff06..0b1e2a6208 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -283,6 +283,7 @@ rtc_library("sample_counter") {
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+if (!build_with_mozilla) { # See Bug 1820869.
rtc_library("timestamp_aligner") {
visibility = [ "*" ]
sources = [
@@ -296,6 +297,7 @@ rtc_library("timestamp_aligner") {
"system:rtc_export",
]
}
+}
rtc_library("zero_memory") {
visibility = [ "*" ]
diff --git a/webrtc.gni b/webrtc.gni
index 1b21d329b2..46a9433141 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -221,7 +221,7 @@ declare_args() {
# video codecs they depends on will not be included in libwebrtc.{a|lib}
# (they will still be included in libjingle_peerconnection_so.so and
# WebRTC.framework)
- rtc_include_builtin_video_codecs = true
+ rtc_include_builtin_video_codecs = !build_with_mozilla # See Bug 1820869.
# When set to true and in a standalone build, it will undefine UNICODE and
# _UNICODE (which are always defined globally by the Chromium Windows
--
2.34.1

181
third_party/libwebrtc/moz-patch-stack/0086.patch поставляемый
Просмотреть файл

@ -1,31 +1,166 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Wed, 12 Apr 2023 16:03:00 +0000
Subject: Bug 1826428 - remove libwebrtc's jvm_android.cc from build
r=ng,webrtc-reviewers
From: Jan Grulich <jgrulich@redhat.com>
Date: Mon, 20 Feb 2023 21:25:00 +0000
Subject: Bug 1817263 - fix OS picker behavior under Wayland r=ng,jib,stransky
Based on info from John Lin and previous try runs, we're almost
certainly not using this. Let's try removing it from the build
and landing it. If no problems emerge, we'll be able to remove
our custom changes to upstream code in jvm_android.cc.
Recent WebRTC backports and changes that are about to be backported from
upstream to Firefox breaks and will break how we work with PipWire based
desktop capturer. Currently when constructing device list, a fallback to
ScreenCapturerX11 is used, as we don't call set_allow_pipewire(), which
wouldn't make a difference anyway. In such case the only thing we need
is a placeholder for a screen that will request OS level prompt. We also
need a way to request both screens and windows in one xdg-desktop-portal
call as recent WebRTC made each type be called separately, therefore the
introduction of GenericCapturer. Lastly we need to make sure when there
is a MediaDevice requesting the OS prompt, that it will be checked as
first.
Differential Revision: https://phabricator.services.mozilla.com/D174793
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/dca1b97525487ae57d43ced1ebdb4a2d9c9dae89
In order to use unmodified libwebrtc, Firefox would need to rework the
OS picker to request each type (screens and windows) separately so we
can just use regular ScreenCapturer and WindowCapturer. This should be
done ideally the way Chromium does it, where users can actually see
even the preview of what they picked over xdg-desktop-portal before it
is actually shared with requesting web page and they also have option
to make the request again in case they picked a wrong window or screen.
Differential Revision: https://phabricator.services.mozilla.com/D169627
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/acd6266642951aacf8915a56777c780cae9e9af3
---
modules/utility/BUILD.gn | 4 ++++
1 file changed, 4 insertions(+)
.../desktop_capture/desktop_capture_types.h | 2 +-
modules/desktop_capture/desktop_capturer.cc | 28 +++++++++++++++++++
modules/desktop_capture/desktop_capturer.h | 13 +++++++++
.../linux/wayland/base_capturer_pipewire.cc | 11 +-------
.../linux/wayland/screencast_portal.cc | 2 ++
5 files changed, 45 insertions(+), 11 deletions(-)
diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
index 3fe4ca8c92..46bca17f02 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -47,6 +47,10 @@ rtc_source_set("utility") {
"../../rtc_base:platform_thread",
"../../rtc_base/system:arch",
]
diff --git a/modules/desktop_capture/desktop_capture_types.h b/modules/desktop_capture/desktop_capture_types.h
index 381d1021c4..e777a45f92 100644
--- a/modules/desktop_capture/desktop_capture_types.h
+++ b/modules/desktop_capture/desktop_capture_types.h
@@ -19,7 +19,7 @@ typedef int pid_t; // matching what used to be in
namespace webrtc {
-enum class CaptureType { kWindow, kScreen };
+enum class CaptureType { kWindow, kScreen, kAnyScreenContent };
// Type used to identify windows on the desktop. Values are platform-specific:
// - On Windows: HWND cast to intptr_t.
diff --git a/modules/desktop_capture/desktop_capturer.cc b/modules/desktop_capture/desktop_capturer.cc
index 4debc75394..65a4045f8c 100644
--- a/modules/desktop_capture/desktop_capturer.cc
+++ b/modules/desktop_capture/desktop_capturer.cc
@@ -25,6 +25,10 @@
#include "rtc_base/win/windows_version.h"
#endif // defined(RTC_ENABLE_WIN_WGC)
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+#include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
+#endif
+
+ if (build_with_mozilla) {
+ sources -= [ "source/jvm_android.cc" ]
+ }
namespace webrtc {
DesktopCapturer::~DesktopCapturer() = default;
@@ -78,6 +82,30 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateWindowCapturer(
return capturer;
}
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateGenericCapturer(
+ const DesktopCaptureOptions& options) {
+ std::unique_ptr<DesktopCapturer> capturer = CreateRawGenericCapturer(options);
+ if (capturer && options.detect_updated_region()) {
+ capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
+ }
+
+ return capturer;
+}
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawGenericCapturer(
+ const DesktopCaptureOptions& options) {
+ if (options.allow_pipewire() && DesktopCapturer::IsRunningUnderWayland()) {
+ return std::make_unique<BaseCapturerPipeWire>(options,
+ CaptureType::kAnyScreenContent);
+ }
+
+ return nullptr;
+}
+#endif // defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+
// static
std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
const DesktopCaptureOptions& options) {
diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
index 3fdcd2c5d0..2713c0e603 100644
--- a/modules/desktop_capture/desktop_capturer.h
+++ b/modules/desktop_capture/desktop_capturer.h
@@ -168,6 +168,12 @@ class RTC_EXPORT DesktopCapturer {
// The return value if `pos` is out of the scope of the source is undefined.
virtual bool IsOccluded(const DesktopVector& pos);
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+ // Creates a DesktopCapturer instance which targets to capture windows and screens.
+ static std::unique_ptr<DesktopCapturer> CreateGenericCapturer(
+ const DesktopCaptureOptions& options);
+#endif
+
// Creates a DesktopCapturer instance which targets to capture windows.
static std::unique_ptr<DesktopCapturer> CreateWindowCapturer(
const DesktopCaptureOptions& options);
@@ -196,6 +202,13 @@ class RTC_EXPORT DesktopCapturer {
// CroppingWindowCapturer needs to create raw capturers without wrappers, so
// the following two functions are protected.
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+ // Creates a platform specific DesktopCapturer instance which targets to
+ // capture windows and screens.
+ static std::unique_ptr<DesktopCapturer> CreateRawGenericCapturer(
+ const DesktopCaptureOptions& options);
+#endif
+
// Creates a platform specific DesktopCapturer instance which targets to
// capture windows.
static std::unique_ptr<DesktopCapturer> CreateRawWindowCapturer(
diff --git a/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc b/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc
index dae2b70510..cf4f7dc9aa 100644
--- a/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc
+++ b/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc
@@ -165,15 +165,6 @@ void BaseCapturerPipeWire::CaptureFrame() {
callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
}
-// Keep in sync with defines at browser/actors/WebRTCParent.jsm
-// With PipeWire we can't select which system resource is shared so
-// we don't create a window/screen list. Instead we place these constants
-// as window name/id so frontend code can identify PipeWire backend
-// and does not try to create screen/window preview.
-
-#define PIPEWIRE_ID 0xaffffff
-#define PIPEWIRE_NAME "####_PIPEWIRE_PORTAL_####"
-
bool BaseCapturerPipeWire::GetSourceList(SourceList* sources) {
RTC_DCHECK(sources->size() == 0);
// List of available screens is already presented by the xdg-desktop-portal,
@@ -190,7 +181,7 @@ bool BaseCapturerPipeWire::GetSourceList(SourceList* sources) {
bool BaseCapturerPipeWire::SelectSource(SourceId id) {
// Screen selection is handled by the xdg-desktop-portal.
selected_source_id_ = id;
- return id == PIPEWIRE_ID;
+ return true;
}
DelegatedSourceListController*
diff --git a/modules/desktop_capture/linux/wayland/screencast_portal.cc b/modules/desktop_capture/linux/wayland/screencast_portal.cc
index abfade56e7..e7aaee001b 100644
--- a/modules/desktop_capture/linux/wayland/screencast_portal.cc
+++ b/modules/desktop_capture/linux/wayland/screencast_portal.cc
@@ -41,6 +41,8 @@ ScreenCastPortal::CaptureSourceType ScreenCastPortal::ToCaptureSourceType(
return ScreenCastPortal::CaptureSourceType::kScreen;
case CaptureType::kWindow:
return ScreenCastPortal::CaptureSourceType::kWindow;
+ case CaptureType::kAnyScreenContent:
+ return ScreenCastPortal::CaptureSourceType::kAnyScreenContent;
}
}

201
third_party/libwebrtc/moz-patch-stack/0087.patch поставляемый
Просмотреть файл

@ -1,169 +1,100 @@
From: Jan Grulich <jgrulich@redhat.com>
Date: Mon, 20 Feb 2023 21:25:00 +0000
Subject: Bug 1817263 - fix OS picker behavior under Wayland r=ng,jib,stransky
Date: Mon, 27 Feb 2023 13:57:00 +0000
Subject: Bug 1819044 - fix build non-pipewire builds
r=webrtc-reviewers,pehrsons
Recent WebRTC backports and changes that are about to be backported from
upstream to Firefox breaks and will break how we work with PipWire based
desktop capturer. Currently when constructing device list, a fallback to
ScreenCapturerX11 is used, as we don't call set_allow_pipewire(), which
wouldn't make a difference anyway. In such case the only thing we need
is a placeholder for a screen that will request OS level prompt. We also
need a way to request both screens and windows in one xdg-desktop-portal
call as recent WebRTC made each type be called separately, therefore the
introduction of GenericCapturer. Lastly we need to make sure when there
is a MediaDevice requesting the OS prompt, that it will be checked as
first.
We should check only for PipeWire presence when building code specific
to PipeWire.
In order to use unmodified libwebrtc, Firefox would need to rework the
OS picker to request each type (screens and windows) separately so we
can just use regular ScreenCapturer and WindowCapturer. This should be
done ideally the way Chromium does it, where users can actually see
even the preview of what they picked over xdg-desktop-portal before it
is actually shared with requesting web page and they also have option
to make the request again in case they picked a wrong window or screen.
Differential Revision: https://phabricator.services.mozilla.com/D171071
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7a879ad084a6e9768479c73cc5c3f4e9d95a2ab9
Differential Revision: https://phabricator.services.mozilla.com/D169627
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/acd6266642951aacf8915a56777c780cae9e9af3
Also includes:
Bug 1819044 - fix build non-pipewire builds (attempt #2) r=webrtc-reviewers,pehrsons
Make the new API available to everyone and just return an empty capturer
in case when building without PipeWire. It will not make any difference
because using X11 based capturers on Wayland is useless anyway so if we
fail for missing PipeWire on Wayland, it will have the same outcome.
Differential Revision: https://phabricator.services.mozilla.com/D171192
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/ad247b0aac896d884eba5e40f0ec8a9f50d8b85b
---
.../desktop_capture/desktop_capture_types.h | 2 +-
modules/desktop_capture/desktop_capturer.cc | 28 +++++++++++++++++++
modules/desktop_capture/desktop_capturer.h | 13 +++++++++
.../linux/wayland/base_capturer_pipewire.cc | 11 +-------
.../linux/wayland/screencast_portal.cc | 2 ++
5 files changed, 45 insertions(+), 11 deletions(-)
modules/desktop_capture/desktop_capturer.cc | 7 +++----
modules/desktop_capture/desktop_capturer.h | 4 ----
2 files changed, 3 insertions(+), 8 deletions(-)
diff --git a/modules/desktop_capture/desktop_capture_types.h b/modules/desktop_capture/desktop_capture_types.h
index 381d1021c4..e777a45f92 100644
--- a/modules/desktop_capture/desktop_capture_types.h
+++ b/modules/desktop_capture/desktop_capture_types.h
@@ -19,7 +19,7 @@ typedef int pid_t; // matching what used to be in
namespace webrtc {
-enum class CaptureType { kWindow, kScreen };
+enum class CaptureType { kWindow, kScreen, kAnyScreenContent };
// Type used to identify windows on the desktop. Values are platform-specific:
// - On Windows: HWND cast to intptr_t.
diff --git a/modules/desktop_capture/desktop_capturer.cc b/modules/desktop_capture/desktop_capturer.cc
index 4debc75394..65a4045f8c 100644
index 65a4045f8c..042d9f7633 100644
--- a/modules/desktop_capture/desktop_capturer.cc
+++ b/modules/desktop_capture/desktop_capturer.cc
@@ -25,6 +25,10 @@
@@ -25,7 +25,7 @@
#include "rtc_base/win/windows_version.h"
#endif // defined(RTC_ENABLE_WIN_WGC)
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+#include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
+#endif
+
namespace webrtc {
-#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+#if defined(WEBRTC_USE_PIPEWIRE)
#include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
#endif
DesktopCapturer::~DesktopCapturer() = default;
@@ -78,6 +82,30 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateWindowCapturer(
@@ -82,7 +82,6 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateWindowCapturer(
return capturer;
}
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateGenericCapturer(
+ const DesktopCaptureOptions& options) {
+ std::unique_ptr<DesktopCapturer> capturer = CreateRawGenericCapturer(options);
+ if (capturer && options.detect_updated_region()) {
+ capturer.reset(new DesktopCapturerDifferWrapper(std::move(capturer)));
+ }
+
+ return capturer;
+}
+
+// static
+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawGenericCapturer(
+ const DesktopCaptureOptions& options) {
+ if (options.allow_pipewire() && DesktopCapturer::IsRunningUnderWayland()) {
+ return std::make_unique<BaseCapturerPipeWire>(options,
+ CaptureType::kAnyScreenContent);
+ }
+
+ return nullptr;
+}
+#endif // defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+
-#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
// static
std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateGenericCapturer(
const DesktopCaptureOptions& options) {
@@ -94,17 +93,17 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateGenericCapturer(
return capturer;
}
-// static
std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawGenericCapturer(
const DesktopCaptureOptions& options) {
+#if defined(WEBRTC_USE_PIPEWIRE)
if (options.allow_pipewire() && DesktopCapturer::IsRunningUnderWayland()) {
return std::make_unique<BaseCapturerPipeWire>(options,
CaptureType::kAnyScreenContent);
}
+#endif // defined(WEBRTC_USE_PIPEWIRE)
return nullptr;
}
-#endif // defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
// static
std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
const DesktopCaptureOptions& options) {
diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
index 3fdcd2c5d0..2713c0e603 100644
index 2713c0e603..99a3606a05 100644
--- a/modules/desktop_capture/desktop_capturer.h
+++ b/modules/desktop_capture/desktop_capturer.h
@@ -168,6 +168,12 @@ class RTC_EXPORT DesktopCapturer {
@@ -168,11 +168,9 @@ class RTC_EXPORT DesktopCapturer {
// The return value if `pos` is out of the scope of the source is undefined.
virtual bool IsOccluded(const DesktopVector& pos);
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+ // Creates a DesktopCapturer instance which targets to capture windows and screens.
+ static std::unique_ptr<DesktopCapturer> CreateGenericCapturer(
+ const DesktopCaptureOptions& options);
+#endif
+
-#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
// Creates a DesktopCapturer instance which targets to capture windows and screens.
static std::unique_ptr<DesktopCapturer> CreateGenericCapturer(
const DesktopCaptureOptions& options);
-#endif
// Creates a DesktopCapturer instance which targets to capture windows.
static std::unique_ptr<DesktopCapturer> CreateWindowCapturer(
const DesktopCaptureOptions& options);
@@ -196,6 +202,13 @@ class RTC_EXPORT DesktopCapturer {
@@ -202,12 +200,10 @@ class RTC_EXPORT DesktopCapturer {
// CroppingWindowCapturer needs to create raw capturers without wrappers, so
// the following two functions are protected.
+#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+ // Creates a platform specific DesktopCapturer instance which targets to
+ // capture windows and screens.
+ static std::unique_ptr<DesktopCapturer> CreateRawGenericCapturer(
+ const DesktopCaptureOptions& options);
+#endif
+
-#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
// Creates a platform specific DesktopCapturer instance which targets to
// capture windows and screens.
static std::unique_ptr<DesktopCapturer> CreateRawGenericCapturer(
const DesktopCaptureOptions& options);
-#endif
// Creates a platform specific DesktopCapturer instance which targets to
// capture windows.
static std::unique_ptr<DesktopCapturer> CreateRawWindowCapturer(
diff --git a/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc b/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc
index dae2b70510..cf4f7dc9aa 100644
--- a/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc
+++ b/modules/desktop_capture/linux/wayland/base_capturer_pipewire.cc
@@ -165,15 +165,6 @@ void BaseCapturerPipeWire::CaptureFrame() {
callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
}
-// Keep in sync with defines at browser/actors/WebRTCParent.jsm
-// With PipeWire we can't select which system resource is shared so
-// we don't create a window/screen list. Instead we place these constants
-// as window name/id so frontend code can identify PipeWire backend
-// and does not try to create screen/window preview.
-
-#define PIPEWIRE_ID 0xaffffff
-#define PIPEWIRE_NAME "####_PIPEWIRE_PORTAL_####"
-
bool BaseCapturerPipeWire::GetSourceList(SourceList* sources) {
RTC_DCHECK(sources->size() == 0);
// List of available screens is already presented by the xdg-desktop-portal,
@@ -190,7 +181,7 @@ bool BaseCapturerPipeWire::GetSourceList(SourceList* sources) {
bool BaseCapturerPipeWire::SelectSource(SourceId id) {
// Screen selection is handled by the xdg-desktop-portal.
selected_source_id_ = id;
- return id == PIPEWIRE_ID;
+ return true;
}
DelegatedSourceListController*
diff --git a/modules/desktop_capture/linux/wayland/screencast_portal.cc b/modules/desktop_capture/linux/wayland/screencast_portal.cc
index abfade56e7..e7aaee001b 100644
--- a/modules/desktop_capture/linux/wayland/screencast_portal.cc
+++ b/modules/desktop_capture/linux/wayland/screencast_portal.cc
@@ -41,6 +41,8 @@ ScreenCastPortal::CaptureSourceType ScreenCastPortal::ToCaptureSourceType(
return ScreenCastPortal::CaptureSourceType::kScreen;
case CaptureType::kWindow:
return ScreenCastPortal::CaptureSourceType::kWindow;
+ case CaptureType::kAnyScreenContent:
+ return ScreenCastPortal::CaptureSourceType::kAnyScreenContent;
}
}
--
2.34.1

116
third_party/libwebrtc/moz-patch-stack/0088.patch поставляемый
Просмотреть файл

@ -1,100 +1,32 @@
From: Jan Grulich <jgrulich@redhat.com>
Date: Mon, 27 Feb 2023 13:57:00 +0000
Subject: Bug 1819044 - fix build non-pipewire builds
r=webrtc-reviewers,pehrsons
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 6 Mar 2023 07:41:00 +0000
Subject: Bug 1817724 - Crash in case VideoCaptureAVFoundation is destroyed
unexpectedly. r=karlt,webrtc-reviewers,ng
We should check only for PipeWire presence when building code specific
to PipeWire.
Differential Revision: https://phabricator.services.mozilla.com/D171071
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7a879ad084a6e9768479c73cc5c3f4e9d95a2ab9
Also includes:
Bug 1819044 - fix build non-pipewire builds (attempt #2) r=webrtc-reviewers,pehrsons
Make the new API available to everyone and just return an empty capturer
in case when building without PipeWire. It will not make any difference
because using X11 based capturers on Wayland is useless anyway so if we
fail for missing PipeWire on Wayland, it will have the same outcome.
Differential Revision: https://phabricator.services.mozilla.com/D171192
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/ad247b0aac896d884eba5e40f0ec8a9f50d8b85b
Differential Revision: https://phabricator.services.mozilla.com/D171470
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/30340541c9a17e21f1c8f7e65850afaa5e3355db
---
modules/desktop_capture/desktop_capturer.cc | 7 +++----
modules/desktop_capture/desktop_capturer.h | 4 ----
2 files changed, 3 insertions(+), 8 deletions(-)
modules/video_capture/video_capture.h | 5 +++++
1 file changed, 5 insertions(+)
diff --git a/modules/desktop_capture/desktop_capturer.cc b/modules/desktop_capture/desktop_capturer.cc
index 65a4045f8c..042d9f7633 100644
--- a/modules/desktop_capture/desktop_capturer.cc
+++ b/modules/desktop_capture/desktop_capturer.cc
@@ -25,7 +25,7 @@
#include "rtc_base/win/windows_version.h"
#endif // defined(RTC_ENABLE_WIN_WGC)
diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
index 7e181c538e..4c800ccb6a 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -161,8 +161,13 @@ class VideoCaptureModule : public rtc::RefCountInterface {
// Mozilla: TrackingId setter for use in profiler markers.
virtual void SetTrackingId(uint32_t aTrackingIdProcId) {}
-#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
+#if defined(WEBRTC_USE_PIPEWIRE)
#include "modules/desktop_capture/linux/wayland/base_capturer_pipewire.h"
#endif
+ // Mozilla: Diagnostic
+ void NotifyReleasing() { mOkToDestroy = true; }
+
protected:
~VideoCaptureModule() override {}
+
+ std::atomic<bool> mOkToDestroy = {false};
};
@@ -82,7 +82,6 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateWindowCapturer(
return capturer;
}
-#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
// static
std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateGenericCapturer(
const DesktopCaptureOptions& options) {
@@ -94,17 +93,17 @@ std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateGenericCapturer(
return capturer;
}
-// static
std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawGenericCapturer(
const DesktopCaptureOptions& options) {
+#if defined(WEBRTC_USE_PIPEWIRE)
if (options.allow_pipewire() && DesktopCapturer::IsRunningUnderWayland()) {
return std::make_unique<BaseCapturerPipeWire>(options,
CaptureType::kAnyScreenContent);
}
+#endif // defined(WEBRTC_USE_PIPEWIRE)
return nullptr;
}
-#endif // defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
// static
std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateScreenCapturer(
diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
index 2713c0e603..99a3606a05 100644
--- a/modules/desktop_capture/desktop_capturer.h
+++ b/modules/desktop_capture/desktop_capturer.h
@@ -168,11 +168,9 @@ class RTC_EXPORT DesktopCapturer {
// The return value if `pos` is out of the scope of the source is undefined.
virtual bool IsOccluded(const DesktopVector& pos);
-#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
// Creates a DesktopCapturer instance which targets to capture windows and screens.
static std::unique_ptr<DesktopCapturer> CreateGenericCapturer(
const DesktopCaptureOptions& options);
-#endif
// Creates a DesktopCapturer instance which targets to capture windows.
static std::unique_ptr<DesktopCapturer> CreateWindowCapturer(
@@ -202,12 +200,10 @@ class RTC_EXPORT DesktopCapturer {
// CroppingWindowCapturer needs to create raw capturers without wrappers, so
// the following two functions are protected.
-#if defined(WEBRTC_USE_PIPEWIRE) || defined(WEBRTC_USE_X11)
// Creates a platform specific DesktopCapturer instance which targets to
// capture windows and screens.
static std::unique_ptr<DesktopCapturer> CreateRawGenericCapturer(
const DesktopCaptureOptions& options);
-#endif
// Creates a platform specific DesktopCapturer instance which targets to
// capture windows.
} // namespace webrtc
--
2.34.1

89
third_party/libwebrtc/moz-patch-stack/0089.patch поставляемый
Просмотреть файл

@ -1,32 +1,77 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 6 Mar 2023 07:41:00 +0000
Subject: Bug 1817724 - Crash in case VideoCaptureAVFoundation is destroyed
unexpectedly. r=karlt,webrtc-reviewers,ng
From: Jan Grulich <jgrulich@redhat.com>
Date: Fri, 10 Mar 2023 09:21:00 +0000
Subject: Bug 1819035 - get EGL display based on the used platform in the
browser r=webrtc-reviewers,ng
Differential Revision: https://phabricator.services.mozilla.com/D171470
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/30340541c9a17e21f1c8f7e65850afaa5e3355db
Because of a possible misconfiguration or a possible driver issue it
might happen that the browser will use a different driver on X11 and
end up using yet another one for wayland/gbm, which might lead to not
working screen sharing in the better case, but also to a crash in the
other driver (Nvidia). This adds a check for platform the browser runs
on, if it's XWayland or Wayland and based on that query EGL display for
that specific platform, rather than going for the Wayland one only.
Differential Revision: https://phabricator.services.mozilla.com/D171858
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c8606497de1f461a6352456e0e511c2ae498d526
---
modules/video_capture/video_capture.h | 5 +++++
1 file changed, 5 insertions(+)
.../linux/wayland/egl_dmabuf.cc | 30 +++++++++++++++++--
1 file changed, 28 insertions(+), 2 deletions(-)
diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
index 7e181c538e..4c800ccb6a 100644
--- a/modules/video_capture/video_capture.h
+++ b/modules/video_capture/video_capture.h
@@ -161,8 +161,13 @@ class VideoCaptureModule : public rtc::RefCountInterface {
// Mozilla: TrackingId setter for use in profiler markers.
virtual void SetTrackingId(uint32_t aTrackingIdProcId) {}
diff --git a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
index 5bbd5d7aba..80da597e29 100644
--- a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
+++ b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
@@ -13,6 +13,7 @@
#include <asm/ioctl.h>
#include <dlfcn.h>
#include <fcntl.h>
+#include <gdk/gdk.h>
#include <libdrm/drm_fourcc.h>
#include <linux/types.h>
#include <spa/param/video/format-utils.h>
@@ -200,6 +201,26 @@ static void CloseLibrary(void* library) {
}
}
+ // Mozilla: Diagnostic
+ void NotifyReleasing() { mOkToDestroy = true; }
+static bool IsWaylandDisplay() {
+ static auto sGdkWaylandDisplayGetType =
+ (GType (*)(void))dlsym(RTLD_DEFAULT, "gdk_wayland_display_get_type");
+ if (!sGdkWaylandDisplayGetType) {
+ return false;
+ }
+ return (G_TYPE_CHECK_INSTANCE_TYPE ((gdk_display_get_default()),
+ sGdkWaylandDisplayGetType()));
+}
+
protected:
~VideoCaptureModule() override {}
+static bool IsX11Display() {
+ static auto sGdkX11DisplayGetType =
+ (GType (*)(void))dlsym(RTLD_DEFAULT, "gdk_x11_display_get_type");
+ if (!sGdkX11DisplayGetType) {
+ return false;
+ }
+ return (G_TYPE_CHECK_INSTANCE_TYPE ((gdk_display_get_default()),
+ sGdkX11DisplayGetType()));
+}
+
+ std::atomic<bool> mOkToDestroy = {false};
};
static void* g_lib_egl = nullptr;
} // namespace webrtc
RTC_NO_SANITIZE("cfi-icall")
@@ -331,8 +352,13 @@ EglDmaBuf::EglDmaBuf() {
return;
}
- egl_.display = EglGetPlatformDisplay(EGL_PLATFORM_WAYLAND_KHR,
- (void*)EGL_DEFAULT_DISPLAY, nullptr);
+ if (IsWaylandDisplay()) {
+ egl_.display = EglGetPlatformDisplay(EGL_PLATFORM_WAYLAND_KHR,
+ (void*)EGL_DEFAULT_DISPLAY, nullptr);
+ } else if (IsX11Display()) {
+ egl_.display = EglGetPlatformDisplay(EGL_PLATFORM_X11_KHR,
+ (void*)EGL_DEFAULT_DISPLAY, nullptr);
+ }
if (egl_.display == EGL_NO_DISPLAY) {
RTC_LOG(LS_ERROR) << "Failed to obtain default EGL display: "
--
2.34.1

97
third_party/libwebrtc/moz-patch-stack/0090.patch поставляемый
Просмотреть файл

@ -1,77 +1,34 @@
From: Jan Grulich <jgrulich@redhat.com>
Date: Fri, 10 Mar 2023 09:21:00 +0000
Subject: Bug 1819035 - get EGL display based on the used platform in the
browser r=webrtc-reviewers,ng
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 27 Feb 2023 16:22:00 +0000
Subject: Bug 1817024 - (fix-fdcfefa708) In PhysicalSocket avoid a non-trivial
designated initializer. r=mjf,webrtc-reviewers
Because of a possible misconfiguration or a possible driver issue it
might happen that the browser will use a different driver on X11 and
end up using yet another one for wayland/gbm, which might lead to not
working screen sharing in the better case, but also to a crash in the
other driver (Nvidia). This adds a check for platform the browser runs
on, if it's XWayland or Wayland and based on that query EGL display for
that specific platform, rather than going for the Wayland one only.
This fixes a build failure in the base-toolchain job with GCC 7.5.0:
In file included from Unified_cpp_threading_gn0.cpp:38:0:
.../third_party/libwebrtc/rtc_base/physical_socket_server.cc: In member function 'int rtc::PhysicalSocket::DoReadFromSocket(void*, size_t, rtc::SocketAddress*, int64_t*)':
.../third_party/libwebrtc/rtc_base/physical_socket_server.cc:463:51: sorry, unimplemented: non-trivial designated initializers not supported
msghdr msg = {.msg_iov = &iov, .msg_iovlen = 1};
^
Differential Revision: https://phabricator.services.mozilla.com/D171858
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c8606497de1f461a6352456e0e511c2ae498d526
Differential Revision: https://phabricator.services.mozilla.com/D171057
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a3447f709befd84a282ca40f29b7a5ea76d5b68d
---
.../linux/wayland/egl_dmabuf.cc | 30 +++++++++++++++++--
1 file changed, 28 insertions(+), 2 deletions(-)
rtc_base/physical_socket_server.cc | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
index 5bbd5d7aba..80da597e29 100644
--- a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
+++ b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
@@ -13,6 +13,7 @@
#include <asm/ioctl.h>
#include <dlfcn.h>
#include <fcntl.h>
+#include <gdk/gdk.h>
#include <libdrm/drm_fourcc.h>
#include <linux/types.h>
#include <spa/param/video/format-utils.h>
@@ -200,6 +201,26 @@ static void CloseLibrary(void* library) {
}
}
+static bool IsWaylandDisplay() {
+ static auto sGdkWaylandDisplayGetType =
+ (GType (*)(void))dlsym(RTLD_DEFAULT, "gdk_wayland_display_get_type");
+ if (!sGdkWaylandDisplayGetType) {
+ return false;
+ }
+ return (G_TYPE_CHECK_INSTANCE_TYPE ((gdk_display_get_default()),
+ sGdkWaylandDisplayGetType()));
+}
+
+static bool IsX11Display() {
+ static auto sGdkX11DisplayGetType =
+ (GType (*)(void))dlsym(RTLD_DEFAULT, "gdk_x11_display_get_type");
+ if (!sGdkX11DisplayGetType) {
+ return false;
+ }
+ return (G_TYPE_CHECK_INSTANCE_TYPE ((gdk_display_get_default()),
+ sGdkX11DisplayGetType()));
+}
+
static void* g_lib_egl = nullptr;
RTC_NO_SANITIZE("cfi-icall")
@@ -331,8 +352,13 @@ EglDmaBuf::EglDmaBuf() {
return;
}
- egl_.display = EglGetPlatformDisplay(EGL_PLATFORM_WAYLAND_KHR,
- (void*)EGL_DEFAULT_DISPLAY, nullptr);
+ if (IsWaylandDisplay()) {
+ egl_.display = EglGetPlatformDisplay(EGL_PLATFORM_WAYLAND_KHR,
+ (void*)EGL_DEFAULT_DISPLAY, nullptr);
+ } else if (IsX11Display()) {
+ egl_.display = EglGetPlatformDisplay(EGL_PLATFORM_X11_KHR,
+ (void*)EGL_DEFAULT_DISPLAY, nullptr);
+ }
if (egl_.display == EGL_NO_DISPLAY) {
RTC_LOG(LS_ERROR) << "Failed to obtain default EGL display: "
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index 60d024c769..3b112e6188 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -460,7 +460,7 @@ int PhysicalSocket::DoReadFromSocket(void* buffer,
int received = 0;
if (read_scm_timestamp_experiment_) {
iovec iov = {.iov_base = buffer, .iov_len = length};
- msghdr msg = {.msg_iov = &iov, .msg_iovlen = 1};
+ msghdr msg = {.msg_name = nullptr, .msg_namelen = 0, .msg_iov = &iov, .msg_iovlen = 1};
if (out_addr) {
out_addr->Clear();
msg.msg_name = addr;
--
2.34.1

546
third_party/libwebrtc/moz-patch-stack/0091.patch поставляемый
Просмотреть файл

@ -1,34 +1,526 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Mon, 27 Feb 2023 16:22:00 +0000
Subject: Bug 1817024 - (fix-fdcfefa708) In PhysicalSocket avoid a non-trivial
designated initializer. r=mjf,webrtc-reviewers
From: Jan Grulich <jgrulich@redhat.com>
Date: Tue, 28 Mar 2023 14:41:00 +0000
Subject: Bug 1823404 - PipeWire capturer: import DMABufs directly into desktop
frame r=webrtc-reviewers,stransky,bwc
This fixes a build failure in the base-toolchain job with GCC 7.5.0:
In file included from Unified_cpp_threading_gn0.cpp:38:0:
.../third_party/libwebrtc/rtc_base/physical_socket_server.cc: In member function 'int rtc::PhysicalSocket::DoReadFromSocket(void*, size_t, rtc::SocketAddress*, int64_t*)':
.../third_party/libwebrtc/rtc_base/physical_socket_server.cc:463:51: sorry, unimplemented: non-trivial designated initializers not supported
msghdr msg = {.msg_iov = &iov, .msg_iovlen = 1};
^
Originally DMABufs were imported into a temporary buffer followed by a
copy operation into the desktop frame itself. This is not needed as we
can import them directly into desktop frames and avoid this overhead.
Differential Revision: https://phabricator.services.mozilla.com/D171057
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a3447f709befd84a282ca40f29b7a5ea76d5b68d
Also drop support for MemPtr buffers as both Mutter and KWin don't seem
to support them and they are going to be too slow anyway.
Testing with latest Chromium, I could see two processes with usage
around 20% and 40% without this change going down to 10% and 20% with
this change applied.
Also drop old DmaBuf support.
Differential Revision: https://phabricator.services.mozilla.com/D173021
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/581fe5ce66f9f3c725f5345b3e57407d1ec1e312
---
rtc_base/physical_socket_server.cc | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
.../linux/wayland/egl_dmabuf.cc | 114 +++++++---
.../linux/wayland/egl_dmabuf.h | 16 +-
.../linux/wayland/shared_screencast_stream.cc | 209 +++++++++---------
3 files changed, 201 insertions(+), 138 deletions(-)
diff --git a/rtc_base/physical_socket_server.cc b/rtc_base/physical_socket_server.cc
index 60d024c769..3b112e6188 100644
--- a/rtc_base/physical_socket_server.cc
+++ b/rtc_base/physical_socket_server.cc
@@ -460,7 +460,7 @@ int PhysicalSocket::DoReadFromSocket(void* buffer,
int received = 0;
if (read_scm_timestamp_experiment_) {
iovec iov = {.iov_base = buffer, .iov_len = length};
- msghdr msg = {.msg_iov = &iov, .msg_iovlen = 1};
+ msghdr msg = {.msg_name = nullptr, .msg_namelen = 0, .msg_iov = &iov, .msg_iovlen = 1};
if (out_addr) {
out_addr->Clear();
msg.msg_name = addr;
diff --git a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
index 80da597e29..6a019c64b4 100644
--- a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
+++ b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
@@ -102,11 +102,23 @@ typedef void (*glDeleteTextures_func)(GLsizei n, const GLuint* textures);
typedef void (*glGenTextures_func)(GLsizei n, GLuint* textures);
typedef GLenum (*glGetError_func)(void);
typedef const GLubyte* (*glGetString_func)(GLenum name);
-typedef void (*glGetTexImage_func)(GLenum target,
- GLint level,
- GLenum format,
- GLenum type,
- void* pixels);
+typedef void (*glReadPixels_func)(GLint x,
+ GLint y,
+ GLsizei width,
+ GLsizei height,
+ GLenum format,
+ GLenum type,
+ void* data);
+typedef void (*glGenFramebuffers_func)(GLsizei n, GLuint* ids);
+typedef void (*glDeleteFramebuffers_func)(GLsizei n,
+ const GLuint* framebuffers);
+typedef void (*glBindFramebuffer_func)(GLenum target, GLuint framebuffer);
+typedef void (*glFramebufferTexture2D_func)(GLenum target,
+ GLenum attachment,
+ GLenum textarget,
+ GLuint texture,
+ GLint level);
+typedef GLenum (*glCheckFramebufferStatus_func)(GLenum target);
typedef void (*glTexParameteri_func)(GLenum target, GLenum pname, GLint param);
typedef void* (*glXGetProcAddressARB_func)(const char*);
@@ -119,7 +131,12 @@ glDeleteTextures_func GlDeleteTextures = nullptr;
glGenTextures_func GlGenTextures = nullptr;
glGetError_func GlGetError = nullptr;
glGetString_func GlGetString = nullptr;
-glGetTexImage_func GlGetTexImage = nullptr;
+glReadPixels_func GlReadPixels = nullptr;
+glGenFramebuffers_func GlGenFramebuffers = nullptr;
+glDeleteFramebuffers_func GlDeleteFramebuffers = nullptr;
+glBindFramebuffer_func GlBindFramebuffer = nullptr;
+glFramebufferTexture2D_func GlFramebufferTexture2D = nullptr;
+glCheckFramebufferStatus_func GlCheckFramebufferStatus = nullptr;
glTexParameteri_func GlTexParameteri = nullptr;
glXGetProcAddressARB_func GlXGetProcAddressARB = nullptr;
@@ -300,12 +317,26 @@ static bool LoadGL() {
(glDeleteTextures_func)GlXGetProcAddressARB("glDeleteTextures");
GlGenTextures = (glGenTextures_func)GlXGetProcAddressARB("glGenTextures");
GlGetError = (glGetError_func)GlXGetProcAddressARB("glGetError");
- GlGetTexImage = (glGetTexImage_func)GlXGetProcAddressARB("glGetTexImage");
+ GlReadPixels = (glReadPixels_func)GlXGetProcAddressARB("glReadPixels");
+ GlGenFramebuffers =
+ (glGenFramebuffers_func)GlXGetProcAddressARB("glGenFramebuffers");
+ GlDeleteFramebuffers =
+ (glDeleteFramebuffers_func)GlXGetProcAddressARB("glDeleteFramebuffers");
+ GlBindFramebuffer =
+ (glBindFramebuffer_func)GlXGetProcAddressARB("glBindFramebuffer");
+ GlFramebufferTexture2D = (glFramebufferTexture2D_func)GlXGetProcAddressARB(
+ "glFramebufferTexture2D");
+ GlCheckFramebufferStatus =
+ (glCheckFramebufferStatus_func)GlXGetProcAddressARB(
+ "glCheckFramebufferStatus");
+
GlTexParameteri =
(glTexParameteri_func)GlXGetProcAddressARB("glTexParameteri");
return GlBindTexture && GlDeleteTextures && GlGenTextures && GlGetError &&
- GlGetTexImage && GlTexParameteri;
+ GlReadPixels && GlGenFramebuffers && GlDeleteFramebuffers &&
+ GlBindFramebuffer && GlFramebufferTexture2D &&
+ GlCheckFramebufferStatus && GlTexParameteri;
}
return false;
@@ -461,6 +492,14 @@ EglDmaBuf::~EglDmaBuf() {
EglTerminate(egl_.display);
}
+ if (fbo_) {
+ GlDeleteFramebuffers(1, &fbo_);
+ }
+
+ if (texture_) {
+ GlDeleteTextures(1, &texture_);
+ }
+
// BUG: crbug.com/1290566
// Closing libEGL.so.1 when using NVidia drivers causes a crash
// when EglGetPlatformDisplayEXT() is used, at least this one is enough
@@ -492,20 +531,20 @@ bool EglDmaBuf::GetClientExtensions(EGLDisplay dpy, EGLint name) {
}
RTC_NO_SANITIZE("cfi-icall")
-std::unique_ptr<uint8_t[]> EglDmaBuf::ImageFromDmaBuf(
- const DesktopSize& size,
- uint32_t format,
- const std::vector<PlaneData>& plane_datas,
- uint64_t modifier) {
- std::unique_ptr<uint8_t[]> src;
-
+bool EglDmaBuf::ImageFromDmaBuf(const DesktopSize& size,
+ uint32_t format,
+ const std::vector<PlaneData>& plane_datas,
+ uint64_t modifier,
+ const DesktopVector& offset,
+ const DesktopSize& buffer_size,
+ uint8_t* data) {
if (!egl_initialized_) {
- return src;
+ return false;
}
if (plane_datas.size() <= 0) {
RTC_LOG(LS_ERROR) << "Failed to process buffer: invalid number of planes";
- return src;
+ return false;
}
EGLint attribs[47];
@@ -594,20 +633,32 @@ std::unique_ptr<uint8_t[]> EglDmaBuf::ImageFromDmaBuf(
if (image == EGL_NO_IMAGE) {
RTC_LOG(LS_ERROR) << "Failed to record frame: Error creating EGLImage - "
<< FormatEGLError(EglGetError());
- return src;
+ return false;
}
// create GL 2D texture for framebuffer
- GLuint texture;
- GlGenTextures(1, &texture);
- GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
- GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_NEAREST);
- GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
- GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
- GlBindTexture(GL_TEXTURE_2D, texture);
+ if (!texture_) {
+ GlGenTextures(1, &texture_);
+ GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
+ GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_NEAREST);
+ GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+ }
+ GlBindTexture(GL_TEXTURE_2D, texture_);
GlEGLImageTargetTexture2DOES(GL_TEXTURE_2D, image);
- src = std::make_unique<uint8_t[]>(plane_datas[0].stride * size.height());
+ if (!fbo_) {
+ GlGenFramebuffers(1, &fbo_);
+ }
+
+ GlBindFramebuffer(GL_FRAMEBUFFER, fbo_);
+ GlFramebufferTexture2D(GL_FRAMEBUFFER, GL_COLOR_ATTACHMENT0, GL_TEXTURE_2D,
+ texture_, 0);
+ if (GlCheckFramebufferStatus(GL_FRAMEBUFFER) != GL_FRAMEBUFFER_COMPLETE) {
+ RTC_LOG(LS_ERROR) << "Failed to bind DMA buf framebuffer";
+ EglDestroyImageKHR(egl_.display, image);
+ return false;
+ }
GLenum gl_format = GL_BGRA;
switch (format) {
@@ -624,17 +675,18 @@ std::unique_ptr<uint8_t[]> EglDmaBuf::ImageFromDmaBuf(
gl_format = GL_BGRA;
break;
}
- GlGetTexImage(GL_TEXTURE_2D, 0, gl_format, GL_UNSIGNED_BYTE, src.get());
- if (GlGetError()) {
+ GlReadPixels(offset.x(), offset.y(), buffer_size.width(),
+ buffer_size.height(), gl_format, GL_UNSIGNED_BYTE, data);
+
+ const GLenum error = GlGetError();
+ if (error) {
RTC_LOG(LS_ERROR) << "Failed to get image from DMA buffer.";
- return src;
}
- GlDeleteTextures(1, &texture);
EglDestroyImageKHR(egl_.display, image);
- return src;
+ return !error;
}
RTC_NO_SANITIZE("cfi-icall")
diff --git a/modules/desktop_capture/linux/wayland/egl_dmabuf.h b/modules/desktop_capture/linux/wayland/egl_dmabuf.h
index f1d96b2f80..22a8f5ab52 100644
--- a/modules/desktop_capture/linux/wayland/egl_dmabuf.h
+++ b/modules/desktop_capture/linux/wayland/egl_dmabuf.h
@@ -41,11 +41,15 @@ class EglDmaBuf {
EglDmaBuf();
~EglDmaBuf();
- std::unique_ptr<uint8_t[]> ImageFromDmaBuf(
- const DesktopSize& size,
- uint32_t format,
- const std::vector<PlaneData>& plane_datas,
- uint64_t modifiers);
+ // Returns whether the image was successfully imported from
+ // given DmaBuf and its parameters
+ bool ImageFromDmaBuf(const DesktopSize& size,
+ uint32_t format,
+ const std::vector<PlaneData>& plane_datas,
+ uint64_t modifiers,
+ const DesktopVector& offset,
+ const DesktopSize& buffer_size,
+ uint8_t* data);
std::vector<uint64_t> QueryDmaBufModifiers(uint32_t format);
bool IsEglInitialized() const { return egl_initialized_; }
@@ -58,6 +62,8 @@ class EglDmaBuf {
int32_t drm_fd_ = -1; // for GBM buffer mmap
gbm_device* gbm_device_ = nullptr; // for passed GBM buffer retrieval
+ GLuint fbo_ = 0;
+ GLuint texture_ = 0;
EGLStruct egl_;
absl::optional<std::string> GetRenderNode();
diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
index 71bde9b212..bcd7e3a33c 100644
--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
@@ -38,7 +38,6 @@ constexpr int CursorMetaSize(int w, int h) {
w * h * kCursorBpp);
}
-constexpr PipeWireVersion kDmaBufMinVersion = {0, 3, 24};
constexpr PipeWireVersion kDmaBufModifierMinVersion = {0, 3, 33};
constexpr PipeWireVersion kDropSingleModifierMinVersion = {0, 3, 40};
@@ -155,6 +154,12 @@ class SharedScreenCastStreamPrivate {
struct spa_video_info_raw spa_video_format_;
void ProcessBuffer(pw_buffer* buffer);
+ bool ProcessMemFDBuffer(pw_buffer* buffer,
+ DesktopFrame& frame,
+ const DesktopVector& offset);
+ bool ProcessDMABuffer(pw_buffer* buffer,
+ DesktopFrame& frame,
+ const DesktopVector& offset);
void ConvertRGBxToBGRx(uint8_t* frame, uint32_t size);
// PipeWire callbacks
@@ -277,10 +282,9 @@ void SharedScreenCastStreamPrivate::OnStreamParamChanged(
has_modifier ? that->spa_video_format_.modifier : DRM_FORMAT_MOD_INVALID;
std::vector<const spa_pod*> params;
const int buffer_types =
- has_modifier || (that->pw_server_version_ >= kDmaBufMinVersion)
- ? (1 << SPA_DATA_DmaBuf) | (1 << SPA_DATA_MemFd) |
- (1 << SPA_DATA_MemPtr)
- : (1 << SPA_DATA_MemFd) | (1 << SPA_DATA_MemPtr);
+ has_modifier
+ ? (1 << SPA_DATA_DmaBuf) | (1 << SPA_DATA_MemFd)
+ : (1 << SPA_DATA_MemFd);
params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
&builder, SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
@@ -613,9 +617,6 @@ DesktopVector SharedScreenCastStreamPrivate::CaptureCursorPosition() {
RTC_NO_SANITIZE("cfi-icall")
void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
spa_buffer* spa_buffer = buffer->buffer;
- ScopedBuf map;
- std::unique_ptr<uint8_t[]> src_unique_ptr;
- uint8_t* src = nullptr;
// Try to update the mouse cursor first, because it can be the only
// information carried by the buffer
@@ -657,79 +658,6 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
return;
}
- if (spa_buffer->datas[0].type == SPA_DATA_MemFd) {
- map.initialize(
- static_cast<uint8_t*>(
- mmap(nullptr,
- spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
- PROT_READ, MAP_PRIVATE, spa_buffer->datas[0].fd, 0)),
- spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
- spa_buffer->datas[0].fd);
-
- if (!map) {
- RTC_LOG(LS_ERROR) << "Failed to mmap the memory: "
- << std::strerror(errno);
- return;
- }
-
- src = SPA_MEMBER(map.get(), spa_buffer->datas[0].mapoffset, uint8_t);
- } else if (spa_buffer->datas[0].type == SPA_DATA_DmaBuf) {
- const uint n_planes = spa_buffer->n_datas;
-
- if (!n_planes) {
- return;
- }
-
- std::vector<EglDmaBuf::PlaneData> plane_datas;
- for (uint32_t i = 0; i < n_planes; ++i) {
- EglDmaBuf::PlaneData data = {
- static_cast<int32_t>(spa_buffer->datas[i].fd),
- static_cast<uint32_t>(spa_buffer->datas[i].chunk->stride),
- static_cast<uint32_t>(spa_buffer->datas[i].chunk->offset)};
- plane_datas.push_back(data);
- }
-
- // When importing DMA-BUFs, we use the stride (number of bytes from one row
- // of pixels in the buffer) provided by PipeWire. The stride from PipeWire
- // is given by the graphics driver and some drivers might add some
- // additional padding for memory layout optimizations so not everytime the
- // stride is equal to BYTES_PER_PIXEL x WIDTH. This is fine, because during
- // the import we will use OpenGL and same graphics driver so it will be able
- // to work with the stride it provided, but later on when we work with
- // images we get from DMA-BUFs we will need to update the stride to be equal
- // to BYTES_PER_PIXEL x WIDTH as that's the size of the DesktopFrame we
- // allocate for each captured frame.
- src_unique_ptr = egl_dmabuf_->ImageFromDmaBuf(
- stream_size_, spa_video_format_.format, plane_datas, modifier_);
- if (src_unique_ptr) {
- src = src_unique_ptr.get();
- } else {
- RTC_LOG(LS_ERROR) << "Dropping DMA-BUF modifier: " << modifier_
- << " and trying to renegotiate stream parameters";
-
- if (pw_server_version_ >= kDropSingleModifierMinVersion) {
- modifiers_.erase(
- std::remove(modifiers_.begin(), modifiers_.end(), modifier_),
- modifiers_.end());
- } else {
- modifiers_.clear();
- }
-
- pw_loop_signal_event(pw_thread_loop_get_loop(pw_main_loop_),
- renegotiate_);
- return;
- }
- } else if (spa_buffer->datas[0].type == SPA_DATA_MemPtr) {
- src = static_cast<uint8_t*>(spa_buffer->datas[0].data);
- }
-
- if (!src) {
- if (observer_) {
- observer_->OnFailedToProcessBuffer();
- }
- return;
- }
-
// Use SPA_META_VideoCrop metadata to get the frame size. KDE and GNOME do
// handle screen/window sharing differently. KDE/KWin doesn't use
// SPA_META_VideoCrop metadata and when sharing a window, it always sets
@@ -787,8 +715,8 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
}
// Get the position of the video crop within the stream. Just double-check
- // that the position doesn't exceed the size of the stream itself. NOTE:
- // Currently it looks there is no implementation using this.
+ // that the position doesn't exceed the size of the stream itself.
+ // NOTE: Currently it looks there is no implementation using this.
uint32_t y_offset =
videocrop_metadata_use &&
(videocrop_metadata->region.position.y + frame_size_.height() <=
@@ -801,22 +729,7 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
stream_size_.width())
? videocrop_metadata->region.position.x
: 0;
-
- const uint32_t stream_stride = kBytesPerPixel * stream_size_.width();
- uint32_t buffer_stride = spa_buffer->datas[0].chunk->stride;
- uint32_t src_stride = buffer_stride;
-
- if (spa_buffer->datas[0].type == SPA_DATA_DmaBuf &&
- buffer_stride > stream_stride) {
- // When DMA-BUFs are used, sometimes spa_buffer->stride we get might
- // contain additional padding, but after we import the buffer, the stride
- // we used is no longer relevant and we should just calculate it based on
- // the stream width. For more context see https://crbug.com/1333304.
- src_stride = stream_stride;
- }
-
- uint8_t* updated_src =
- src + (src_stride * y_offset) + (kBytesPerPixel * x_offset);
+ DesktopVector offset = DesktopVector(x_offset, y_offset);
webrtc::MutexLock lock(&queue_lock_);
@@ -836,9 +749,20 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(std::move(frame)));
}
- queue_.current_frame()->CopyPixelsFrom(
- updated_src, (src_stride - (kBytesPerPixel * x_offset)),
- DesktopRect::MakeWH(frame_size_.width(), frame_size_.height()));
+ bool bufferProcessed = false;
+ if (spa_buffer->datas[0].type == SPA_DATA_MemFd) {
+ bufferProcessed =
+ ProcessMemFDBuffer(buffer, *queue_.current_frame(), offset);
+ } else if (spa_buffer->datas[0].type == SPA_DATA_DmaBuf) {
+ bufferProcessed = ProcessDMABuffer(buffer, *queue_.current_frame(), offset);
+ }
+
+ if (!bufferProcessed) {
+ if (observer_) {
+ observer_->OnFailedToProcessBuffer();
+ }
+ return;
+ }
if (spa_video_format_.format == SPA_VIDEO_FORMAT_RGBx ||
spa_video_format_.format == SPA_VIDEO_FORMAT_RGBA) {
@@ -885,6 +809,87 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
queue_.current_frame()->set_may_contain_cursor(is_cursor_embedded_);
}
+RTC_NO_SANITIZE("cfi-icall")
+bool SharedScreenCastStreamPrivate::ProcessMemFDBuffer(
+ pw_buffer* buffer,
+ DesktopFrame& frame,
+ const DesktopVector& offset) {
+ spa_buffer* spa_buffer = buffer->buffer;
+ ScopedBuf map;
+ uint8_t* src = nullptr;
+
+ map.initialize(
+ static_cast<uint8_t*>(
+ mmap(nullptr,
+ spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
+ PROT_READ, MAP_PRIVATE, spa_buffer->datas[0].fd, 0)),
+ spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
+ spa_buffer->datas[0].fd);
+
+ if (!map) {
+ RTC_LOG(LS_ERROR) << "Failed to mmap the memory: " << std::strerror(errno);
+ return false;
+ }
+
+ src = SPA_MEMBER(map.get(), spa_buffer->datas[0].mapoffset, uint8_t);
+
+ uint32_t buffer_stride = spa_buffer->datas[0].chunk->stride;
+ uint32_t src_stride = buffer_stride;
+
+ uint8_t* updated_src =
+ src + (src_stride * offset.y()) + (kBytesPerPixel * offset.x());
+
+ frame.CopyPixelsFrom(
+ updated_src, (src_stride - (kBytesPerPixel * offset.x())),
+ DesktopRect::MakeWH(frame.size().width(), frame.size().height()));
+
+ return true;
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+bool SharedScreenCastStreamPrivate::ProcessDMABuffer(
+ pw_buffer* buffer,
+ DesktopFrame& frame,
+ const DesktopVector& offset) {
+ spa_buffer* spa_buffer = buffer->buffer;
+
+ const uint n_planes = spa_buffer->n_datas;
+
+ if (!n_planes) {
+ return false;
+ }
+
+ std::vector<EglDmaBuf::PlaneData> plane_datas;
+ for (uint32_t i = 0; i < n_planes; ++i) {
+ EglDmaBuf::PlaneData data = {
+ static_cast<int32_t>(spa_buffer->datas[i].fd),
+ static_cast<uint32_t>(spa_buffer->datas[i].chunk->stride),
+ static_cast<uint32_t>(spa_buffer->datas[i].chunk->offset)};
+ plane_datas.push_back(data);
+ }
+
+ const bool imported = egl_dmabuf_->ImageFromDmaBuf(
+ stream_size_, spa_video_format_.format, plane_datas, modifier_, offset,
+ frame.size(), frame.data());
+ if (!imported) {
+ RTC_LOG(LS_ERROR) << "Dropping DMA-BUF modifier: " << modifier_
+ << " and trying to renegotiate stream parameters";
+
+ if (pw_server_version_ >= kDropSingleModifierMinVersion) {
+ modifiers_.erase(
+ std::remove(modifiers_.begin(), modifiers_.end(), modifier_),
+ modifiers_.end());
+ } else {
+ modifiers_.clear();
+ }
+
+ pw_loop_signal_event(pw_thread_loop_get_loop(pw_main_loop_), renegotiate_);
+ return false;
+ }
+
+ return true;
+}
+
void SharedScreenCastStreamPrivate::ConvertRGBxToBGRx(uint8_t* frame,
uint32_t size) {
for (uint32_t i = 0; i < size; i += 4) {
--
2.34.1

769
third_party/libwebrtc/moz-patch-stack/0092.patch поставляемый
Просмотреть файл

@ -1,526 +1,269 @@
From: Jan Grulich <jgrulich@redhat.com>
Date: Tue, 28 Mar 2023 14:41:00 +0000
Subject: Bug 1823404 - PipeWire capturer: import DMABufs directly into desktop
frame r=webrtc-reviewers,stransky,bwc
From: Byron Campen <docfaraday@gmail.com>
Date: Fri, 31 Mar 2023 16:21:00 -0500
Subject: Bug 1822194 - (fix-acabb3641b) Break the new SetParametersCallback
stuff into stand-alone files.
Originally DMABufs were imported into a temporary buffer followed by a
copy operation into the desktop frame itself. This is not needed as we
can import them directly into desktop frames and avoid this overhead.
Also drop support for MemPtr buffers as both Mutter and KWin don't seem
to support them and they are going to be too slow anyway.
Testing with latest Chromium, I could see two processes with usage
around 20% and 40% without this change going down to 10% and 20% with
this change applied.
Also drop old DmaBuf support.
Differential Revision: https://phabricator.services.mozilla.com/D173021
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/581fe5ce66f9f3c725f5345b3e57407d1ec1e312
acabb3641b from upstream added a callback mechanism to allow failures to be
propagated back to RTCRtpSender.setParameters. Unfortunately, this callback
mechanism was (needlessly) tightly coupled to libwebrtc's implementation of
RTCRtpSender, and also their media channel code. This introduced a lot of
unnecessary dependencies throughout libwebrtc, that spilled into our code as
well.
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/59232687efa00e5f7b7bd3d6befca129149e2bf5
---
.../linux/wayland/egl_dmabuf.cc | 114 +++++++---
.../linux/wayland/egl_dmabuf.h | 16 +-
.../linux/wayland/shared_screencast_stream.cc | 209 +++++++++---------
3 files changed, 201 insertions(+), 138 deletions(-)
api/BUILD.gn | 14 +++++++++++-
api/rtp_sender_interface.h | 4 ++--
api/rtp_sender_setparameters_callback.cc | 27 +++++++++++++++++++++++
api/rtp_sender_setparameters_callback.h | 28 ++++++++++++++++++++++++
call/BUILD.gn | 4 ++--
call/audio_send_stream.h | 2 +-
call/video_send_stream.h | 2 +-
media/BUILD.gn | 2 +-
media/base/media_channel.h | 4 ----
media/base/media_channel_impl.cc | 13 -----------
video/BUILD.gn | 4 ++--
11 files changed, 77 insertions(+), 27 deletions(-)
create mode 100644 api/rtp_sender_setparameters_callback.cc
create mode 100644 api/rtp_sender_setparameters_callback.h
diff --git a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
index 80da597e29..6a019c64b4 100644
--- a/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
+++ b/modules/desktop_capture/linux/wayland/egl_dmabuf.cc
@@ -102,11 +102,23 @@ typedef void (*glDeleteTextures_func)(GLsizei n, const GLuint* textures);
typedef void (*glGenTextures_func)(GLsizei n, GLuint* textures);
typedef GLenum (*glGetError_func)(void);
typedef const GLubyte* (*glGetString_func)(GLenum name);
-typedef void (*glGetTexImage_func)(GLenum target,
- GLint level,
- GLenum format,
- GLenum type,
- void* pixels);
+typedef void (*glReadPixels_func)(GLint x,
+ GLint y,
+ GLsizei width,
+ GLsizei height,
+ GLenum format,
+ GLenum type,
+ void* data);
+typedef void (*glGenFramebuffers_func)(GLsizei n, GLuint* ids);
+typedef void (*glDeleteFramebuffers_func)(GLsizei n,
+ const GLuint* framebuffers);
+typedef void (*glBindFramebuffer_func)(GLenum target, GLuint framebuffer);
+typedef void (*glFramebufferTexture2D_func)(GLenum target,
+ GLenum attachment,
+ GLenum textarget,
+ GLuint texture,
+ GLint level);
+typedef GLenum (*glCheckFramebufferStatus_func)(GLenum target);
typedef void (*glTexParameteri_func)(GLenum target, GLenum pname, GLint param);
typedef void* (*glXGetProcAddressARB_func)(const char*);
@@ -119,7 +131,12 @@ glDeleteTextures_func GlDeleteTextures = nullptr;
glGenTextures_func GlGenTextures = nullptr;
glGetError_func GlGetError = nullptr;
glGetString_func GlGetString = nullptr;
-glGetTexImage_func GlGetTexImage = nullptr;
+glReadPixels_func GlReadPixels = nullptr;
+glGenFramebuffers_func GlGenFramebuffers = nullptr;
+glDeleteFramebuffers_func GlDeleteFramebuffers = nullptr;
+glBindFramebuffer_func GlBindFramebuffer = nullptr;
+glFramebufferTexture2D_func GlFramebufferTexture2D = nullptr;
+glCheckFramebufferStatus_func GlCheckFramebufferStatus = nullptr;
glTexParameteri_func GlTexParameteri = nullptr;
glXGetProcAddressARB_func GlXGetProcAddressARB = nullptr;
@@ -300,12 +317,26 @@ static bool LoadGL() {
(glDeleteTextures_func)GlXGetProcAddressARB("glDeleteTextures");
GlGenTextures = (glGenTextures_func)GlXGetProcAddressARB("glGenTextures");
GlGetError = (glGetError_func)GlXGetProcAddressARB("glGetError");
- GlGetTexImage = (glGetTexImage_func)GlXGetProcAddressARB("glGetTexImage");
+ GlReadPixels = (glReadPixels_func)GlXGetProcAddressARB("glReadPixels");
+ GlGenFramebuffers =
+ (glGenFramebuffers_func)GlXGetProcAddressARB("glGenFramebuffers");
+ GlDeleteFramebuffers =
+ (glDeleteFramebuffers_func)GlXGetProcAddressARB("glDeleteFramebuffers");
+ GlBindFramebuffer =
+ (glBindFramebuffer_func)GlXGetProcAddressARB("glBindFramebuffer");
+ GlFramebufferTexture2D = (glFramebufferTexture2D_func)GlXGetProcAddressARB(
+ "glFramebufferTexture2D");
+ GlCheckFramebufferStatus =
+ (glCheckFramebufferStatus_func)GlXGetProcAddressARB(
+ "glCheckFramebufferStatus");
+
GlTexParameteri =
(glTexParameteri_func)GlXGetProcAddressARB("glTexParameteri");
return GlBindTexture && GlDeleteTextures && GlGenTextures && GlGetError &&
- GlGetTexImage && GlTexParameteri;
+ GlReadPixels && GlGenFramebuffers && GlDeleteFramebuffers &&
+ GlBindFramebuffer && GlFramebufferTexture2D &&
+ GlCheckFramebufferStatus && GlTexParameteri;
}
return false;
@@ -461,6 +492,14 @@ EglDmaBuf::~EglDmaBuf() {
EglTerminate(egl_.display);
}
+ if (fbo_) {
+ GlDeleteFramebuffers(1, &fbo_);
+ }
+
+ if (texture_) {
+ GlDeleteTextures(1, &texture_);
+ }
+
// BUG: crbug.com/1290566
// Closing libEGL.so.1 when using NVidia drivers causes a crash
// when EglGetPlatformDisplayEXT() is used, at least this one is enough
@@ -492,20 +531,20 @@ bool EglDmaBuf::GetClientExtensions(EGLDisplay dpy, EGLint name) {
}
RTC_NO_SANITIZE("cfi-icall")
-std::unique_ptr<uint8_t[]> EglDmaBuf::ImageFromDmaBuf(
- const DesktopSize& size,
- uint32_t format,
- const std::vector<PlaneData>& plane_datas,
- uint64_t modifier) {
- std::unique_ptr<uint8_t[]> src;
-
+bool EglDmaBuf::ImageFromDmaBuf(const DesktopSize& size,
+ uint32_t format,
+ const std::vector<PlaneData>& plane_datas,
+ uint64_t modifier,
+ const DesktopVector& offset,
+ const DesktopSize& buffer_size,
+ uint8_t* data) {
if (!egl_initialized_) {
- return src;
+ return false;
}
if (plane_datas.size() <= 0) {
RTC_LOG(LS_ERROR) << "Failed to process buffer: invalid number of planes";
- return src;
+ return false;
}
EGLint attribs[47];
@@ -594,20 +633,32 @@ std::unique_ptr<uint8_t[]> EglDmaBuf::ImageFromDmaBuf(
if (image == EGL_NO_IMAGE) {
RTC_LOG(LS_ERROR) << "Failed to record frame: Error creating EGLImage - "
<< FormatEGLError(EglGetError());
- return src;
+ return false;
}
// create GL 2D texture for framebuffer
- GLuint texture;
- GlGenTextures(1, &texture);
- GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
- GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_NEAREST);
- GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
- GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
- GlBindTexture(GL_TEXTURE_2D, texture);
+ if (!texture_) {
+ GlGenTextures(1, &texture_);
+ GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
+ GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_NEAREST);
+ GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+ GlTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+ }
+ GlBindTexture(GL_TEXTURE_2D, texture_);
GlEGLImageTargetTexture2DOES(GL_TEXTURE_2D, image);
- src = std::make_unique<uint8_t[]>(plane_datas[0].stride * size.height());
+ if (!fbo_) {
+ GlGenFramebuffers(1, &fbo_);
+ }
+
+ GlBindFramebuffer(GL_FRAMEBUFFER, fbo_);
+ GlFramebufferTexture2D(GL_FRAMEBUFFER, GL_COLOR_ATTACHMENT0, GL_TEXTURE_2D,
+ texture_, 0);
+ if (GlCheckFramebufferStatus(GL_FRAMEBUFFER) != GL_FRAMEBUFFER_COMPLETE) {
+ RTC_LOG(LS_ERROR) << "Failed to bind DMA buf framebuffer";
+ EglDestroyImageKHR(egl_.display, image);
+ return false;
+ }
GLenum gl_format = GL_BGRA;
switch (format) {
@@ -624,17 +675,18 @@ std::unique_ptr<uint8_t[]> EglDmaBuf::ImageFromDmaBuf(
gl_format = GL_BGRA;
break;
}
- GlGetTexImage(GL_TEXTURE_2D, 0, gl_format, GL_UNSIGNED_BYTE, src.get());
- if (GlGetError()) {
+ GlReadPixels(offset.x(), offset.y(), buffer_size.width(),
+ buffer_size.height(), gl_format, GL_UNSIGNED_BYTE, data);
+
+ const GLenum error = GlGetError();
+ if (error) {
RTC_LOG(LS_ERROR) << "Failed to get image from DMA buffer.";
- return src;
}
- GlDeleteTextures(1, &texture);
EglDestroyImageKHR(egl_.display, image);
- return src;
+ return !error;
}
RTC_NO_SANITIZE("cfi-icall")
diff --git a/modules/desktop_capture/linux/wayland/egl_dmabuf.h b/modules/desktop_capture/linux/wayland/egl_dmabuf.h
index f1d96b2f80..22a8f5ab52 100644
--- a/modules/desktop_capture/linux/wayland/egl_dmabuf.h
+++ b/modules/desktop_capture/linux/wayland/egl_dmabuf.h
@@ -41,11 +41,15 @@ class EglDmaBuf {
EglDmaBuf();
~EglDmaBuf();
- std::unique_ptr<uint8_t[]> ImageFromDmaBuf(
- const DesktopSize& size,
- uint32_t format,
- const std::vector<PlaneData>& plane_datas,
- uint64_t modifiers);
+ // Returns whether the image was successfully imported from
+ // given DmaBuf and its parameters
+ bool ImageFromDmaBuf(const DesktopSize& size,
+ uint32_t format,
+ const std::vector<PlaneData>& plane_datas,
+ uint64_t modifiers,
+ const DesktopVector& offset,
+ const DesktopSize& buffer_size,
+ uint8_t* data);
std::vector<uint64_t> QueryDmaBufModifiers(uint32_t format);
bool IsEglInitialized() const { return egl_initialized_; }
@@ -58,6 +62,8 @@ class EglDmaBuf {
int32_t drm_fd_ = -1; // for GBM buffer mmap
gbm_device* gbm_device_ = nullptr; // for passed GBM buffer retrieval
+ GLuint fbo_ = 0;
+ GLuint texture_ = 0;
EGLStruct egl_;
absl::optional<std::string> GetRenderNode();
diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
index 71bde9b212..bcd7e3a33c 100644
--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
@@ -38,7 +38,6 @@ constexpr int CursorMetaSize(int w, int h) {
w * h * kCursorBpp);
}
-constexpr PipeWireVersion kDmaBufMinVersion = {0, 3, 24};
constexpr PipeWireVersion kDmaBufModifierMinVersion = {0, 3, 33};
constexpr PipeWireVersion kDropSingleModifierMinVersion = {0, 3, 40};
@@ -155,6 +154,12 @@ class SharedScreenCastStreamPrivate {
struct spa_video_info_raw spa_video_format_;
void ProcessBuffer(pw_buffer* buffer);
+ bool ProcessMemFDBuffer(pw_buffer* buffer,
+ DesktopFrame& frame,
+ const DesktopVector& offset);
+ bool ProcessDMABuffer(pw_buffer* buffer,
+ DesktopFrame& frame,
+ const DesktopVector& offset);
void ConvertRGBxToBGRx(uint8_t* frame, uint32_t size);
// PipeWire callbacks
@@ -277,10 +282,9 @@ void SharedScreenCastStreamPrivate::OnStreamParamChanged(
has_modifier ? that->spa_video_format_.modifier : DRM_FORMAT_MOD_INVALID;
std::vector<const spa_pod*> params;
const int buffer_types =
- has_modifier || (that->pw_server_version_ >= kDmaBufMinVersion)
- ? (1 << SPA_DATA_DmaBuf) | (1 << SPA_DATA_MemFd) |
- (1 << SPA_DATA_MemPtr)
- : (1 << SPA_DATA_MemFd) | (1 << SPA_DATA_MemPtr);
+ has_modifier
+ ? (1 << SPA_DATA_DmaBuf) | (1 << SPA_DATA_MemFd)
+ : (1 << SPA_DATA_MemFd);
params.push_back(reinterpret_cast<spa_pod*>(spa_pod_builder_add_object(
&builder, SPA_TYPE_OBJECT_ParamBuffers, SPA_PARAM_Buffers,
@@ -613,9 +617,6 @@ DesktopVector SharedScreenCastStreamPrivate::CaptureCursorPosition() {
RTC_NO_SANITIZE("cfi-icall")
void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
spa_buffer* spa_buffer = buffer->buffer;
- ScopedBuf map;
- std::unique_ptr<uint8_t[]> src_unique_ptr;
- uint8_t* src = nullptr;
// Try to update the mouse cursor first, because it can be the only
// information carried by the buffer
@@ -657,79 +658,6 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
return;
}
- if (spa_buffer->datas[0].type == SPA_DATA_MemFd) {
- map.initialize(
- static_cast<uint8_t*>(
- mmap(nullptr,
- spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
- PROT_READ, MAP_PRIVATE, spa_buffer->datas[0].fd, 0)),
- spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
- spa_buffer->datas[0].fd);
-
- if (!map) {
- RTC_LOG(LS_ERROR) << "Failed to mmap the memory: "
- << std::strerror(errno);
- return;
- }
-
- src = SPA_MEMBER(map.get(), spa_buffer->datas[0].mapoffset, uint8_t);
- } else if (spa_buffer->datas[0].type == SPA_DATA_DmaBuf) {
- const uint n_planes = spa_buffer->n_datas;
-
- if (!n_planes) {
- return;
- }
-
- std::vector<EglDmaBuf::PlaneData> plane_datas;
- for (uint32_t i = 0; i < n_planes; ++i) {
- EglDmaBuf::PlaneData data = {
- static_cast<int32_t>(spa_buffer->datas[i].fd),
- static_cast<uint32_t>(spa_buffer->datas[i].chunk->stride),
- static_cast<uint32_t>(spa_buffer->datas[i].chunk->offset)};
- plane_datas.push_back(data);
- }
-
- // When importing DMA-BUFs, we use the stride (number of bytes from one row
- // of pixels in the buffer) provided by PipeWire. The stride from PipeWire
- // is given by the graphics driver and some drivers might add some
- // additional padding for memory layout optimizations so not everytime the
- // stride is equal to BYTES_PER_PIXEL x WIDTH. This is fine, because during
- // the import we will use OpenGL and same graphics driver so it will be able
- // to work with the stride it provided, but later on when we work with
- // images we get from DMA-BUFs we will need to update the stride to be equal
- // to BYTES_PER_PIXEL x WIDTH as that's the size of the DesktopFrame we
- // allocate for each captured frame.
- src_unique_ptr = egl_dmabuf_->ImageFromDmaBuf(
- stream_size_, spa_video_format_.format, plane_datas, modifier_);
- if (src_unique_ptr) {
- src = src_unique_ptr.get();
- } else {
- RTC_LOG(LS_ERROR) << "Dropping DMA-BUF modifier: " << modifier_
- << " and trying to renegotiate stream parameters";
-
- if (pw_server_version_ >= kDropSingleModifierMinVersion) {
- modifiers_.erase(
- std::remove(modifiers_.begin(), modifiers_.end(), modifier_),
- modifiers_.end());
- } else {
- modifiers_.clear();
- }
-
- pw_loop_signal_event(pw_thread_loop_get_loop(pw_main_loop_),
- renegotiate_);
- return;
- }
- } else if (spa_buffer->datas[0].type == SPA_DATA_MemPtr) {
- src = static_cast<uint8_t*>(spa_buffer->datas[0].data);
- }
-
- if (!src) {
- if (observer_) {
- observer_->OnFailedToProcessBuffer();
- }
- return;
- }
-
// Use SPA_META_VideoCrop metadata to get the frame size. KDE and GNOME do
// handle screen/window sharing differently. KDE/KWin doesn't use
// SPA_META_VideoCrop metadata and when sharing a window, it always sets
@@ -787,8 +715,8 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
}
// Get the position of the video crop within the stream. Just double-check
- // that the position doesn't exceed the size of the stream itself. NOTE:
- // Currently it looks there is no implementation using this.
+ // that the position doesn't exceed the size of the stream itself.
+ // NOTE: Currently it looks there is no implementation using this.
uint32_t y_offset =
videocrop_metadata_use &&
(videocrop_metadata->region.position.y + frame_size_.height() <=
@@ -801,22 +729,7 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
stream_size_.width())
? videocrop_metadata->region.position.x
: 0;
-
- const uint32_t stream_stride = kBytesPerPixel * stream_size_.width();
- uint32_t buffer_stride = spa_buffer->datas[0].chunk->stride;
- uint32_t src_stride = buffer_stride;
-
- if (spa_buffer->datas[0].type == SPA_DATA_DmaBuf &&
- buffer_stride > stream_stride) {
- // When DMA-BUFs are used, sometimes spa_buffer->stride we get might
- // contain additional padding, but after we import the buffer, the stride
- // we used is no longer relevant and we should just calculate it based on
- // the stream width. For more context see https://crbug.com/1333304.
- src_stride = stream_stride;
- }
-
- uint8_t* updated_src =
- src + (src_stride * y_offset) + (kBytesPerPixel * x_offset);
+ DesktopVector offset = DesktopVector(x_offset, y_offset);
webrtc::MutexLock lock(&queue_lock_);
@@ -836,9 +749,20 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(std::move(frame)));
}
- queue_.current_frame()->CopyPixelsFrom(
- updated_src, (src_stride - (kBytesPerPixel * x_offset)),
- DesktopRect::MakeWH(frame_size_.width(), frame_size_.height()));
+ bool bufferProcessed = false;
+ if (spa_buffer->datas[0].type == SPA_DATA_MemFd) {
+ bufferProcessed =
+ ProcessMemFDBuffer(buffer, *queue_.current_frame(), offset);
+ } else if (spa_buffer->datas[0].type == SPA_DATA_DmaBuf) {
+ bufferProcessed = ProcessDMABuffer(buffer, *queue_.current_frame(), offset);
+ }
+
+ if (!bufferProcessed) {
+ if (observer_) {
+ observer_->OnFailedToProcessBuffer();
+ }
+ return;
+ }
if (spa_video_format_.format == SPA_VIDEO_FORMAT_RGBx ||
spa_video_format_.format == SPA_VIDEO_FORMAT_RGBA) {
@@ -885,6 +809,87 @@ void SharedScreenCastStreamPrivate::ProcessBuffer(pw_buffer* buffer) {
queue_.current_frame()->set_may_contain_cursor(is_cursor_embedded_);
}
+RTC_NO_SANITIZE("cfi-icall")
+bool SharedScreenCastStreamPrivate::ProcessMemFDBuffer(
+ pw_buffer* buffer,
+ DesktopFrame& frame,
+ const DesktopVector& offset) {
+ spa_buffer* spa_buffer = buffer->buffer;
+ ScopedBuf map;
+ uint8_t* src = nullptr;
+
+ map.initialize(
+ static_cast<uint8_t*>(
+ mmap(nullptr,
+ spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
+ PROT_READ, MAP_PRIVATE, spa_buffer->datas[0].fd, 0)),
+ spa_buffer->datas[0].maxsize + spa_buffer->datas[0].mapoffset,
+ spa_buffer->datas[0].fd);
+
+ if (!map) {
+ RTC_LOG(LS_ERROR) << "Failed to mmap the memory: " << std::strerror(errno);
+ return false;
+ }
+
+ src = SPA_MEMBER(map.get(), spa_buffer->datas[0].mapoffset, uint8_t);
+
+ uint32_t buffer_stride = spa_buffer->datas[0].chunk->stride;
+ uint32_t src_stride = buffer_stride;
+
+ uint8_t* updated_src =
+ src + (src_stride * offset.y()) + (kBytesPerPixel * offset.x());
+
+ frame.CopyPixelsFrom(
+ updated_src, (src_stride - (kBytesPerPixel * offset.x())),
+ DesktopRect::MakeWH(frame.size().width(), frame.size().height()));
+
+ return true;
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 3f313e2743..56afc5efce 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -214,8 +214,8 @@ rtc_library("rtp_sender_interface") {
":dtmf_sender_interface",
":frame_transformer_interface",
":media_stream_interface",
- ":rtc_error",
":rtp_parameters",
+ ":rtp_sender_setparameters_callback",
":scoped_refptr",
"../rtc_base:checks",
"../rtc_base:refcount",
@@ -223,6 +223,18 @@ rtc_library("rtp_sender_interface") {
"crypto:frame_encryptor_interface",
"video_codecs:video_codecs_api",
]
+}
+
+RTC_NO_SANITIZE("cfi-icall")
+bool SharedScreenCastStreamPrivate::ProcessDMABuffer(
+ pw_buffer* buffer,
+ DesktopFrame& frame,
+ const DesktopVector& offset) {
+ spa_buffer* spa_buffer = buffer->buffer;
+rtc_library("rtp_sender_setparameters_callback") {
+ visibility = [ "*" ]
+
+ const uint n_planes = spa_buffer->n_datas;
+ sources = [
+ "rtp_sender_setparameters_callback.cc",
+ "rtp_sender_setparameters_callback.h",
+ ]
+ deps = [
+ ":rtc_error",
+ ]
absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ]
}
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index 2786a2ac19..98ee91b1cc 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -32,9 +32,9 @@
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
-namespace webrtc {
+#include "api/rtp_sender_setparameters_callback.h"
-using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
+namespace webrtc {
class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
public:
diff --git a/api/rtp_sender_setparameters_callback.cc b/api/rtp_sender_setparameters_callback.cc
new file mode 100644
index 0000000000..99728ef95e
--- /dev/null
+++ b/api/rtp_sender_setparameters_callback.cc
@@ -0,0 +1,27 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+ if (!n_planes) {
+ return false;
+// File added by mozilla, to decouple this from libwebrtc's implementation of
+// RTCRtpSender.
+
+#include "api/rtp_sender_setparameters_callback.h"
+
+namespace webrtc {
+
+webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
+ RTCError error) {
+ if (callback) {
+ std::move(callback)(error);
+ callback = nullptr;
+ }
+
+ std::vector<EglDmaBuf::PlaneData> plane_datas;
+ for (uint32_t i = 0; i < n_planes; ++i) {
+ EglDmaBuf::PlaneData data = {
+ static_cast<int32_t>(spa_buffer->datas[i].fd),
+ static_cast<uint32_t>(spa_buffer->datas[i].chunk->stride),
+ static_cast<uint32_t>(spa_buffer->datas[i].chunk->offset)};
+ plane_datas.push_back(data);
+ }
+
+ const bool imported = egl_dmabuf_->ImageFromDmaBuf(
+ stream_size_, spa_video_format_.format, plane_datas, modifier_, offset,
+ frame.size(), frame.data());
+ if (!imported) {
+ RTC_LOG(LS_ERROR) << "Dropping DMA-BUF modifier: " << modifier_
+ << " and trying to renegotiate stream parameters";
+
+ if (pw_server_version_ >= kDropSingleModifierMinVersion) {
+ modifiers_.erase(
+ std::remove(modifiers_.begin(), modifiers_.end(), modifier_),
+ modifiers_.end());
+ } else {
+ modifiers_.clear();
+ }
+
+ pw_loop_signal_event(pw_thread_loop_get_loop(pw_main_loop_), renegotiate_);
+ return false;
+ }
+
+ return true;
+ return error;
+}
+
void SharedScreenCastStreamPrivate::ConvertRGBxToBGRx(uint8_t* frame,
uint32_t size) {
for (uint32_t i = 0; i < size; i += 4) {
+} // namespace webrtc
diff --git a/api/rtp_sender_setparameters_callback.h b/api/rtp_sender_setparameters_callback.h
new file mode 100644
index 0000000000..45194f5ace
--- /dev/null
+++ b/api/rtp_sender_setparameters_callback.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// File added by mozilla, to decouple this from libwebrtc's implementation of
+// RTCRtpSender.
+
+#ifndef API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
+#define API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
+
+#include "api/rtc_error.h"
+#include "absl/functional/any_invocable.h"
+
+namespace webrtc {
+
+using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
+
+webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
+ RTCError error);
+} // namespace webrtc
+
+#endif // API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
diff --git a/call/BUILD.gn b/call/BUILD.gn
index fb23b7ef39..2bc7aaec92 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -52,7 +52,7 @@ rtc_library("call_interfaces") {
"../api:rtc_error",
"../api:rtp_headers",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:transport_api",
"../api/adaptation:resource_adaptation_api",
@@ -389,7 +389,7 @@ rtc_library("video_stream_api") {
"../api:frame_transformer_interface",
"../api:rtp_headers",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:transport_api",
"../api/adaptation:resource_adaptation_api",
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index bafa22d312..187ec65ed8 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -25,7 +25,7 @@
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_parameters.h"
-#include "api/rtp_sender_interface.h"
+#include "api/rtp_sender_setparameters_callback.h"
#include "api/scoped_refptr.h"
#include "call/audio_sender.h"
#include "call/rtp_config.h"
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 431c267e1e..de18fc7b92 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -23,7 +23,7 @@
#include "api/crypto/crypto_options.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_parameters.h"
-#include "api/rtp_sender_interface.h"
+#include "api/rtp_sender_setparameters_callback.h"
#include "api/scoped_refptr.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
diff --git a/media/BUILD.gn b/media/BUILD.gn
index daca67e033..80662563b0 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -64,7 +64,7 @@ rtc_library("rtc_media_base") {
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:transport_api",
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 43e09290bd..138d28ae4c 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -64,10 +64,6 @@ class Timing;
namespace webrtc {
class VideoFrame;
-
-webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
- RTCError error);
-
} // namespace webrtc
namespace cricket {
diff --git a/media/base/media_channel_impl.cc b/media/base/media_channel_impl.cc
index a72b97413d..0e72f47d6d 100644
--- a/media/base/media_channel_impl.cc
+++ b/media/base/media_channel_impl.cc
@@ -31,19 +31,6 @@
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "rtc_base/checks.h"
-namespace webrtc {
-
-webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
- RTCError error) {
- if (callback) {
- std::move(callback)(error);
- callback = nullptr;
- }
- return error;
-}
-
-} // namespace webrtc
-
namespace cricket {
using webrtc::FrameDecryptorInterface;
using webrtc::FrameEncryptorInterface;
diff --git a/video/BUILD.gn b/video/BUILD.gn
index e21e1c7ea8..2c0a411e35 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -17,7 +17,7 @@ rtc_library("video_stream_encoder_interface") {
"../api:fec_controller_api",
"../api:rtc_error",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api/adaptation:resource_adaptation_api",
"../api/units:data_rate",
@@ -410,7 +410,7 @@ rtc_library("video_stream_encoder_impl") {
":video_stream_encoder_interface",
"../api:field_trials_view",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:sequence_checker",
"../api/adaptation:resource_adaptation_api",
"../api/task_queue:pending_task_safety_flag",
--
2.34.1

280
third_party/libwebrtc/moz-patch-stack/0093.patch поставляемый
Просмотреть файл

@ -1,269 +1,25 @@
From: Byron Campen <docfaraday@gmail.com>
Date: Fri, 31 Mar 2023 16:21:00 -0500
Subject: Bug 1822194 - (fix-acabb3641b) Break the new SetParametersCallback
stuff into stand-alone files.
Date: Tue, 4 Apr 2023 16:34:00 -0500
Subject: Bug 1822194 - (fix-3b51cd328e) - Add missing designated initializer
that gcc is sad about.
acabb3641b from upstream added a callback mechanism to allow failures to be
propagated back to RTCRtpSender.setParameters. Unfortunately, this callback
mechanism was (needlessly) tightly coupled to libwebrtc's implementation of
RTCRtpSender, and also their media channel code. This introduced a lot of
unnecessary dependencies throughout libwebrtc, that spilled into our code as
well.
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/59232687efa00e5f7b7bd3d6befca129149e2bf5
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/108046c7cbb21c6cf19320c0804e9aee1a3eb4bf
---
api/BUILD.gn | 14 +++++++++++-
api/rtp_sender_interface.h | 4 ++--
api/rtp_sender_setparameters_callback.cc | 27 +++++++++++++++++++++++
api/rtp_sender_setparameters_callback.h | 28 ++++++++++++++++++++++++
call/BUILD.gn | 4 ++--
call/audio_send_stream.h | 2 +-
call/video_send_stream.h | 2 +-
media/BUILD.gn | 2 +-
media/base/media_channel.h | 4 ----
media/base/media_channel_impl.cc | 13 -----------
video/BUILD.gn | 4 ++--
11 files changed, 77 insertions(+), 27 deletions(-)
create mode 100644 api/rtp_sender_setparameters_callback.cc
create mode 100644 api/rtp_sender_setparameters_callback.h
modules/audio_processing/audio_processing_impl.cc | 1 +
1 file changed, 1 insertion(+)
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 3f313e2743..56afc5efce 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -214,8 +214,8 @@ rtc_library("rtp_sender_interface") {
":dtmf_sender_interface",
":frame_transformer_interface",
":media_stream_interface",
- ":rtc_error",
":rtp_parameters",
+ ":rtp_sender_setparameters_callback",
":scoped_refptr",
"../rtc_base:checks",
"../rtc_base:refcount",
@@ -223,6 +223,18 @@ rtc_library("rtp_sender_interface") {
"crypto:frame_encryptor_interface",
"video_codecs:video_codecs_api",
]
+}
+
+rtc_library("rtp_sender_setparameters_callback") {
+ visibility = [ "*" ]
+
+ sources = [
+ "rtp_sender_setparameters_callback.cc",
+ "rtp_sender_setparameters_callback.h",
+ ]
+ deps = [
+ ":rtc_error",
+ ]
absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ]
}
diff --git a/api/rtp_sender_interface.h b/api/rtp_sender_interface.h
index 2786a2ac19..98ee91b1cc 100644
--- a/api/rtp_sender_interface.h
+++ b/api/rtp_sender_interface.h
@@ -32,9 +32,9 @@
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
-namespace webrtc {
+#include "api/rtp_sender_setparameters_callback.h"
-using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
+namespace webrtc {
class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
public:
diff --git a/api/rtp_sender_setparameters_callback.cc b/api/rtp_sender_setparameters_callback.cc
new file mode 100644
index 0000000000..99728ef95e
--- /dev/null
+++ b/api/rtp_sender_setparameters_callback.cc
@@ -0,0 +1,27 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// File added by mozilla, to decouple this from libwebrtc's implementation of
+// RTCRtpSender.
+
+#include "api/rtp_sender_setparameters_callback.h"
+
+namespace webrtc {
+
+webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
+ RTCError error) {
+ if (callback) {
+ std::move(callback)(error);
+ callback = nullptr;
+ }
+ return error;
+}
+
+} // namespace webrtc
diff --git a/api/rtp_sender_setparameters_callback.h b/api/rtp_sender_setparameters_callback.h
new file mode 100644
index 0000000000..45194f5ace
--- /dev/null
+++ b/api/rtp_sender_setparameters_callback.h
@@ -0,0 +1,28 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+// File added by mozilla, to decouple this from libwebrtc's implementation of
+// RTCRtpSender.
+
+#ifndef API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
+#define API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
+
+#include "api/rtc_error.h"
+#include "absl/functional/any_invocable.h"
+
+namespace webrtc {
+
+using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
+
+webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
+ RTCError error);
+} // namespace webrtc
+
+#endif // API_RTP_SENDER_SETPARAMETERS_CALLBACK_H_
diff --git a/call/BUILD.gn b/call/BUILD.gn
index fb23b7ef39..2bc7aaec92 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -52,7 +52,7 @@ rtc_library("call_interfaces") {
"../api:rtc_error",
"../api:rtp_headers",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:transport_api",
"../api/adaptation:resource_adaptation_api",
@@ -389,7 +389,7 @@ rtc_library("video_stream_api") {
"../api:frame_transformer_interface",
"../api:rtp_headers",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:transport_api",
"../api/adaptation:resource_adaptation_api",
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index bafa22d312..187ec65ed8 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -25,7 +25,7 @@
#include "api/crypto/frame_encryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_parameters.h"
-#include "api/rtp_sender_interface.h"
+#include "api/rtp_sender_setparameters_callback.h"
#include "api/scoped_refptr.h"
#include "call/audio_sender.h"
#include "call/rtp_config.h"
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 431c267e1e..de18fc7b92 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -23,7 +23,7 @@
#include "api/crypto/crypto_options.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_parameters.h"
-#include "api/rtp_sender_interface.h"
+#include "api/rtp_sender_setparameters_callback.h"
#include "api/scoped_refptr.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
diff --git a/media/BUILD.gn b/media/BUILD.gn
index daca67e033..80662563b0 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -64,7 +64,7 @@ rtc_library("rtc_media_base") {
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api:sequence_checker",
"../api:transport_api",
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 43e09290bd..138d28ae4c 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -64,10 +64,6 @@ class Timing;
namespace webrtc {
class VideoFrame;
-
-webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
- RTCError error);
-
} // namespace webrtc
namespace cricket {
diff --git a/media/base/media_channel_impl.cc b/media/base/media_channel_impl.cc
index a72b97413d..0e72f47d6d 100644
--- a/media/base/media_channel_impl.cc
+++ b/media/base/media_channel_impl.cc
@@ -31,19 +31,6 @@
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "rtc_base/checks.h"
-namespace webrtc {
-
-webrtc::RTCError InvokeSetParametersCallback(SetParametersCallback& callback,
- RTCError error) {
- if (callback) {
- std::move(callback)(error);
- callback = nullptr;
- }
- return error;
-}
-
-} // namespace webrtc
-
namespace cricket {
using webrtc::FrameDecryptorInterface;
using webrtc::FrameEncryptorInterface;
diff --git a/video/BUILD.gn b/video/BUILD.gn
index e21e1c7ea8..2c0a411e35 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -17,7 +17,7 @@ rtc_library("video_stream_encoder_interface") {
"../api:fec_controller_api",
"../api:rtc_error",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:scoped_refptr",
"../api/adaptation:resource_adaptation_api",
"../api/units:data_rate",
@@ -410,7 +410,7 @@ rtc_library("video_stream_encoder_impl") {
":video_stream_encoder_interface",
"../api:field_trials_view",
"../api:rtp_parameters",
- "../api:rtp_sender_interface",
+ "../api:rtp_sender_setparameters_callback",
"../api:sequence_checker",
"../api/adaptation:resource_adaptation_api",
"../api/task_queue:pending_task_safety_flag",
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index c80cc76a3d..c304453388 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -450,6 +450,7 @@ AudioProcessingImpl::GetGainController2ExperimentParams() {
},
.adaptive_digital_controller =
{
+ .enabled = false,
.headroom_db = static_cast<float>(headroom_db.Get()),
.max_gain_db = static_cast<float>(max_gain_db.Get()),
.initial_gain_db =
--
2.34.1

38
third_party/libwebrtc/moz-patch-stack/0094.patch поставляемый
Просмотреть файл

@ -1,25 +1,27 @@
From: Byron Campen <docfaraday@gmail.com>
Date: Tue, 4 Apr 2023 16:34:00 -0500
Subject: Bug 1822194 - (fix-3b51cd328e) - Add missing designated initializer
that gcc is sad about.
Date: Fri, 7 Apr 2023 20:28:00 +0000
Subject: Bug 1819048: Remove this bad assertion. r=webrtc-reviewers,jib
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/108046c7cbb21c6cf19320c0804e9aee1a3eb4bf
Differential Revision: https://phabricator.services.mozilla.com/D174978
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/5a52e1b0c808edfda82f0abea668699eb68098dc
---
modules/audio_processing/audio_processing_impl.cc | 1 +
1 file changed, 1 insertion(+)
video/task_queue_frame_decode_scheduler.cc | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index c80cc76a3d..c304453388 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -450,6 +450,7 @@ AudioProcessingImpl::GetGainController2ExperimentParams() {
},
.adaptive_digital_controller =
{
+ .enabled = false,
.headroom_db = static_cast<float>(headroom_db.Get()),
.max_gain_db = static_cast<float>(max_gain_db.Get()),
.initial_gain_db =
diff --git a/video/task_queue_frame_decode_scheduler.cc b/video/task_queue_frame_decode_scheduler.cc
index cd109c2932..6dd7b47f17 100644
--- a/video/task_queue_frame_decode_scheduler.cc
+++ b/video/task_queue_frame_decode_scheduler.cc
@@ -37,7 +37,8 @@ void TaskQueueFrameDecodeScheduler::ScheduleFrame(
uint32_t rtp,
FrameDecodeTiming::FrameSchedule schedule,
FrameReleaseCallback cb) {
- RTC_DCHECK(!stopped_) << "Can not schedule frames after stopped.";
+ // Mozilla modification, until https://bugs.webrtc.org/14944 is fixed
+ //RTC_DCHECK(!stopped_) << "Can not schedule frames after stopped.";
RTC_DCHECK(!scheduled_rtp_.has_value())
<< "Can not schedule two frames for release at the same time.";
RTC_DCHECK(cb);
--
2.34.1

161
third_party/libwebrtc/moz-patch-stack/0095.patch поставляемый
Просмотреть файл

@ -1,27 +1,146 @@
From: Byron Campen <docfaraday@gmail.com>
Date: Fri, 7 Apr 2023 20:28:00 +0000
Subject: Bug 1819048: Remove this bad assertion. r=webrtc-reviewers,jib
From: Michael Froman <mfroman@mozilla.com>
Date: Thu, 20 Apr 2023 09:59:00 -0500
Subject: Bug 1828517 - (fix-794d599741) account for moved files in BUILD.gn
that we don't want to build.
Differential Revision: https://phabricator.services.mozilla.com/D174978
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/5a52e1b0c808edfda82f0abea668699eb68098dc
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/4a969f6709183d4f55215adaffb8a52b790a8492
---
video/task_queue_frame_decode_scheduler.cc | 3 ++-
1 file changed, 2 insertions(+), 1 deletion(-)
api/BUILD.gn | 10 ++++++++++
media/BUILD.gn | 20 ++++++++++----------
2 files changed, 20 insertions(+), 10 deletions(-)
diff --git a/video/task_queue_frame_decode_scheduler.cc b/video/task_queue_frame_decode_scheduler.cc
index cd109c2932..6dd7b47f17 100644
--- a/video/task_queue_frame_decode_scheduler.cc
+++ b/video/task_queue_frame_decode_scheduler.cc
@@ -37,7 +37,8 @@ void TaskQueueFrameDecodeScheduler::ScheduleFrame(
uint32_t rtp,
FrameDecodeTiming::FrameSchedule schedule,
FrameReleaseCallback cb) {
- RTC_DCHECK(!stopped_) << "Can not schedule frames after stopped.";
+ // Mozilla modification, until https://bugs.webrtc.org/14944 is fixed
+ //RTC_DCHECK(!stopped_) << "Can not schedule frames after stopped.";
RTC_DCHECK(!scheduled_rtp_.has_value())
<< "Can not schedule two frames for release at the same time.";
RTC_DCHECK(cb);
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 56afc5efce..7c16b45e05 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -175,6 +175,10 @@ rtc_source_set("ice_transport_interface") {
}
rtc_library("dtls_transport_interface") {
+# Previously, Mozilla has tried to limit including this dep, but as
+# upstream changes, it requires whack-a-mole. Making it an empty
+# definition has the same effect, but only requires one change.
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
@@ -191,6 +195,7 @@ rtc_library("dtls_transport_interface") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_library("dtmf_sender_interface") {
visibility = [ "*" ]
@@ -203,6 +208,10 @@ rtc_library("dtmf_sender_interface") {
}
rtc_library("rtp_sender_interface") {
+# Previously, Mozilla has tried to limit including this dep, but as
+# upstream changes, it requires whack-a-mole. Making it an empty
+# definition has the same effect, but only requires one change.
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
@@ -224,6 +233,7 @@ rtc_library("rtp_sender_interface") {
"video_codecs:video_codecs_api",
]
}
+}
rtc_library("rtp_sender_setparameters_callback") {
visibility = [ "*" ]
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 80662563b0..36c3aa9d1c 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -154,23 +154,14 @@ rtc_library("rtc_media_base") {
"base/audio_source.h",
"base/delayable.h",
"base/media_channel.h",
- "base/media_channel_impl.cc",
- "base/media_channel_impl.h",
"base/media_engine.cc",
"base/media_engine.h",
- "base/rid_description.cc",
- "base/rid_description.h",
- "base/rtp_utils.cc",
- "base/rtp_utils.h",
- "base/stream_params.cc",
- "base/stream_params.h",
- "base/turn_utils.cc",
- "base/turn_utils.h",
]
}
}
rtc_library("media_channel_impl") {
+if (!build_with_mozilla) {
sources = [
"base/media_channel_impl.cc",
"base/media_channel_impl.h",
@@ -219,6 +210,7 @@ rtc_library("media_channel_impl") {
"//third_party/abseil-cpp/absl/types:optional",
]
}
+}
rtc_source_set("media_channel") {
sources = [ "base/media_channel.h" ]
@@ -292,6 +284,7 @@ rtc_library("codec") {
}
rtc_library("rtp_utils") {
+if (!build_with_mozilla) {
sources = [
"base/rtp_utils.cc",
"base/rtp_utils.h",
@@ -308,8 +301,10 @@ rtc_library("rtp_utils") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_library("stream_params") {
+if (!build_with_mozilla) {
sources = [
"base/stream_params.cc",
"base/stream_params.h",
@@ -322,6 +317,7 @@ rtc_library("stream_params") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
}
+}
rtc_library("media_constants") {
sources = [
@@ -332,6 +328,7 @@ rtc_library("media_constants") {
}
rtc_library("turn_utils") {
+if (!build_with_mozilla) {
sources = [
"base/turn_utils.cc",
"base/turn_utils.h",
@@ -342,14 +339,17 @@ rtc_library("turn_utils") {
"../rtc_base/system:rtc_export",
]
}
+}
rtc_library("rid_description") {
+if (!build_with_mozilla) {
sources = [
"base/rid_description.cc",
"base/rid_description.h",
]
deps = []
}
+}
rtc_library("rtc_simulcast_encoder_adapter") {
visibility = [ "*" ]
--
2.34.1

245
third_party/libwebrtc/moz-patch-stack/0096.patch поставляемый
Просмотреть файл

@ -1,146 +1,177 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Thu, 20 Apr 2023 09:59:00 -0500
Subject: Bug 1828517 - (fix-794d599741) account for moved files in BUILD.gn
that we don't want to build.
Date: Thu, 20 Apr 2023 14:52:00 -0500
Subject: Bug 1828517 - (fix-a138c6c8a5) handle file moves in BUILD.gn
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/4a969f6709183d4f55215adaffb8a52b790a8492
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cf7e333da17689b3c115a6ffd07fab042bc5f086
---
api/BUILD.gn | 10 ++++++++++
media/BUILD.gn | 20 ++++++++++----------
2 files changed, 20 insertions(+), 10 deletions(-)
rtc_base/BUILD.gn | 24 ++++++++++++++++++++++++
1 file changed, 24 insertions(+)
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 56afc5efce..7c16b45e05 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -175,6 +175,10 @@ rtc_source_set("ice_transport_interface") {
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 0b1e2a6208..7573a137ab 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -1147,6 +1147,7 @@ if (!build_with_chromium) {
}
rtc_library("dtls_transport_interface") {
+# Previously, Mozilla has tried to limit including this dep, but as
+# upstream changes, it requires whack-a-mole. Making it an empty
+# definition has the same effect, but only requires one change.
rtc_library("network") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
@@ -191,6 +195,7 @@ rtc_library("dtls_transport_interface") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_library("dtmf_sender_interface") {
visibility = [ "*" ]
@@ -203,6 +208,10 @@ rtc_library("dtmf_sender_interface") {
}
rtc_library("rtp_sender_interface") {
+# Previously, Mozilla has tried to limit including this dep, but as
+# upstream changes, it requires whack-a-mole. Making it an empty
+# definition has the same effect, but only requires one change.
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
@@ -224,6 +233,7 @@ rtc_library("rtp_sender_interface") {
"video_codecs:video_codecs_api",
]
}
+}
rtc_library("rtp_sender_setparameters_callback") {
visibility = [ "*" ]
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 80662563b0..36c3aa9d1c 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -154,23 +154,14 @@ rtc_library("rtc_media_base") {
"base/audio_source.h",
"base/delayable.h",
"base/media_channel.h",
- "base/media_channel_impl.cc",
- "base/media_channel_impl.h",
"base/media_engine.cc",
"base/media_engine.h",
- "base/rid_description.cc",
- "base/rid_description.h",
- "base/rtp_utils.cc",
- "base/rtp_utils.h",
- "base/stream_params.cc",
- "base/stream_params.h",
- "base/turn_utils.cc",
- "base/turn_utils.h",
]
"network.cc",
@@ -1185,16 +1186,20 @@ rtc_library("network") {
deps += [ ":win32" ]
}
}
+}
rtc_library("media_channel_impl") {
rtc_library("socket_address_pair") {
+if (!build_with_mozilla) {
sources = [
"base/media_channel_impl.cc",
"base/media_channel_impl.h",
@@ -219,6 +210,7 @@ rtc_library("media_channel_impl") {
"//third_party/abseil-cpp/absl/types:optional",
"socket_address_pair.cc",
"socket_address_pair.h",
]
deps = [ ":socket_address" ]
}
+}
rtc_source_set("media_channel") {
sources = [ "base/media_channel.h" ]
@@ -292,6 +284,7 @@ rtc_library("codec") {
}
rtc_library("rtp_utils") {
rtc_library("net_helper") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"base/rtp_utils.cc",
"base/rtp_utils.h",
@@ -308,8 +301,10 @@ rtc_library("rtp_utils") {
"net_helper.cc",
@@ -1203,8 +1208,10 @@ rtc_library("net_helper") {
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
deps = [ "system:rtc_export" ]
}
+}
rtc_library("socket_adapters") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"socket_adapters.cc",
@@ -1224,6 +1231,7 @@ rtc_library("socket_adapters") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_library("stream_params") {
+if (!build_with_mozilla) {
rtc_library("network_route") {
sources = [
"base/stream_params.cc",
"base/stream_params.h",
@@ -322,6 +317,7 @@ rtc_library("stream_params") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
}
+}
rtc_library("media_constants") {
sources = [
@@ -332,6 +328,7 @@ rtc_library("media_constants") {
@@ -1238,6 +1246,7 @@ rtc_library("network_route") {
}
rtc_library("turn_utils") {
rtc_library("async_tcp_socket") {
+if (!build_with_mozilla) {
sources = [
"base/turn_utils.cc",
"base/turn_utils.h",
@@ -342,14 +339,17 @@ rtc_library("turn_utils") {
"../rtc_base/system:rtc_export",
"async_tcp_socket.cc",
"async_tcp_socket.h",
@@ -1256,8 +1265,10 @@ rtc_library("async_tcp_socket") {
"third_party/sigslot",
]
}
+}
rtc_library("rid_description") {
rtc_library("async_udp_socket") {
+if (!build_with_mozilla) {
sources = [
"base/rid_description.cc",
"base/rid_description.h",
]
deps = []
}
+}
rtc_library("rtc_simulcast_encoder_adapter") {
visibility = [ "*" ]
sources = [
"async_udp_socket.cc",
@@ -1279,8 +1290,10 @@ rtc_library("async_udp_socket") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_library("async_packet_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_packet_socket.cc",
@@ -1298,6 +1311,7 @@ rtc_library("async_packet_socket") {
"third_party/sigslot",
]
}
+}
rtc_library("mdns_responder_interface") {
sources = [ "mdns_responder_interface.h" ]
@@ -1310,6 +1324,7 @@ rtc_library("dscp") {
}
rtc_library("proxy_info") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"proxy_info.cc",
@@ -1320,6 +1335,7 @@ rtc_library("proxy_info") {
":socket_address",
]
}
+}
rtc_library("file_rotating_stream") {
sources = [
@@ -1348,6 +1364,7 @@ rtc_library("data_rate_limiter") {
}
rtc_library("unique_id_generator") {
+if (!build_with_mozilla) {
sources = [
"unique_id_generator.cc",
"unique_id_generator.h",
@@ -1362,6 +1379,7 @@ rtc_library("unique_id_generator") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_library("crc32") {
sources = [
@@ -1389,6 +1407,7 @@ rtc_library("stream") {
}
rtc_library("rtc_certificate_generator") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"rtc_certificate_generator.cc",
@@ -1406,8 +1425,10 @@ rtc_library("rtc_certificate_generator") {
"//third_party/abseil-cpp/absl/types:optional",
]
}
+}
rtc_library("ssl") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"helpers.cc",
@@ -1506,6 +1527,7 @@ rtc_library("ssl") {
deps += [ ":win32" ]
}
}
+}
rtc_library("crypt_string") {
sources = [
@@ -1515,6 +1537,7 @@ rtc_library("crypt_string") {
}
rtc_library("http_common") {
+if (!build_with_mozilla) {
sources = [
"http_common.cc",
"http_common.h",
@@ -1531,6 +1554,7 @@ rtc_library("http_common") {
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_source_set("gtest_prod") {
sources = [ "gtest_prod_util.h" ]
--
2.34.1

177
third_party/libwebrtc/moz-patch-stack/0097.patch поставляемый
Просмотреть файл

@ -1,177 +0,0 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Thu, 20 Apr 2023 14:52:00 -0500
Subject: Bug 1828517 - (fix-a138c6c8a5) handle file moves in BUILD.gn
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/cf7e333da17689b3c115a6ffd07fab042bc5f086
---
rtc_base/BUILD.gn | 24 ++++++++++++++++++++++++
1 file changed, 24 insertions(+)
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 0b1e2a6208..7573a137ab 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -1147,6 +1147,7 @@ if (!build_with_chromium) {
}
rtc_library("network") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"network.cc",
@@ -1185,16 +1186,20 @@ rtc_library("network") {
deps += [ ":win32" ]
}
}
+}
rtc_library("socket_address_pair") {
+if (!build_with_mozilla) {
sources = [
"socket_address_pair.cc",
"socket_address_pair.h",
]
deps = [ ":socket_address" ]
}
+}
rtc_library("net_helper") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"net_helper.cc",
@@ -1203,8 +1208,10 @@ rtc_library("net_helper") {
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
deps = [ "system:rtc_export" ]
}
+}
rtc_library("socket_adapters") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"socket_adapters.cc",
@@ -1224,6 +1231,7 @@ rtc_library("socket_adapters") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_library("network_route") {
sources = [
@@ -1238,6 +1246,7 @@ rtc_library("network_route") {
}
rtc_library("async_tcp_socket") {
+if (!build_with_mozilla) {
sources = [
"async_tcp_socket.cc",
"async_tcp_socket.h",
@@ -1256,8 +1265,10 @@ rtc_library("async_tcp_socket") {
"third_party/sigslot",
]
}
+}
rtc_library("async_udp_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_udp_socket.cc",
@@ -1279,8 +1290,10 @@ rtc_library("async_udp_socket") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
+}
rtc_library("async_packet_socket") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"async_packet_socket.cc",
@@ -1298,6 +1311,7 @@ rtc_library("async_packet_socket") {
"third_party/sigslot",
]
}
+}
rtc_library("mdns_responder_interface") {
sources = [ "mdns_responder_interface.h" ]
@@ -1310,6 +1324,7 @@ rtc_library("dscp") {
}
rtc_library("proxy_info") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"proxy_info.cc",
@@ -1320,6 +1335,7 @@ rtc_library("proxy_info") {
":socket_address",
]
}
+}
rtc_library("file_rotating_stream") {
sources = [
@@ -1348,6 +1364,7 @@ rtc_library("data_rate_limiter") {
}
rtc_library("unique_id_generator") {
+if (!build_with_mozilla) {
sources = [
"unique_id_generator.cc",
"unique_id_generator.h",
@@ -1362,6 +1379,7 @@ rtc_library("unique_id_generator") {
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_library("crc32") {
sources = [
@@ -1389,6 +1407,7 @@ rtc_library("stream") {
}
rtc_library("rtc_certificate_generator") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"rtc_certificate_generator.cc",
@@ -1406,8 +1425,10 @@ rtc_library("rtc_certificate_generator") {
"//third_party/abseil-cpp/absl/types:optional",
]
}
+}
rtc_library("ssl") {
+if (!build_with_mozilla) {
visibility = [ "*" ]
sources = [
"helpers.cc",
@@ -1506,6 +1527,7 @@ rtc_library("ssl") {
deps += [ ":win32" ]
}
}
+}
rtc_library("crypt_string") {
sources = [
@@ -1515,6 +1537,7 @@ rtc_library("crypt_string") {
}
rtc_library("http_common") {
+if (!build_with_mozilla) {
sources = [
"http_common.cc",
"http_common.h",
@@ -1531,6 +1554,7 @@ rtc_library("http_common") {
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
}
+}
rtc_source_set("gtest_prod") {
sources = [ "gtest_prod_util.h" ]
--
2.34.1

1
third_party/libwebrtc/moz-patch-stack/f2a083f262.no-op-cherry-pick-msg поставляемый
Просмотреть файл

@ -1 +0,0 @@
We already cherry-picked this when we vendored 897ea04db5.