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Bug 1828517 - Vendor libwebrtc from 175f06f112 

Upstream commit: https://webrtc.googlesource.com/src/+/175f06f112a7b896a6789236037a1dd46d99a77e
    Reland "Remove 'trackId' dependency in stats selector algorithm."

    This is a reland of commit 81aab488781c1a736c9d85ff1532631be2989523

    See diff between Patch Set 1 and latest Patch Set.

    The original CL broke this WPT[1] because getStats() with the receiver
    as the selector stopped working in the event of unsignalled SSRCs due
    to the receiver not knowing what the SSRC was.

    This fix is to query media_channel_ for the unsignalled SSRC in the
    event that the receiver does not know the SSRC.

    [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/simulcast/setParameters-active.https.html

    Original change's description:
    > Remove 'trackId' dependency in stats selector algorithm.
    >
    > In preparation for the deletion of deprecated 'track' stats, the
    > stats selector algorithm needs to be rewritten not to use 'trackId'.
    >
    > This is achieved by finding RTP stats by their SSRC, as obtained via
    > getParameters(). This unfortunately adds a block-invoke (in the sender
    > case the block-invoke happens inside GetParametersInternal and in the
    > receiver case the block-invoke is explicit at the calling place), but
    > it can't be helped and it's just once per getStats() call and only if
    > the selector argument is used.
    >
    > Bug: webrtc:14175
    > Change-Id: If0e14cdbdc76d141e0042e43757970893bf32119
    > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289101
    > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    > Commit-Queue: Henrik Boström <hbos@webrtc.org>
    > Cr-Commit-Position: refs/heads/main@{#38981}

    Bug: webrtc:14175, webrtc:14811
    Change-Id: I0d16724af4efeb93d50e36dbfcc798564daff5c0
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290600
    Commit-Queue: Henrik Boström <hbos@webrtc.org>
    Reviewed-by: Harald Alvestrand <hta@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#39010}
This commit is contained in:
Michael Froman 2023-04-20 13:47:25 -05:00
Родитель 6d895700f8
Коммит de53d150cc
22 изменённых файлов: 106 добавлений и 65 удалений
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3
third_party/libwebrtc/README.moz-ff-commit поставляемый
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@ -20646,3 +20646,6 @@ b81823a5f0
# 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
d152a6d51c
# 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
175f06f112

2
third_party/libwebrtc/README.mozilla поставляемый
Просмотреть файл

@ -13786,3 +13786,5 @@ 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 2023-04-20T18:45:17.770427.
# ./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 2023-04-20T18:46:17.511994.
# ./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 2023-04-20T18:47:13.643593.

3
third_party/libwebrtc/media/base/fake_media_engine.h поставляемый
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@ -121,6 +121,9 @@ class RtpHelper : public Base {
return RemoveStreamBySsrc(&send_streams_, ssrc);
}
virtual void ResetUnsignaledRecvStream() {}
virtual absl::optional<uint32_t> GetUnsignaledSsrc() const {
return absl::nullopt;
}
virtual void OnDemuxerCriteriaUpdatePending() {}
virtual void OnDemuxerCriteriaUpdateComplete() {}

2
third_party/libwebrtc/media/base/media_channel.h поставляемый
Просмотреть файл

@ -278,6 +278,8 @@ class MediaReceiveChannelInterface
// Resets any cached StreamParams for an unsignaled RecvStream, and removes
// any existing unsignaled streams.
virtual void ResetUnsignaledRecvStream() = 0;
// Gets the current unsignaled receive stream's SSRC, if there is one.
virtual absl::optional<uint32_t> GetUnsignaledSsrc() const = 0;
// This is currently a workaround because of the demuxer state being managed
// across two separate threads. Once the state is consistently managed on
// the same thread (network), this workaround can be removed.

6
third_party/libwebrtc/media/base/media_channel_impl.h поставляемый
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@ -420,6 +420,9 @@ class VoiceMediaReceiveChannel : public VoiceMediaReceiveChannelInterface {
void ResetUnsignaledRecvStream() override {
return impl()->ResetUnsignaledRecvStream();
}
absl::optional<uint32_t> GetUnsignaledSsrc() const override {
return impl()->GetUnsignaledSsrc();
}
void OnDemuxerCriteriaUpdatePending() override {
impl()->OnDemuxerCriteriaUpdatePending();
}
@ -614,6 +617,9 @@ class VideoMediaReceiveChannel : public VideoMediaReceiveChannelInterface {
void ResetUnsignaledRecvStream() override {
return impl()->ResetUnsignaledRecvStream();
}
absl::optional<uint32_t> GetUnsignaledSsrc() const override {
return impl()->GetUnsignaledSsrc();
}
void OnDemuxerCriteriaUpdatePending() override {
impl()->OnDemuxerCriteriaUpdatePending();
}

36
third_party/libwebrtc/media/engine/webrtc_video_engine.cc поставляемый
Просмотреть файл

@ -548,8 +548,7 @@ UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
WebRtcVideoChannel* channel,
uint32_t ssrc,
absl::optional<uint32_t> rtx_ssrc) {
absl::optional<uint32_t> default_recv_ssrc =
channel->GetDefaultReceiveStreamSsrc();
absl::optional<uint32_t> default_recv_ssrc = channel->GetUnsignaledSsrc();
if (default_recv_ssrc) {
RTC_LOG(LS_INFO) << "Destroying old default receive stream for SSRC="
@ -588,8 +587,7 @@ void DefaultUnsignalledSsrcHandler::SetDefaultSink(
WebRtcVideoChannel* channel,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
default_sink_ = sink;
absl::optional<uint32_t> default_recv_ssrc =
channel->GetDefaultReceiveStreamSsrc();
absl::optional<uint32_t> default_recv_ssrc = channel->GetUnsignaledSsrc();
if (default_recv_ssrc) {
channel->SetSink(*default_recv_ssrc, default_sink_);
}
@ -1566,6 +1564,18 @@ void WebRtcVideoChannel::ResetUnsignaledRecvStream() {
}
}
absl::optional<uint32_t> WebRtcVideoChannel::GetUnsignaledSsrc() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
absl::optional<uint32_t> ssrc;
for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) {
if (it->second->IsDefaultStream()) {
ssrc.emplace(it->first);
break;
}
}
return ssrc;
}
void WebRtcVideoChannel::OnDemuxerCriteriaUpdatePending() {
RTC_DCHECK_RUN_ON(&thread_checker_);
++demuxer_criteria_id_;
@ -1756,7 +1766,7 @@ void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer packet,
// stream, which will be associated with unsignaled media stream.
// It is not possible to update the ssrcs of a receive stream, so we
// recreate it insead if found.
auto default_ssrc = GetDefaultReceiveStreamSsrc();
auto default_ssrc = GetUnsignaledSsrc();
if (!default_ssrc) {
return;
}
@ -1919,7 +1929,7 @@ void WebRtcVideoChannel::SetVideoCodecSwitchingEnabled(bool enabled) {
bool WebRtcVideoChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc,
int delay_ms) {
RTC_DCHECK_RUN_ON(&thread_checker_);
absl::optional<uint32_t> default_ssrc = GetDefaultReceiveStreamSsrc();
absl::optional<uint32_t> default_ssrc = GetUnsignaledSsrc();
// SSRC of 0 represents the default receive stream.
if (ssrc == 0) {
@ -1961,18 +1971,6 @@ absl::optional<int> WebRtcVideoChannel::GetBaseMinimumPlayoutDelayMs(
}
}
absl::optional<uint32_t> WebRtcVideoChannel::GetDefaultReceiveStreamSsrc() {
RTC_DCHECK_RUN_ON(&thread_checker_);
absl::optional<uint32_t> ssrc;
for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) {
if (it->second->IsDefaultStream()) {
ssrc.emplace(it->first);
break;
}
}
return ssrc;
}
std::vector<webrtc::RtpSource> WebRtcVideoChannel::GetSources(
uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
@ -3496,7 +3494,7 @@ WebRtcVideoChannel::MapCodecs(const std::vector<VideoCodec>& codecs) {
WebRtcVideoChannel::WebRtcVideoReceiveStream*
WebRtcVideoChannel::FindReceiveStream(uint32_t ssrc) {
if (ssrc == 0) {
absl::optional<uint32_t> default_ssrc = GetDefaultReceiveStreamSsrc();
absl::optional<uint32_t> default_ssrc = GetUnsignaledSsrc();
if (!default_ssrc) {
return nullptr;
}

3
third_party/libwebrtc/media/engine/webrtc_video_engine.h поставляемый
Просмотреть файл

@ -165,6 +165,7 @@ class WebRtcVideoChannel : public VideoMediaChannel,
bool AddRecvStream(const StreamParams& sp, bool default_stream);
bool RemoveRecvStream(uint32_t ssrc) override;
void ResetUnsignaledRecvStream() override;
absl::optional<uint32_t> GetUnsignaledSsrc() const override;
void OnDemuxerCriteriaUpdatePending() override;
void OnDemuxerCriteriaUpdateComplete() override;
bool SetSink(uint32_t ssrc,
@ -216,8 +217,6 @@ class WebRtcVideoChannel : public VideoMediaChannel,
return sending_;
}
absl::optional<uint32_t> GetDefaultReceiveStreamSsrc();
StreamParams unsignaled_stream_params() {
RTC_DCHECK_RUN_ON(&thread_checker_);
return unsignaled_stream_params_;

9
third_party/libwebrtc/media/engine/webrtc_voice_engine.cc поставляемый
Просмотреть файл

@ -1966,6 +1966,15 @@ void WebRtcVoiceMediaChannel::ResetUnsignaledRecvStream() {
}
}
absl::optional<uint32_t> WebRtcVoiceMediaChannel::GetUnsignaledSsrc() const {
if (unsignaled_recv_ssrcs_.empty()) {
return absl::nullopt;
}
// In the event of multiple unsignaled ssrcs, the last in the vector will be
// the most recent one (the one forwarded to the MediaStreamTrack).
return unsignaled_recv_ssrcs_.back();
}
// Not implemented.
// TODO(https://crbug.com/webrtc/12676): Implement a fix for the unsignalled
// SSRC race that can happen when an m= section goes from receiving to not

1
third_party/libwebrtc/media/engine/webrtc_voice_engine.h поставляемый
Просмотреть файл

@ -173,6 +173,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
void ResetUnsignaledRecvStream() override;
absl::optional<uint32_t> GetUnsignaledSsrc() const override;
void OnDemuxerCriteriaUpdatePending() override;
void OnDemuxerCriteriaUpdateComplete() override;

4
third_party/libwebrtc/moz-patch-stack/0001.patch поставляемый
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@ -30,10 +30,10 @@ Cr-Branched-From: 2e1a9a4ae0234d4b1ea7a6fd4188afa1fb20379d-refs/heads/main@{#389
1 file changed, 10 insertions(+), 9 deletions(-)
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index da86afa52d..00ae88ab64 100644
index 78e7f4d96b..13c6590614 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -2189,16 +2189,17 @@ void RTCStatsCollector::ProduceTransportStats_n(
@@ -2185,16 +2185,17 @@ void RTCStatsCollector::ProduceTransportStats_n(
// exist.
const auto& certificate_stats_it =
transport_cert_stats.find(transport_name);

2
third_party/libwebrtc/moz-patch-stack/0095.patch поставляемый
Просмотреть файл

@ -204,7 +204,7 @@ index 120e411772..731989f820 100644
"../api:sequence_checker",
"../api:transport_api",
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 9dced9485e..5b6bcad524 100644
index 30e7571ab8..7f433bf69b 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -63,10 +63,6 @@ class Timing;

7
third_party/libwebrtc/pc/audio_rtp_receiver.cc поставляемый
Просмотреть файл

@ -212,9 +212,12 @@ void AudioRtpReceiver::SetupUnsignaledMediaChannel() {
RestartMediaChannel(absl::nullopt);
}
uint32_t AudioRtpReceiver::ssrc() const {
absl::optional<uint32_t> AudioRtpReceiver::ssrc() const {
RTC_DCHECK_RUN_ON(worker_thread_);
return ssrc_.value_or(0);
if (!ssrc_.has_value() && media_channel_) {
return media_channel_->GetUnsignaledSsrc();
}
return ssrc_;
}
void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {

2
third_party/libwebrtc/pc/audio_rtp_receiver.h поставляемый
Просмотреть файл

@ -100,7 +100,7 @@ class AudioRtpReceiver : public ObserverInterface,
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
uint32_t ssrc() const override;
absl::optional<uint32_t> ssrc() const override;
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(

2
third_party/libwebrtc/pc/legacy_stats_collector.cc поставляемый
Просмотреть файл

@ -1231,7 +1231,7 @@ void LegacyStatsCollector::ExtractMediaInfo(
for (const auto& receiver : transceiver->internal()->receivers()) {
gatherer->receiver_track_id_by_ssrc.insert(std::make_pair(
receiver->internal()->ssrc(), receiver->track()->id()));
receiver->internal()->ssrc().value_or(0), receiver->track()->id()));
}
}

58
third_party/libwebrtc/pc/rtc_stats_collector.cc поставляемый
Просмотреть файл

@ -1249,7 +1249,10 @@ void ProduceReceiverMediaTrackStats(
}
}
rtc::scoped_refptr<RTCStatsReport> CreateReportFilteredBySelector(
} // namespace
rtc::scoped_refptr<RTCStatsReport>
RTCStatsCollector::CreateReportFilteredBySelector(
bool filter_by_sender_selector,
rtc::scoped_refptr<const RTCStatsReport> report,
rtc::scoped_refptr<RtpSenderInternal> sender_selector,
@ -1258,40 +1261,35 @@ rtc::scoped_refptr<RTCStatsReport> CreateReportFilteredBySelector(
if (filter_by_sender_selector) {
// Filter mode: RTCStatsCollector::RequestInfo::kSenderSelector
if (sender_selector) {
// Find outbound-rtp(s) of the sender, i.e. the outbound-rtp(s) that
// reference the sender stats.
// Because we do not implement sender stats, we look at outbound-rtp(s)
// that reference the track attachment stats for the sender instead.
std::string track_id =
DEPRECATED_RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kDirectionOutbound, sender_selector->AttachmentId());
for (const auto& stats : *report) {
if (stats.type() != RTCOutboundRTPStreamStats::kType)
continue;
const auto& outbound_rtp = stats.cast_to<RTCOutboundRTPStreamStats>();
if (outbound_rtp.track_id.is_defined() &&
*outbound_rtp.track_id == track_id) {
rtpstream_ids.push_back(outbound_rtp.id());
// Find outbound-rtp(s) of the sender using ssrc lookup.
auto encodings = sender_selector->GetParametersInternal().encodings;
for (const auto* outbound_rtp :
report->GetStatsOfType<RTCOutboundRTPStreamStats>()) {
RTC_DCHECK(outbound_rtp->ssrc.is_defined());
auto it = std::find_if(
encodings.begin(), encodings.end(),
[ssrc =
*outbound_rtp->ssrc](const RtpEncodingParameters& encoding) {
return encoding.ssrc.has_value() && encoding.ssrc.value() == ssrc;
});
if (it != encodings.end()) {
rtpstream_ids.push_back(outbound_rtp->id());
}
}
}
} else {
// Filter mode: RTCStatsCollector::RequestInfo::kReceiverSelector
if (receiver_selector) {
// Find inbound-rtp(s) of the receiver, i.e. the inbound-rtp(s) that
// reference the receiver stats.
// Because we do not implement receiver stats, we look at inbound-rtp(s)
// that reference the track attachment stats for the receiver instead.
std::string track_id =
DEPRECATED_RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
kDirectionInbound, receiver_selector->AttachmentId());
for (const auto& stats : *report) {
if (stats.type() != RTCInboundRTPStreamStats::kType)
continue;
const auto& inbound_rtp = stats.cast_to<RTCInboundRTPStreamStats>();
if (inbound_rtp.track_id.is_defined() &&
*inbound_rtp.track_id == track_id) {
rtpstream_ids.push_back(inbound_rtp.id());
// Find the inbound-rtp of the receiver using ssrc lookup.
absl::optional<uint32_t> ssrc;
worker_thread_->BlockingCall([&] { ssrc = receiver_selector->ssrc(); });
if (ssrc.has_value()) {
for (const auto* inbound_rtp :
report->GetStatsOfType<RTCInboundRTPStreamStats>()) {
RTC_DCHECK(inbound_rtp->ssrc.is_defined());
if (*inbound_rtp->ssrc == *ssrc) {
rtpstream_ids.push_back(inbound_rtp->id());
}
}
}
}
@ -1301,8 +1299,6 @@ rtc::scoped_refptr<RTCStatsReport> CreateReportFilteredBySelector(
return TakeReferencedStats(report->Copy(), rtpstream_ids);
}
} // namespace
RTCStatsCollector::CertificateStatsPair
RTCStatsCollector::CertificateStatsPair::Copy() const {
CertificateStatsPair copy;

6
third_party/libwebrtc/pc/rtc_stats_collector.h поставляемый
Просмотреть файл

@ -244,6 +244,12 @@ class RTCStatsCollector : public rtc::RefCountInterface,
// This is a NO-OP if `network_report_` is null.
void MergeNetworkReport_s();
rtc::scoped_refptr<RTCStatsReport> CreateReportFilteredBySelector(
bool filter_by_sender_selector,
rtc::scoped_refptr<const RTCStatsReport> report,
rtc::scoped_refptr<RtpSenderInternal> sender_selector,
rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);
// Slots for signals (sigslot) that are wired up to `pc_`.
void OnSctpDataChannelCreated(SctpDataChannel* channel);
// Slots for signals (sigslot) that are wired up to `channel`.

8
third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc поставляемый
Просмотреть файл

@ -388,7 +388,10 @@ rtc::scoped_refptr<MockRtpSenderInternal> CreateMockSender(
EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track));
EXPECT_CALL(*sender, ssrc()).WillRepeatedly(Return(ssrc));
EXPECT_CALL(*sender, media_type()).WillRepeatedly(Return(media_type));
EXPECT_CALL(*sender, GetParameters()).WillRepeatedly(Invoke([ssrc]() {
EXPECT_CALL(*sender, GetParameters())
.WillRepeatedly(
Invoke([s = sender.get()]() { return s->GetParametersInternal(); }));
EXPECT_CALL(*sender, GetParametersInternal()).WillRepeatedly(Invoke([ssrc]() {
RtpParameters params;
params.encodings.push_back(RtpEncodingParameters());
params.encodings[0].ssrc = ssrc;
@ -406,6 +409,9 @@ rtc::scoped_refptr<MockRtpReceiverInternal> CreateMockReceiver(
int attachment_id) {
auto receiver = rtc::make_ref_counted<MockRtpReceiverInternal>();
EXPECT_CALL(*receiver, track()).WillRepeatedly(Return(track));
EXPECT_CALL(*receiver, ssrc()).WillRepeatedly(Invoke([ssrc]() {
return ssrc;
}));
EXPECT_CALL(*receiver, streams())
.WillRepeatedly(
Return(std::vector<rtc::scoped_refptr<MediaStreamInterface>>({})));

2
third_party/libwebrtc/pc/rtp_receiver.h поставляемый
Просмотреть файл

@ -68,7 +68,7 @@ class RtpReceiverInternal : public RtpReceiverInterface {
rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
// This SSRC is used as an identifier for the receiver between the API layer
// and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
virtual uint32_t ssrc() const = 0;
virtual absl::optional<uint32_t> ssrc() const = 0;
// Call this to notify the RtpReceiver when the first packet has been received
// on the corresponding channel.

2
third_party/libwebrtc/pc/test/mock_rtp_receiver_internal.h поставляемый
Просмотреть файл

@ -63,7 +63,7 @@ class MockRtpReceiverInternal : public RtpReceiverInternal {
(override));
MOCK_METHOD(void, SetupMediaChannel, (uint32_t), (override));
MOCK_METHOD(void, SetupUnsignaledMediaChannel, (), (override));
MOCK_METHOD(uint32_t, ssrc, (), (const, override));
MOCK_METHOD(absl::optional<uint32_t>, ssrc, (), (const, override));
MOCK_METHOD(void, NotifyFirstPacketReceived, (), (override));
MOCK_METHOD(void, set_stream_ids, (std::vector<std::string>), (override));
MOCK_METHOD(void,

4
third_party/libwebrtc/pc/test/mock_voice_media_channel.h поставляемый
Просмотреть файл

@ -53,6 +53,10 @@ class MockVoiceMediaChannel : public VoiceMediaChannel {
MOCK_METHOD(bool, AddRecvStream, (const StreamParams& sp), (override));
MOCK_METHOD(bool, RemoveRecvStream, (uint32_t ssrc), (override));
MOCK_METHOD(void, ResetUnsignaledRecvStream, (), (override));
MOCK_METHOD(absl::optional<uint32_t>,
GetUnsignaledSsrc,
(),
(const, override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdatePending, (), (override));
MOCK_METHOD(void, OnDemuxerCriteriaUpdateComplete, (), (override));
MOCK_METHOD(int, GetRtpSendTimeExtnId, (), (const, override));

7
third_party/libwebrtc/pc/video_rtp_receiver.cc поставляемый
Просмотреть файл

@ -184,9 +184,12 @@ void VideoRtpReceiver::SetupUnsignaledMediaChannel() {
RestartMediaChannel(absl::nullopt);
}
uint32_t VideoRtpReceiver::ssrc() const {
absl::optional<uint32_t> VideoRtpReceiver::ssrc() const {
RTC_DCHECK_RUN_ON(worker_thread_);
return ssrc_.value_or(0);
if (!ssrc_.has_value() && media_channel_) {
return media_channel_->GetUnsignaledSsrc();
}
return ssrc_;
}
void VideoRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {

2
third_party/libwebrtc/pc/video_rtp_receiver.h поставляемый
Просмотреть файл

@ -89,7 +89,7 @@ class VideoRtpReceiver : public RtpReceiverInternal {
void Stop() override;
void SetupMediaChannel(uint32_t ssrc) override;
void SetupUnsignaledMediaChannel() override;
uint32_t ssrc() const override;
absl::optional<uint32_t> ssrc() const override;
void NotifyFirstPacketReceived() override;
void set_stream_ids(std::vector<std::string> stream_ids) override;
void set_transport(