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Bug 1766646 - Vendor libwebrtc from 8695282243 

Upstream commit: https://webrtc.googlesource.com/src/+/8695282243a5bf232ad1c21c9799620a955b32ce
    Remove unnecessary copy of suspended_ssrcs.

    Also removing pass-by-value in ctor.

    Bug: none
    Change-Id: I09e36fd955c8f306c4a347d8befc6eea38384cb9
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239183
    Auto-Submit: Tommi <tommi@webrtc.org>
    Reviewed-by: Niels Moller <nisse@webrtc.org>
    Commit-Queue: Tommi <tommi@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#35427}
This commit is contained in:
Michael Froman 2022-06-13 11:13:33 -05:00 коммит произвёл Connor Sheehan
Родитель 75672e7348
Коммит 3f8b1a20e1
8 изменённых файлов: 20 добавлений и 17 удалений
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3
third_party/libwebrtc/README.moz-ff-commit поставляемый
Просмотреть файл

@ -9819,3 +9819,6 @@ ef5b21e637
# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
# base of lastest vendoring
9345bee860
# MOZ_LIBWEBRTC_SRC=/home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src MOZ_LIBWEBRTC_COMMIT=mjfdev bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
# base of lastest vendoring
8695282243

2
third_party/libwebrtc/README.mozilla поставляемый
Просмотреть файл

@ -6554,3 +6554,5 @@ libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwe
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-13T16:11:43.584498.
# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-13T16:12:46.987340.
# python3 vendor-libwebrtc.py --from-local /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src --commit mjfdev libwebrtc
libwebrtc updated from /home/mfroman/git-checkouts/trial-webrtc-builds/moz-libwebrtc-checkout/src commit mjfdev on 2022-06-13T16:13:28.173187.

2
third_party/libwebrtc/call/rtp_transport_controller_send.cc поставляемый
Просмотреть файл

@ -156,7 +156,7 @@ RtpTransportControllerSend::~RtpTransportControllerSend() {
}
RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,

2
third_party/libwebrtc/call/rtp_transport_controller_send.h поставляемый
Просмотреть файл

@ -65,7 +65,7 @@ class RtpTransportControllerSend final
// TODO(tommi): Change to std::unique_ptr<>.
RtpVideoSenderInterface* CreateRtpVideoSender(
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>&
states, // move states into RtpTransportControllerSend
const RtpConfig& rtp_config,

2
third_party/libwebrtc/call/rtp_transport_controller_send_interface.h поставляемый
Просмотреть файл

@ -96,7 +96,7 @@ class RtpTransportControllerSendInterface {
virtual PacketRouter* packet_router() = 0;
virtual RtpVideoSenderInterface* CreateRtpVideoSender(
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
// TODO(holmer): Move states into RtpTransportControllerSend.
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,

18
third_party/libwebrtc/call/rtp_video_sender.cc поставляемый
Просмотреть файл

@ -347,7 +347,7 @@ bool IsFirstFrameOfACodedVideoSequence(
RtpVideoSender::RtpVideoSender(
Clock* clock,
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
@ -371,7 +371,6 @@ RtpVideoSender::RtpVideoSender(
field_trials_.Lookup("WebRTC-GenericCodecDependencyDescriptor"),
"Enabled")),
active_(false),
suspended_ssrcs_(std::move(suspended_ssrcs)),
fec_controller_(std::move(fec_controller)),
fec_allowed_(true),
rtp_streams_(CreateRtpStreamSenders(clock,
@ -381,7 +380,7 @@ RtpVideoSender::RtpVideoSender(
send_transport,
transport->GetBandwidthObserver(),
transport,
suspended_ssrcs_,
suspended_ssrcs,
event_log,
retransmission_limiter,
frame_encryptor,
@ -421,7 +420,7 @@ RtpVideoSender::RtpVideoSender(
}
}
ConfigureSsrcs();
ConfigureSsrcs(suspended_ssrcs);
ConfigureRids();
if (!rtp_config_.mid.empty()) {
@ -683,7 +682,8 @@ void RtpVideoSender::DeliverRtcp(const uint8_t* packet, size_t length) {
stream.rtp_rtcp->IncomingRtcpPacket(packet, length);
}
void RtpVideoSender::ConfigureSsrcs() {
void RtpVideoSender::ConfigureSsrcs(
const std::map<uint32_t, RtpState>& suspended_ssrcs) {
// Configure regular SSRCs.
RTC_CHECK(ssrc_to_rtp_module_.empty());
for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
@ -691,8 +691,8 @@ void RtpVideoSender::ConfigureSsrcs() {
RtpRtcpInterface* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
// Restore RTP state if previous existed.
auto it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
auto it = suspended_ssrcs.find(ssrc);
if (it != suspended_ssrcs.end())
rtp_rtcp->SetRtpState(it->second);
ssrc_to_rtp_module_[ssrc] = rtp_rtcp;
@ -706,8 +706,8 @@ void RtpVideoSender::ConfigureSsrcs() {
for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config_.rtx.ssrcs[i];
RtpRtcpInterface* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
auto it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
auto it = suspended_ssrcs.find(ssrc);
if (it != suspended_ssrcs.end())
rtp_rtcp->SetRtxState(it->second);
}

6
third_party/libwebrtc/call/rtp_video_sender.h поставляемый
Просмотреть файл

@ -74,7 +74,7 @@ class RtpVideoSender : public RtpVideoSenderInterface,
// Rtp modules are assumed to be sorted in simulcast index order.
RtpVideoSender(
Clock* clock,
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
@ -155,7 +155,7 @@ class RtpVideoSender : public RtpVideoSenderInterface,
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
void ConfigureProtection();
void ConfigureSsrcs();
void ConfigureSsrcs(const std::map<uint32_t, RtpState>& suspended_ssrcs);
void ConfigureRids();
bool NackEnabled() const;
uint32_t GetPacketizationOverheadRate() const;
@ -175,8 +175,6 @@ class RtpVideoSender : public RtpVideoSenderInterface,
mutable Mutex mutex_;
bool active_ RTC_GUARDED_BY(mutex_);
std::map<uint32_t, RtpState> suspended_ssrcs_;
const std::unique_ptr<FecController> fec_controller_;
bool fec_allowed_ RTC_GUARDED_BY(mutex_);

2
third_party/libwebrtc/call/test/mock_rtp_transport_controller_send.h поставляемый
Просмотреть файл

@ -34,7 +34,7 @@ class MockRtpTransportControllerSend
public:
MOCK_METHOD(RtpVideoSenderInterface*,
CreateRtpVideoSender,
((std::map<uint32_t, RtpState>),
((const std::map<uint32_t, RtpState>&),
(const std::map<uint32_t, RtpPayloadState>&),
const RtpConfig&,
int rtcp_report_interval_ms,