Bug 1766646 - Vendor libwebrtc from 7a15ff3f14

Upstream commit: https://webrtc.googlesource.com/src/+/7a15ff3f1412fc640bb92c8aab511640a4513bbe
    Add a transport_cc() getter and remove rtp_config().

    Bug: webrtc:11993
    Change-Id: Ie435a702c91b4d3827e528083f474e378fc75cc5
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261318
    Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
    Reviewed-by: Niels Moller <nisse@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#36822}

third_party/libwebrtc/README.moz-ff-commit

@@ -14004,3 +14004,6 @@ cb7c7366d0
 # 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
 cc001e2d2a
+# 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
+7a15ff3f14

third_party/libwebrtc/README.mozilla

@@ -9348,3 +9348,5 @@ 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 2022-07-14T22:08:48.892943.
 # 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 2022-07-14T22:31:57.810733.
+# 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 2022-07-14T22:32:37.214682.

third_party/libwebrtc/audio/audio_receive_stream.cc

@@ -213,6 +213,11 @@ void AudioReceiveStream::Stop() {
   audio_state()->RemoveReceivingStream(this);
 }
 
+bool AudioReceiveStream::transport_cc() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return config_.rtp.transport_cc;
+}
+
 bool AudioReceiveStream::IsRunning() const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   return playing_;

third_party/libwebrtc/audio/audio_receive_stream.h

@@ -83,7 +83,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   // webrtc::AudioReceiveStream implementation.
   void Start() override;
   void Stop() override;
-  const RtpConfig& rtp_config() const override { return config_.rtp; }
+  bool transport_cc() const override;
   bool IsRunning() const override;
   void SetDepacketizerToDecoderFrameTransformer(
       rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)

third_party/libwebrtc/call/audio_receive_stream.h

@@ -109,7 +109,7 @@ class AudioReceiveStream : public MediaReceiveStream {
     std::string ToString() const;
 
     // Receive-stream specific RTP settings.
-    struct Rtp : public RtpConfig {
+    struct Rtp : public ReceiveStreamRtpConfig {
       Rtp();
       ~Rtp();
 

third_party/libwebrtc/call/call.cc

@@ -80,7 +80,7 @@ bool SendPeriodicFeedback(const std::vector<RtpExtension>& extensions) {
 }
 
 bool UseSendSideBwe(const ReceiveStream* stream) {
-  if (!stream->rtp_config().transport_cc)
+  if (!stream->transport_cc())
     return false;
   for (const auto& extension : stream->GetRtpExtensions()) {
     if (extension.uri == RtpExtension::kTransportSequenceNumberUri ||

third_party/libwebrtc/call/flexfec_receive_stream.h

@@ -50,7 +50,7 @@ class FlexfecReceiveStream : public RtpPacketSinkInterface,
     // Payload type for FlexFEC.
     int payload_type = -1;
 
-    RtpConfig rtp;
+    ReceiveStreamRtpConfig rtp;
 
     // Vector containing a single element, corresponding to the SSRC of the
     // media stream being protected by this FlexFEC stream. The vector MUST have

third_party/libwebrtc/call/flexfec_receive_stream_impl.h

@@ -61,8 +61,11 @@ class FlexfecReceiveStreamImpl : public FlexfecReceiveStream {
   // ReceiveStream impl.
   void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
   const std::vector<RtpExtension>& GetRtpExtensions() const override;
-  const RtpConfig& rtp_config() const override { return config_.rtp; }
   uint32_t remote_ssrc() const { return config_.rtp.remote_ssrc; }
+  bool transport_cc() const override {
+    RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+    return config_.rtp.transport_cc;
+  }
 
  private:
   RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_;

third_party/libwebrtc/call/receive_stream.h

@@ -26,7 +26,9 @@ namespace webrtc {
 class ReceiveStream {
  public:
   // Receive-stream specific RTP settings.
-  struct RtpConfig {
+  // TODO(tommi): This struct isn't needed at this level anymore. Move it closer
+  // to where it's used.
+  struct ReceiveStreamRtpConfig {
     // Synchronization source (stream identifier) to be received.
     // This member will not change mid-stream and can be assumed to be const
     // post initialization.
@@ -60,11 +62,13 @@ class ReceiveStream {
   // TODO(tommi): Consider using `RtpHeaderExtensionMap` instead.
   virtual const std::vector<RtpExtension>& GetRtpExtensions() const = 0;
 
-  // Called on the packet delivery thread since some members of the config may
-  // change mid-stream (e.g. the local ssrc). All mutation must also happen on
-  // the packet delivery thread. Return value can be assumed to
-  // only be used in the calling context (on the stack basically).
-  virtual const RtpConfig& rtp_config() const = 0;
+  // Returns a bool for whether feedback for send side bandwidth estimation is
+  // enabled. See
+  // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
+  // for details.
+  // This value may change mid-stream and must be done on the same thread
+  // that the value is read on (i.e. packet delivery).
+  virtual bool transport_cc() const = 0;
 
  protected:
   virtual ~ReceiveStream() {}

third_party/libwebrtc/call/video_receive_stream.h

@@ -183,7 +183,7 @@ class VideoReceiveStream : public MediaReceiveStream {
     VideoDecoderFactory* decoder_factory = nullptr;
 
     // Receive-stream specific RTP settings.
-    struct Rtp : public RtpConfig {
+    struct Rtp : public ReceiveStreamRtpConfig {
       Rtp();
       Rtp(const Rtp&);
       ~Rtp();

third_party/libwebrtc/media/engine/fake_webrtc_call.h

@@ -110,9 +110,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   }
 
  private:
-  const webrtc::ReceiveStream::RtpConfig& rtp_config() const override {
-    return config_.rtp;
-  }
+  bool transport_cc() const override { return config_.rtp.transport_cc; }
   uint32_t remote_ssrc() const override { return config_.rtp.remote_ssrc; }
   void Start() override { started_ = true; }
   void Stop() override { started_ = false; }
@@ -268,10 +266,7 @@ class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream {
   // webrtc::VideoReceiveStream implementation.
   void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
   const std::vector<webrtc::RtpExtension>& GetRtpExtensions() const override;
-
-  const webrtc::ReceiveStream::RtpConfig& rtp_config() const override {
-    return config_.rtp;
-  }
+  bool transport_cc() const override { return config_.rtp.transport_cc; }
 
   void Start() override;
   void Stop() override;
@@ -301,10 +296,7 @@ class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
 
   void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
   const std::vector<webrtc::RtpExtension>& GetRtpExtensions() const override;
-
-  const webrtc::ReceiveStream::RtpConfig& rtp_config() const override {
-    return config_.rtp;
-  }
+  bool transport_cc() const override { return config_.rtp.transport_cc; }
 
   const webrtc::FlexfecReceiveStream::Config& GetConfig() const;
 

third_party/libwebrtc/video/video_receive_stream2.h

@@ -136,8 +136,7 @@ class VideoReceiveStream2
 
   void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
   const std::vector<RtpExtension>& GetRtpExtensions() const override;
-
-  const RtpConfig& rtp_config() const override { return rtp(); }
+  bool transport_cc() const override { return rtp().transport_cc; }
 
   webrtc::VideoReceiveStream::Stats GetStats() const override;