Add getter for rtp header extensions for receiver classes.

This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.

Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36820}

audio/audio_receive_stream.cc

@@ -265,6 +265,11 @@ void AudioReceiveStream::SetRtpExtensions(
   config_.rtp.extensions = std::move(extensions);
 }
 
+const std::vector<RtpExtension>& AudioReceiveStream::GetRtpExtensions() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return config_.rtp.extensions;
+}
+
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats(
     bool get_and_clear_legacy_stats) const {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);

audio/audio_receive_stream.h

@@ -95,6 +95,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
   void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
                              frame_decryptor) override;
   void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
+  const std::vector<RtpExtension>& GetRtpExtensions() const override;
 
   webrtc::AudioReceiveStream::Stats GetStats(
       bool get_and_clear_legacy_stats) const override;

call/call.cc

@@ -79,10 +79,10 @@ bool SendPeriodicFeedback(const std::vector<RtpExtension>& extensions) {
   return true;
 }
 
-bool UseSendSideBwe(const ReceiveStream::RtpConfig& rtp) {
-  if (!rtp.transport_cc)
+bool UseSendSideBwe(const ReceiveStream* stream) {
+  if (!stream->rtp_config().transport_cc)
     return false;
-  for (const auto& extension : rtp.extensions) {
+  for (const auto& extension : stream->GetRtpExtensions()) {
     if (extension.uri == RtpExtension::kTransportSequenceNumberUri ||
         extension.uri == RtpExtension::kTransportSequenceNumberV2Uri)
       return true;
@@ -1010,8 +1010,7 @@ void Call::DestroyAudioReceiveStream(
 
   uint32_t ssrc = audio_receive_stream->remote_ssrc();
   receive_side_cc_
-      .GetRemoteBitrateEstimator(
-          UseSendSideBwe(audio_receive_stream->rtp_config()))
+      .GetRemoteBitrateEstimator(UseSendSideBwe(audio_receive_stream))
       ->RemoveStream(ssrc);
 
   audio_receive_streams_.erase(audio_receive_stream);
@@ -1189,6 +1188,7 @@ void Call::DestroyVideoReceiveStream(
   // TODO(bugs.webrtc.org/11993): Unregister on the network thread.
   receive_stream_impl->UnregisterFromTransport();
 
+  // TODO(tommi): Remove `rtp()` accessor.
   const webrtc::VideoReceiveStream::Config::Rtp& rtp =
       receive_stream_impl->rtp();
 
@@ -1201,7 +1201,8 @@ void Call::DestroyVideoReceiveStream(
   video_receive_streams_.erase(receive_stream_impl);
   ConfigureSync(receive_stream_impl->sync_group());
 
-  receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(rtp))
+  receive_side_cc_
+      .GetRemoteBitrateEstimator(UseSendSideBwe(receive_stream_impl))
       ->RemoveStream(rtp.remote_ssrc);
 
   UpdateAggregateNetworkState();
@@ -1251,8 +1252,7 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
   // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
   // destroyed.
   receive_side_cc_
-      .GetRemoteBitrateEstimator(
-          UseSendSideBwe(receive_stream_impl->rtp_config()))
+      .GetRemoteBitrateEstimator(UseSendSideBwe(receive_stream_impl))
       ->RemoveStream(ssrc);
 
   delete receive_stream_impl;
@@ -1694,10 +1694,10 @@ bool Call::IdentifyReceivedPacket(RtpPacketReceived& packet,
   }
 
   packet.IdentifyExtensions(
-      RtpHeaderExtensionMap(it->second->rtp_config().extensions));
+      RtpHeaderExtensionMap(it->second->GetRtpExtensions()));
 
   if (use_send_side_bwe) {
-    *use_send_side_bwe = UseSendSideBwe(it->second->rtp_config());
+    *use_send_side_bwe = UseSendSideBwe(it->second);
   }
 
   return true;

call/flexfec_receive_stream_impl.cc

@@ -210,4 +210,10 @@ void FlexfecReceiveStreamImpl::SetRtpExtensions(
       std::move(extensions);
 }
 
+const std::vector<RtpExtension>& FlexfecReceiveStreamImpl::GetRtpExtensions()
+    const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return config_.rtp.extensions;
+}
+
 }  // namespace webrtc

call/flexfec_receive_stream_impl.h

@@ -60,6 +60,7 @@ 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; }
 

call/receive_stream.h

@@ -55,6 +55,11 @@ class ReceiveStream {
   // delivery thread.
   virtual void SetRtpExtensions(std::vector<RtpExtension> extensions) = 0;
 
+  // Access the currently set rtp extensions. Must be called on the packet
+  // delivery thread.
+  // 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

media/engine/fake_webrtc_call.cc

@@ -130,6 +130,11 @@ void FakeAudioReceiveStream::SetRtpExtensions(
   config_.rtp.extensions = std::move(extensions);
 }
 
+const std::vector<webrtc::RtpExtension>&
+FakeAudioReceiveStream::GetRtpExtensions() const {
+  return config_.rtp.extensions;
+}
+
 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats(
     bool get_and_clear_legacy_stats) const {
   return stats_;
@@ -380,6 +385,11 @@ void FakeVideoReceiveStream::SetRtpExtensions(
   config_.rtp.extensions = std::move(extensions);
 }
 
+const std::vector<webrtc::RtpExtension>&
+FakeVideoReceiveStream::GetRtpExtensions() const {
+  return config_.rtp.extensions;
+}
+
 void FakeVideoReceiveStream::Start() {
   receiving_ = true;
 }
@@ -402,6 +412,11 @@ void FakeFlexfecReceiveStream::SetRtpExtensions(
   config_.rtp.extensions = std::move(extensions);
 }
 
+const std::vector<webrtc::RtpExtension>&
+FakeFlexfecReceiveStream::GetRtpExtensions() const {
+  return config_.rtp.extensions;
+}
+
 const webrtc::FlexfecReceiveStream::Config&
 FakeFlexfecReceiveStream::GetConfig() const {
   return config_;

media/engine/fake_webrtc_call.h

@@ -128,6 +128,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
                              frame_decryptor) override;
   void SetRtpExtensions(std::vector<webrtc::RtpExtension> extensions) override;
+  const std::vector<webrtc::RtpExtension>& GetRtpExtensions() const override;
 
   webrtc::AudioReceiveStream::Stats GetStats(
       bool get_and_clear_legacy_stats) const override;
@@ -266,6 +267,7 @@ class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream {
  private:
   // 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;
@@ -298,6 +300,7 @@ class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
       const webrtc::FlexfecReceiveStream::Config& config);
 
   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;

media/engine/webrtc_voice_engine.cc

@@ -1262,9 +1262,8 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
   webrtc::RtpParameters GetRtpParameters() const {
     webrtc::RtpParameters rtp_parameters;
     rtp_parameters.encodings.emplace_back();
-    const auto& config = stream_->rtp_config();
     rtp_parameters.encodings[0].ssrc = stream_->remote_ssrc();
-    rtp_parameters.header_extensions = config.extensions;
+    rtp_parameters.header_extensions = stream_->GetRtpExtensions();
     return rtp_parameters;
   }
 

video/video_receive_stream2.cc

@@ -487,6 +487,12 @@ void VideoReceiveStream2::SetRtpExtensions(
   c.rtp.extensions = std::move(extensions);
 }
 
+const std::vector<RtpExtension>& VideoReceiveStream2::GetRtpExtensions() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  // TODO(tommi): return the state held by `rtp_video_stream_receiver_`.
+  return config_.rtp.extensions;
+}
+
 void VideoReceiveStream2::CreateAndRegisterExternalDecoder(
     const Decoder& decoder) {
   TRACE_EVENT0("webrtc",

video/video_receive_stream2.h

@@ -135,6 +135,7 @@ class VideoReceiveStream2
   void Stop() override;
 
   void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
+  const std::vector<RtpExtension>& GetRtpExtensions() const override;
 
   const RtpConfig& rtp_config() const override { return rtp(); }