Separate reading remote_ssrc from using the rtp_config() getter.

`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.

Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36818}

audio/audio_receive_stream.h

@@ -128,7 +128,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
 
   uint32_t local_ssrc() const;
 
-  uint32_t remote_ssrc() const {
+  uint32_t remote_ssrc() const override {
     // The remote_ssrc member variable of config_ will never change and can be
     // considered const.
     return config_.rtp.remote_ssrc;

call/audio_receive_stream.h

@@ -194,6 +194,11 @@ class AudioReceiveStream : public MediaReceiveStream {
   // Returns current value of base minimum delay in milliseconds.
   virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
 
+  // Synchronization source (stream identifier) to be received.
+  // This member will not change mid-stream and can be assumed to be const
+  // post initialization.
+  virtual uint32_t remote_ssrc() const = 0;
+
  protected:
   virtual ~AudioReceiveStream() {}
 };

call/call.cc

@@ -1245,16 +1245,17 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
   // TODO(bugs.webrtc.org/11993): Unregister on the network thread.
   receive_stream_impl->UnregisterFromTransport();
 
-  RTC_DCHECK(receive_stream != nullptr);
-  const FlexfecReceiveStream::RtpConfig& rtp = receive_stream->rtp_config();
-  UnregisterReceiveStream(rtp.remote_ssrc);
+  auto ssrc = receive_stream_impl->remote_ssrc();
+  UnregisterReceiveStream(ssrc);
 
   // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
   // destroyed.
-  receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(rtp))
-      ->RemoveStream(rtp.remote_ssrc);
+  receive_side_cc_
+      .GetRemoteBitrateEstimator(
+          UseSendSideBwe(receive_stream_impl->rtp_config()))
+      ->RemoveStream(ssrc);
 
-  delete receive_stream;
+  delete receive_stream_impl;
 }
 
 void Call::AddAdaptationResource(rtc::scoped_refptr<Resource> resource) {

call/flexfec_receive_stream_impl.cc

@@ -204,7 +204,10 @@ FlexfecReceiveStreamImpl::Stats FlexfecReceiveStreamImpl::GetStats() const {
 void FlexfecReceiveStreamImpl::SetRtpExtensions(
     std::vector<RtpExtension> extensions) {
   RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
-  config_.rtp.extensions = std::move(extensions);
+  // TODO(tommi): Remove this cast once header extensions are managed outside
+  // of the config struct.
+  const_cast<std::vector<RtpExtension>&>(config_.rtp.extensions) =
+      std::move(extensions);
 }
 
 }  // namespace webrtc

call/flexfec_receive_stream_impl.h

@@ -61,12 +61,15 @@ class FlexfecReceiveStreamImpl : public FlexfecReceiveStream {
   // ReceiveStream impl.
   void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
   const RtpConfig& rtp_config() const override { return config_.rtp; }
+  uint32_t remote_ssrc() const { return config_.rtp.remote_ssrc; }
 
  private:
   RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_;
 
-  // Config. Mostly const, header extensions may change.
-  Config config_ RTC_GUARDED_BY(packet_sequence_checker_);
+  // Config. Mostly const, header extensions may change, which is an exception
+  // case that's specifically handled in `SetRtpExtensions`, which must be
+  // called on the `packet_sequence_checker` thread.
+  const Config config_;
 
   // Erasure code interfacing.
   const std::unique_ptr<FlexfecReceiver> receiver_;

media/engine/fake_webrtc_call.h

@@ -113,6 +113,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   const webrtc::ReceiveStream::RtpConfig& rtp_config() const override {
     return config_.rtp;
   }
+  uint32_t remote_ssrc() const override { return config_.rtp.remote_ssrc; }
   void Start() override { started_ = true; }
   void Stop() override { started_ = false; }
   bool IsRunning() const override { return started_; }
@@ -304,6 +305,8 @@ class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
 
   const webrtc::FlexfecReceiveStream::Config& GetConfig() const;
 
+  uint32_t remote_ssrc() const { return config_.rtp.remote_ssrc; }
+
  private:
   webrtc::FlexfecReceiveStream::Stats GetStats() const override;
 

media/engine/webrtc_video_engine_unittest.cc

@@ -5041,7 +5041,7 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetRecvParamsWithoutFecDisablesFec) {
   ASSERT_EQ(1U, streams.size());
   const FakeFlexfecReceiveStream* stream = streams.front();
   EXPECT_EQ(GetEngineCodec("flexfec-03").id, stream->GetConfig().payload_type);
-  EXPECT_EQ(kFlexfecSsrc, stream->rtp_config().remote_ssrc);
+  EXPECT_EQ(kFlexfecSsrc, stream->remote_ssrc());
   ASSERT_EQ(1U, stream->GetConfig().protected_media_ssrcs.size());
   EXPECT_EQ(kSsrcs1[0], stream->GetConfig().protected_media_ssrcs[0]);
 

media/engine/webrtc_voice_engine.cc

@@ -1244,7 +1244,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
 
     RTC_LOG(LS_ERROR) << "Failed to SetBaseMinimumPlayoutDelayMs"
                          " on AudioReceiveStream on SSRC="
-                      << stream_->rtp_config().remote_ssrc
+                      << stream_->remote_ssrc()
                       << " with delay_ms=" << delay_ms;
     return false;
   }
@@ -1263,7 +1263,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
     webrtc::RtpParameters rtp_parameters;
     rtp_parameters.encodings.emplace_back();
     const auto& config = stream_->rtp_config();
-    rtp_parameters.encodings[0].ssrc = config.remote_ssrc;
+    rtp_parameters.encodings[0].ssrc = stream_->remote_ssrc();
     rtp_parameters.header_extensions = config.extensions;
     return rtp_parameters;
   }