Move responsibility for RTP header extensions on video receive.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index c7e49dee1a2ef587052eadd88edd406ba74bd218..aa888fb6520a1f6f23308af49fdbb4d8d08ef090 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -63,11 +63,7 @@ const int Call::Config::kDefaultStartBitrateBps = 300000;
 
 namespace {
 
-// TODO(nisse): This really begs for a shared context struct.
-bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
-                    bool transport_cc) {
-  if (!transport_cc)
-    return false;
+bool UseTransportSeqno(const std::vector<RtpExtension>& extensions) {
   for (const auto& extension : extensions) {
     if (extension.uri == RtpExtension::kTransportSequenceNumberUri)
       return true;
@@ -75,18 +71,16 @@ bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
   return false;
 }
 
-bool UseSendSideBwe(const VideoReceiveStream::Config& config) {
-  return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
+// TODO(nisse): This really begs for a shared context struct.
+bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
+                    bool transport_cc) {
+  return transport_cc && UseTransportSeqno(extensions);
 }
 
 bool UseSendSideBwe(const AudioReceiveStream::Config& config) {
   return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
 }
 
-bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
-  return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
-}
-
 class RtpTransportControllerSend : public RtpTransportControllerSendInterface {
  public:
   RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
@@ -138,6 +132,9 @@ class Call : public webrtc::Call,
   // Implements webrtc::Call.
   PacketReceiver* Receiver() override;
 
+  void SetVideoReceiveRtpHeaderExtensions(
+      const std::vector<RtpExtension>& extensions) override;
+
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
   void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
@@ -255,6 +252,9 @@ class Call : public webrtc::Call,
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
       GUARDED_BY(receive_crit_);
 
+  std::vector<RtpExtension> video_rtp_header_extensions_
+      GUARDED_BY(receive_crit_);
+
   // This extra map is used for receive processing which is
   // independent of media type.
 
@@ -533,6 +533,12 @@ PacketReceiver* Call::Receiver() {
   return this;
 }
 
+void Call::SetVideoReceiveRtpHeaderExtensions(
+    const std::vector<RtpExtension>& extensions) {
+  WriteLockScoped write_lock(*receive_crit_);
+  video_rtp_header_extensions_ = extensions;
+}
+
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
@@ -724,12 +730,15 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
                              module_process_thread_.get(), call_stats_.get());
 
   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
-  ReceiveRtpConfig receive_config(config.rtp.extensions,
-                                  UseSendSideBwe(config));
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                video_receive_ssrcs_.end());
+    ReceiveRtpConfig receive_config(
+        video_rtp_header_extensions_,
+        config.rtp.transport_cc &&
+        UseTransportSeqno(video_rtp_header_extensions_));
+
     video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     if (config.rtp.rtx_ssrc) {
       video_receive_ssrcs_[config.rtp.rtx_ssrc] = receive_stream;
@@ -755,6 +764,7 @@ void Call::DestroyVideoReceiveStream(
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(receive_stream != nullptr);
   VideoReceiveStream* receive_stream_impl = nullptr;
+  bool use_send_side_bwe = false;
   {
     WriteLockScoped write_lock(*receive_crit_);
     // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
@@ -762,8 +772,10 @@ void Call::DestroyVideoReceiveStream(
     auto it = video_receive_ssrcs_.begin();
     while (it != video_receive_ssrcs_.end()) {
       if (it->second == static_cast<VideoReceiveStream*>(receive_stream)) {
-        if (receive_stream_impl != nullptr)
+        if (receive_stream_impl != nullptr) {
           RTC_DCHECK(receive_stream_impl == it->second);
+          use_send_side_bwe = receive_rtp_config_[it->first].use_send_side_bwe;
+        }
         receive_stream_impl = it->second;
         receive_rtp_config_.erase(it->first);
         it = video_receive_ssrcs_.erase(it);
@@ -771,13 +783,13 @@ void Call::DestroyVideoReceiveStream(
         ++it;
       }
     }
-    video_receive_streams_.erase(receive_stream_impl);
     RTC_CHECK(receive_stream_impl != nullptr);
+    video_receive_streams_.erase(receive_stream_impl);
     ConfigureSync(receive_stream_impl->config().sync_group);
   }
   const VideoReceiveStream::Config& config = receive_stream_impl->config();
 
-  receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
+  receive_side_cc_.GetRemoteBitrateEstimator(use_send_side_bwe)
       ->RemoveStream(config.rtp.remote_ssrc);
 
   UpdateAggregateNetworkState();
@@ -811,7 +823,9 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
     RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) ==
                receive_rtp_config_.end());
     receive_rtp_config_[config.remote_ssrc] =
-        ReceiveRtpConfig(config.rtp_header_extensions, UseSendSideBwe(config));
+        ReceiveRtpConfig(video_rtp_header_extensions_,
+                         config.transport_cc &&
+                         UseTransportSeqno(video_rtp_header_extensions_));
   }
 
   // TODO(brandtr): Store config in RtcEventLog here.
@@ -834,6 +848,9 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
     const FlexfecReceiveStream::Config& config =
         receive_stream_impl->GetConfig();
     uint32_t ssrc = config.remote_ssrc;
+    receive_side_cc_.GetRemoteBitrateEstimator(
+        receive_rtp_config_[ssrc].use_send_side_bwe)
+        ->RemoveStream(ssrc);
     receive_rtp_config_.erase(ssrc);
 
     // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
@@ -853,9 +870,6 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
         ++media_it;
     }
 
-    receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
-        ->RemoveStream(ssrc);
-
     flexfec_receive_streams_.erase(receive_stream_impl);
   }