Store/restore RTP state for audio streams with same SSRC within a call

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index d178406b2972307b3d77998ebaa303b31c1aef35..22f5af36d2965c8aca40333518a8ba4286a66d6d 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -251,7 +251,10 @@ class Call : public webrtc::Call,
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
   std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
 
-  VideoSendStream::RtpStateMap suspended_video_send_ssrcs_;

[pbos-webrtc, 2017/05/22 15:40:18]

Does this mean that there's some VideoSendStream::RtpStateMap that should be
removed?

[ossu, 2017/05/22 17:13:59]

I chose to let RtpStateMap be a shorthand for std::map<uint32_t, RtpState> in
both Call and VideoSendStream, rather than find a common place to put it.

+  typedef std::map<uint32_t, RtpState> RtpStateMap;

[kwiberg-webrtc, 2017/05/19 01:05:09]

using?

[ossu, 2017/05/22 17:13:59]

Alright!

+  RtpStateMap suspended_audio_send_ssrcs_;

[pbos-webrtc, 2017/05/22 15:39:50]

Can you do GUARDED_BY (configuration_thread_checker_) for these? I recall us
having added thread checker static validation, which is useful now since this is
racy if DestroyAudioSendStream is ever called concurrently. GUARDED_BY enforces
that through the thread check.

Then replace CalledOnValidThread below with invocations of RTC_DCHECK_RUN_ON

[ossu, 2017/05/22 17:13:59]

Alright!

+  RtpStateMap suspended_video_send_ssrcs_;
+
   webrtc::RtcEventLog* event_log_;
 
   // The following members are only accessed (exclusively) from one thread and
@@ -509,9 +512,19 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   event_log_->LogAudioSendStreamConfig(config);
+
+  rtc::Optional<RtpState> suspended_rtp_state;
+  {
+    const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
+    if (iter != suspended_audio_send_ssrcs_.end()) {
+      suspended_rtp_state.emplace(iter->second);
+    }
+  }
+
   AudioSendStream* send_stream = new AudioSendStream(
       config, config_.audio_state, &worker_queue_, transport_send_.get(),
-      bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());
+      bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats(),
+      suspended_rtp_state);
   {
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
@@ -540,7 +553,8 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
 
   webrtc::internal::AudioSendStream* audio_send_stream =
       static_cast<webrtc::internal::AudioSendStream*>(send_stream);
-  uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
+  const uint32_t ssrc = audio_send_stream->config().rtp.ssrc;
+  suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState();
   {
     WriteLockScoped write_lock(*send_crit_);
     size_t num_deleted = audio_send_ssrcs_.erase(ssrc);