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

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 233 matching lines @@
   };
   std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
       GUARDED_BY(receive_crit_);
 
   std::unique_ptr<RWLockWrapper> send_crit_;
   // Audio and Video send streams are owned by the client that creates them.
   std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
   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
   // from the destructor, and therefore doesn't need any explicit
   // synchronization.
   RateCounter received_bytes_per_second_counter_;
   RateCounter received_audio_bytes_per_second_counter_;
   RateCounter received_video_bytes_per_second_counter_;
   RateCounter received_rtcp_bytes_per_second_counter_;
 
@@ skipping 237 matching lines @@
   // thread. Re-enable once that is fixed.
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   return this;
 }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   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) ==
                audio_send_ssrcs_.end());
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
   }
   {
     ReadLockScoped read_lock(*receive_crit_);
     for (const auto& kv : audio_receive_ssrcs_) {
       if (kv.second->config().rtp.local_ssrc == config.rtp.ssrc) {
         kv.second->AssociateSendStream(send_stream);
       }
     }
   }
   send_stream->SignalNetworkState(audio_network_state_);
   UpdateAggregateNetworkState();
   return send_stream;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(send_stream != nullptr);
 
   send_stream->Stop();
 
   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);
     RTC_DCHECK_EQ(1, num_deleted);
   }
   {
     ReadLockScoped read_lock(*receive_crit_);
     for (const auto& kv : audio_receive_ssrcs_) {
       if (kv.second->config().rtp.local_ssrc == ssrc) {
         kv.second->AssociateSendStream(nullptr);
@@ skipping 741 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc