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

webrtc/audio/audio_send_stream.cc

 /*
  *  Copyright (c) 2015 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 10 matching lines @@
 #include "webrtc/base/event.h"
 #include "webrtc/base/function_view.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
 namespace internal {
 // TODO(elad.alon): Subsequent CL will make these values experiment-dependent.
 constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
@@ skipping 10 matching lines @@
 }
 }  // namespace
 
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     rtc::TaskQueue* worker_queue,
     RtpTransportControllerSendInterface* transport,
     BitrateAllocator* bitrate_allocator,
     RtcEventLog* event_log,
-    RtcpRttStats* rtcp_rtt_stats)
+    RtcpRttStats* rtcp_rtt_stats,
+    const rtc::Optional<RtpState>& suspended_rtp_state)
     : worker_queue_(worker_queue),
       config_(Config(nullptr)),
       audio_state_(audio_state),
       event_log_(event_log),
       bitrate_allocator_(bitrate_allocator),
       transport_(transport),
       packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
                            kPacketLossRateMinNumAckedPackets,
-                           kRecoverablePacketLossRateMinNumAckedPairs) {
+                           kRecoverablePacketLossRateMinNumAckedPairs),
+      rtp_rtcp_module_(nullptr),
+      suspended_rtp_state_(suspended_rtp_state) {
   LOG(LS_INFO) << "AudioSendStream: " << config.ToString();
   RTC_DCHECK_NE(config.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(transport);
   RTC_DCHECK(transport->send_side_cc());
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log_);
   channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
   channel_proxy_->SetRTCPStatus(true);
   transport_->send_side_cc()->RegisterPacketFeedbackObserver(this);
+  RtpReceiver* rtpReceiver = nullptr;  // Unused, but required for call.
+  channel_proxy_->GetRtpRtcp(&rtp_rtcp_module_, &rtpReceiver);
+  RTC_DCHECK(rtp_rtcp_module_);
 
   ConfigureStream(this, config, true);
 
   pacer_thread_checker_.DetachFromThread();
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   transport_->send_side_cc()->DeRegisterPacketFeedbackObserver(this);
@@ skipping 11 matching lines @@
 void AudioSendStream::ConfigureStream(
     webrtc::internal::AudioSendStream* stream,
     const webrtc::AudioSendStream::Config& new_config,
     bool first_time) {
   LOG(LS_INFO) << "AudioSendStream::Configuring: " << new_config.ToString();
   const auto& channel_proxy = stream->channel_proxy_;
   const auto& old_config = stream->config_;
 
   if (first_time || old_config.rtp.ssrc != new_config.rtp.ssrc) {
     channel_proxy->SetLocalSSRC(new_config.rtp.ssrc);
+    if (stream->suspended_rtp_state_) {
+      stream->rtp_rtcp_module_->SetRtpState(*stream->suspended_rtp_state_);
+    }
   }
   if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
     channel_proxy->SetRTCP_CNAME(new_config.rtp.c_name);
   }
   // TODO(solenberg): Config NACK history window (which is a packet count),
   // using the actual packet size for the configured codec.
   if (first_time || old_config.rtp.nack.rtp_history_ms !=
                         new_config.rtp.nack.rtp_history_ms) {
     channel_proxy->SetNACKStatus(new_config.rtp.nack.rtp_history_ms != 0,
                                  new_config.rtp.nack.rtp_history_ms / 20);
@@ skipping 243 matching lines @@
   return config_;
 }
 
 void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   transport_->send_side_cc()->SetTransportOverhead(
       transport_overhead_per_packet);
   channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
 }
 
+RtpState AudioSendStream::GetRtpState() const {
+  return rtp_rtcp_module_->GetRtpState();
+}
+
 VoiceEngine* AudioSendStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 
 // Apply current codec settings to a single voe::Channel used for sending.
 bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
@@ skipping 193 matching lines @@
   rtc::Event thread_sync_event(false /* manual_reset */, false);
   worker_queue_->PostTask([this, &thread_sync_event] {
     bitrate_allocator_->RemoveObserver(this);
     thread_sync_event.Set();
   });
   thread_sync_event.Wait(rtc::Event::kForever);
 }
 
 void AudioSendStream::RegisterCngPayloadType(int payload_type,
                                              int clockrate_hz) {
-  RtpRtcp* rtpRtcpModule = nullptr;
-  RtpReceiver* rtpReceiver = nullptr;  // Unused, but required for call.
-  channel_proxy_->GetRtpRtcp(&rtpRtcpModule, &rtpReceiver);
   const CodecInst codec = {payload_type, "CN", clockrate_hz, 0, 1, 0};
-  if (rtpRtcpModule->RegisterSendPayload(codec) != 0) {
-    rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
-    if (rtpRtcpModule->RegisterSendPayload(codec) != 0) {
+  if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
+    rtp_rtcp_module_->DeRegisterSendPayload(codec.pltype);
+    if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
       LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
                        "RTP/RTCP module";
     }
   }
 }
 
 
 }  // namespace internal
 }  // namespace webrtc