Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/audio/audio_receive_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.
  */
 
 #include "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 #include <utility>
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.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/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_neteq_stats.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
@@ skipping 184 matching lines @@
 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   channel_proxy_->SetSink(std::move(sink));
 }
 
 void AudioReceiveStream::SetGain(float gain) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   channel_proxy_->SetChannelOutputVolumeScaling(gain);
 }
 
-const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
+    int sample_rate_hz,
+    AudioFrame* audio_frame) {
+  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+}
+
+int AudioReceiveStream::Ssrc() const {
+  return config_.rtp.remote_ssrc;
+}
+
+int AudioReceiveStream::PreferredSampleRate() const {
+  return channel_proxy_->NeededFrequency();
+}
+
+int AudioReceiveStream::id() const {
   RTC_DCHECK_RUN_ON(&thread_checker_);
-  return config_;
+  return config_.rtp.remote_ssrc;
+}
+
+rtc::Optional<Syncable::Info> AudioReceiveStream::GetInfo() const {

[stefan-webrtc, 2017/01/19 11:59:59]

Should you make sure this runs on the right thread too? Seems like the same
requirement applies as for id()?

[the sun, 2017/01/19 13:15:36]

id() is called when sync is set up, part of stream create/destroy calls, on
worker thread.

GetInfo() is called on Call's processing thread. Nevertheless, I'll add a TC.

+  // Called on Call's module_process_thread_.
+  Syncable::Info info;
+
+  RtpRtcp* rtp_rtcp = nullptr;
+  RtpReceiver* rtp_receiver = nullptr;
+  channel_proxy_->GetRtpRtcp(&rtp_rtcp, &rtp_receiver);
+  RTC_DCHECK(rtp_rtcp);
+  RTC_DCHECK(rtp_receiver);
+
+  if (!rtp_receiver->Timestamp(&info.latest_timestamp))
+    return rtc::Optional<Syncable::Info>();
+  if (!rtp_receiver->LastReceivedTimeMs(&info.latest_receive_time_ms))
+    return rtc::Optional<Syncable::Info>();
+  if (rtp_rtcp->RemoteNTP(&info.ntp_secs, &info.ntp_frac, nullptr, nullptr,
+                          &info.rtp_timestamp) != 0) {
+    return rtc::Optional<Syncable::Info>();
+  }
+
+  int jitter_buffer_delay_ms = 0;
+  int playout_buffer_delay_ms = 0;
+  channel_proxy_->GetDelayEstimate(&jitter_buffer_delay_ms,
+                                   &playout_buffer_delay_ms);
+  info.current_delay_ms = jitter_buffer_delay_ms + playout_buffer_delay_ms;
+  return rtc::Optional<Syncable::Info>(info);
+}
+
+uint32_t AudioReceiveStream::GetPlayoutTimestamp() const {
+  // Called on video capture thread.
+  return channel_proxy_->GetPlayoutTimestamp();
+}
+
+void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
+  // Called on Call's module_process_thread_.
+  return channel_proxy_->SetMinimumPlayoutDelay(delay_ms);
 }
 
 void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   if (send_stream) {
     VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
     std::unique_ptr<voe::ChannelProxy> send_channel_proxy =
         voe_impl->GetChannelProxy(send_stream->config().voe_channel_id);
     channel_proxy_->AssociateSendChannel(*send_channel_proxy.get());
   } else {
     channel_proxy_->DisassociateSendChannel();
   }
 }
 
@@ skipping 30 matching lines @@
     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header);
   }
 
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
-AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
-    int sample_rate_hz,
-    AudioFrame* audio_frame) {
-  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return config_;
 }
 
-int AudioReceiveStream::PreferredSampleRate() const {
-  return channel_proxy_->NeededFrequency();
-}
-
-int AudioReceiveStream::Ssrc() const {
-  return config_.rtp.remote_ssrc;
+VoiceEngine* AudioReceiveStream::voice_engine() const {
+  auto* voice_engine = audio_state()->voice_engine();
+  RTC_DCHECK(voice_engine);
+  return voice_engine;
 }
 
 internal::AudioState* AudioReceiveStream::audio_state() const {
   auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
   RTC_DCHECK(audio_state);
   return audio_state;
 }
 
-VoiceEngine* AudioReceiveStream::voice_engine() const {
-  auto* voice_engine = audio_state()->voice_engine();
-  RTC_DCHECK(voice_engine);
-  return voice_engine;
-}
-
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
-
 }  // namespace internal
 }  // namespace webrtc