Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/video/video_receive_stream.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.
  */
 
 #include "webrtc/video/video_receive_stream.h"
 
 #include <stdlib.h>
 
 #include <set>
 #include <string>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/common_video/h264/profile_level_id.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/modules/video_coding/frame_object.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
 #include "webrtc/modules/video_coding/jitter_estimator.h"
 #include "webrtc/modules/video_coding/timing.h"
 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/receive_statistics_proxy.h"
 #include "webrtc/video_receive_stream.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
 
 static bool UseSendSideBwe(const VideoReceiveStream::Config& config) {
   if (!config.rtp.transport_cc)
     return false;
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kTransportSequenceNumberUri)
       return true;
   }
@@ skipping 137 matching lines @@
 }
 }  // namespace
 
 namespace internal {
 
 VideoReceiveStream::VideoReceiveStream(
     int num_cpu_cores,
     CongestionController* congestion_controller,
     PacketRouter* packet_router,
     VideoReceiveStream::Config config,
-    webrtc::VoiceEngine* voice_engine,
     ProcessThread* process_thread,
     CallStats* call_stats,
     VieRemb* remb)
     : transport_adapter_(config.rtcp_send_transport),
       config_(std::move(config)),
       num_cpu_cores_(num_cpu_cores),
       process_thread_(process_thread),
       clock_(Clock::GetRealTimeClock()),
       decode_thread_(DecodeThreadFunction, this, "DecodingThread"),
       congestion_controller_(congestion_controller),
@@ skipping 11 matching lines @@
           packet_router,
           remb,
           &config_,
           &stats_proxy_,
           process_thread_,
           congestion_controller_->GetRetransmissionRateLimiter(),
           this,  // NackSender
           this,  // KeyFrameRequestSender
           this,  // OnCompleteFrameCallback
           timing_.get()),
-      rtp_stream_sync_(&video_receiver_, &rtp_stream_receiver_) {
+      rtp_stream_sync_(this) {
   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
   RTC_DCHECK(process_thread_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(call_stats_);
 
   RTC_DCHECK(!config_.decoders.empty());
   std::set<int> decoder_payload_types;
   for (const Decoder& decoder : config_.decoders) {
     RTC_CHECK(decoder.decoder);
@@ skipping 113 matching lines @@
     for (const Decoder& decoder : config_.decoders)
       video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type);
   }
 
   call_stats_->DeregisterStatsObserver(video_stream_decoder_.get());
   video_stream_decoder_.reset();
   incoming_video_stream_.reset();
   transport_adapter_.Disable();
 }
 
-void VideoReceiveStream::SetSyncChannel(VoiceEngine* voice_engine,
-                                        int audio_channel_id) {
-  if (voice_engine && audio_channel_id != -1) {
-    VoEVideoSync* voe_sync_interface = VoEVideoSync::GetInterface(voice_engine);
-    rtp_stream_sync_.ConfigureSync(audio_channel_id, voe_sync_interface);
-    voe_sync_interface->Release();
-  } else {
-    rtp_stream_sync_.ConfigureSync(-1, nullptr);
-  }
+void VideoReceiveStream::SetSync(Syncable* audio_syncable) {
+  rtp_stream_sync_.ConfigureSync(audio_syncable);
 }
 
 VideoReceiveStream::Stats VideoReceiveStream::GetStats() const {
   return stats_proxy_.GetStats();
 }
 
 // TODO(tommi): This method grabs a lock 6 times.
 void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) {
   // TODO(tommi): OnDecodedFrame grabs a lock, incidentally the same lock
   // that OnSyncOffsetUpdated() and OnRenderedFrame() below grab.
@@ skipping 94 matching lines @@
     // Make a keyframe appear as early as possible in the logs, to give actually
     // decodable output.
     RequestKeyFrame();
   }
 }
 
 void VideoReceiveStream::RequestKeyFrame() {
   rtp_stream_receiver_.RequestKeyFrame();
 }
 
+int VideoReceiveStream::id() const {
+  // TODO(solenberg): This appears to be what the current code does, but I
+  // believe we should be using remote_ssrc instead?
+  return config_.rtp.local_ssrc;
+}
+
+rtc::Optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
+  // Called on Call's module_process_thread_.

[stefan-webrtc, 2017/01/26 08:40:50]

I wouldn't mind if thread checks were added here too, as for audio, please? :)

[the sun, 2017/01/30 15:43:12]

Happy to oblige.

+  Syncable::Info info;
+
+  RtpReceiver* rtp_receiver = rtp_stream_receiver_.GetRtpReceiver();
+  RTC_DCHECK(rtp_receiver);
+  if (!rtp_receiver->Timestamp(&info.latest_receive_timestamp))
+    return rtc::Optional<Syncable::Info>();
+  if (!rtp_receiver->LastReceivedTimeMs(&info.latest_receive_time_ms))
+    return rtc::Optional<Syncable::Info>();
+
+  RtpRtcp* rtp_rtcp = rtp_stream_receiver_.rtp_rtcp();
+  RTC_DCHECK(rtp_rtcp);
+  if (rtp_rtcp->RemoteNTP(&info.ntp_secs, &info.ntp_frac, nullptr, nullptr,
+                          &info.rtp_timestamp) != 0) {
+    return rtc::Optional<Syncable::Info>();
+  }
+
+  info.current_delay_ms = video_receiver_.Delay();
+  return rtc::Optional<Syncable::Info>(info);
+}
+
+uint32_t VideoReceiveStream::GetPlayoutTimestamp() const {
+  RTC_NOTREACHED();
+}
+
+void VideoReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
+  // Called on Call's module_process_thread_.
+  video_receiver_.SetMinimumPlayoutDelay(delay_ms);
+}
+
 }  // namespace internal
 }  // namespace webrtc