Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/video/video_receive_stream.cc

Index: webrtc/video/video_receive_stream.cc
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 7835e779b0dca5aa11fe5c79f20a3976bd47b9c5..6450e59d880c563e3116c6a9dd891922d382bfbd 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -22,6 +22,8 @@
 #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"
@@ -33,7 +35,6 @@
 #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 {
 
@@ -191,7 +192,6 @@ VideoReceiveStream::VideoReceiveStream(
     CongestionController* congestion_controller,
     PacketRouter* packet_router,
     VideoReceiveStream::Config config,
-    webrtc::VoiceEngine* voice_engine,
     ProcessThread* process_thread,
     CallStats* call_stats,
     VieRemb* remb)
@@ -223,7 +223,7 @@ VideoReceiveStream::VideoReceiveStream(
           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_);
@@ -357,15 +357,8 @@ void VideoReceiveStream::Stop() {
   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 {
@@ -480,5 +473,42 @@ 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