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 136 matching lines @@
 }  // namespace
 
 namespace internal {
 
 VideoReceiveStream::VideoReceiveStream(
     int num_cpu_cores,
     bool protected_by_flexfec,
     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),
       protected_by_flexfec_(protected_by_flexfec),
       process_thread_(process_thread),
       clock_(Clock::GetRealTimeClock()),
       decode_thread_(DecodeThreadFunction, this, "DecodingThread"),
       congestion_controller_(congestion_controller),
       call_stats_(call_stats),
       timing_(new VCMTiming(clock_)),
       video_receiver_(clock_, nullptr, this, timing_.get(), this, this),
       stats_proxy_(&config_, clock_),
       rtp_stream_receiver_(&video_receiver_,
                            congestion_controller_->GetRemoteBitrateEstimator(
                                UseSendSideBwe(config_)),
                            &transport_adapter_,
                            call_stats_->rtcp_rtt_stats(),
                            packet_router,
                            remb,
                            &config_,
                            &stats_proxy_,
                            process_thread_,
                            this,  // NackSender
                            this,  // KeyFrameRequestSender
                            this,  // OnCompleteFrameCallback
                            timing_.get()),
-      rtp_stream_sync_(&video_receiver_, &rtp_stream_receiver_),
+      rtp_stream_sync_(this),
       jitter_buffer_experiment_(
           field_trial::FindFullName("WebRTC-NewVideoJitterBuffer") ==
           "Enabled") {
   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
   RTC_DCHECK(process_thread_);
   RTC_DCHECK(congestion_controller_);
   RTC_DCHECK(call_stats_);
 
+  module_process_thread_checker_.DetachFromThread();
+
   RTC_DCHECK(!config_.decoders.empty());
   std::set<int> decoder_payload_types;
   for (const Decoder& decoder : config_.decoders) {
     RTC_CHECK(decoder.decoder);
     RTC_CHECK(decoder_payload_types.find(decoder.payload_type) ==
               decoder_payload_types.end())
         << "Duplicate payload type (" << decoder.payload_type
         << ") for different decoders.";
     decoder_payload_types.insert(decoder.payload_type);
   }
 
   video_receiver_.SetRenderDelay(config.render_delay_ms);
 
   if (jitter_buffer_experiment_) {
     jitter_estimator_.reset(new VCMJitterEstimator(clock_));
     frame_buffer_.reset(new video_coding::FrameBuffer(
         clock_, jitter_estimator_.get(), timing_.get()));
   }
 
   process_thread_->RegisterModule(&video_receiver_);
   process_thread_->RegisterModule(&rtp_stream_sync_);
 }
 
 VideoReceiveStream::~VideoReceiveStream() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
   Stop();
 
   process_thread_->DeRegisterModule(&rtp_stream_sync_);
   process_thread_->DeRegisterModule(&video_receiver_);
 
   congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config_))
       ->RemoveStream(rtp_stream_receiver_.GetRemoteSsrc());
 }
 
 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   rtp_stream_receiver_.SignalNetworkState(state);
 }
 
 
 bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return rtp_stream_receiver_.DeliverRtcp(packet, length);
 }
 
 bool VideoReceiveStream::DeliverRtp(const uint8_t* packet,
                                     size_t length,
                                     const PacketTime& packet_time) {
   return rtp_stream_receiver_.DeliverRtp(packet, length, packet_time);
 }
 
 bool VideoReceiveStream::OnRecoveredPacket(const uint8_t* packet,
                                            size_t length) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   return rtp_stream_receiver_.OnRecoveredPacket(packet, length);
 }
 
-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) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  rtp_stream_sync_.ConfigureSync(audio_syncable);
 }
 
 void VideoReceiveStream::Start() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   if (decode_thread_.IsRunning())
     return;
 
   bool protected_by_fec =
       protected_by_flexfec_ || rtp_stream_receiver_.IsUlpfecEnabled();
 
   if (jitter_buffer_experiment_) {
     frame_buffer_->Start();
     call_stats_->RegisterStatsObserver(&rtp_stream_receiver_);
 
@@ skipping 30 matching lines @@
       &stats_proxy_, renderer, config_.pre_render_callback));
   // Register the channel to receive stats updates.
   call_stats_->RegisterStatsObserver(video_stream_decoder_.get());
   // Start the decode thread
   decode_thread_.Start();
   decode_thread_.SetPriority(rtc::kHighestPriority);
   rtp_stream_receiver_.StartReceive();
 }
 
 void VideoReceiveStream::Stop() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   rtp_stream_receiver_.StopReceive();
   // TriggerDecoderShutdown will release any waiting decoder thread and make it
   // stop immediately, instead of waiting for a timeout. Needs to be called
   // before joining the decoder thread thread.
   video_receiver_.TriggerDecoderShutdown();
 
   if (jitter_buffer_experiment_) {
     frame_buffer_->Stop();
     call_stats_->DeregisterStatsObserver(&rtp_stream_receiver_);
   }
@@ skipping 41 matching lines @@
   // TODO(tommi): OnDecodedFrame grabs a lock, incidentally the same lock
   // that OnSyncOffsetUpdated() and OnRenderedFrame() below grab.
   stats_proxy_.OnDecodedFrame();
 
   int64_t sync_offset_ms;
   double estimated_freq_khz;
   // TODO(tommi): GetStreamSyncOffsetInMs grabs three locks.  One inside the
   // function itself, another in GetChannel() and a third in
   // GetPlayoutTimestamp.  Seems excessive.  Anyhow, I'm assuming the function
   // succeeds most of the time, which leads to grabbing a fourth lock.
-  if (rtp_stream_sync_.GetStreamSyncOffsetInMs(video_frame, &sync_offset_ms,
+  if (rtp_stream_sync_.GetStreamSyncOffsetInMs(video_frame.timestamp(),
+                                               video_frame.render_time_ms(),
+                                               &sync_offset_ms,
                                                &estimated_freq_khz)) {
     // TODO(tommi): OnSyncOffsetUpdated grabs a lock.
     stats_proxy_.OnSyncOffsetUpdated(sync_offset_ms, estimated_freq_khz);
   }
 
   // config_.renderer must never be null if we're getting this callback.
   config_.renderer->OnFrame(video_frame);
 
   // TODO(tommi): OnRenderFrame grabs a lock too.
   stats_proxy_.OnRenderedFrame(video_frame);
@@ skipping 33 matching lines @@
   rtp_stream_receiver_.RequestKeyFrame();
 }
 
 void VideoReceiveStream::OnCompleteFrame(
     std::unique_ptr<video_coding::FrameObject> frame) {
   int last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame));
   if (last_continuous_pid != -1)
     rtp_stream_receiver_.FrameContinuous(last_continuous_pid);
 }
 
+int VideoReceiveStream::id() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return config_.rtp.remote_ssrc;
+}
+
+rtc::Optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
+  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+  Syncable::Info info;
+
+  RtpReceiver* rtp_receiver = rtp_stream_receiver_.GetRtpReceiver();
+  RTC_DCHECK(rtp_receiver);
+  if (!rtp_receiver->Timestamp(&info.latest_received_capture_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.capture_time_ntp_secs,
+                          &info.capture_time_ntp_frac,
+                          nullptr,
+                          nullptr,
+                          &info.capture_time_source_clock) != 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();
+  return 0;
+}
+
+void VideoReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
+  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
+  video_receiver_.SetMinimumPlayoutDelay(delay_ms);
+}
+
 bool VideoReceiveStream::DecodeThreadFunction(void* ptr) {
   static_cast<VideoReceiveStream*>(ptr)->Decode();
   return true;
 }
 
 void VideoReceiveStream::Decode() {
   static const int kMaxDecodeWaitTimeMs = 50;
   if (jitter_buffer_experiment_) {
     static const int kMaxWaitForFrameMs = 3000;
     std::unique_ptr<video_coding::FrameObject> frame;
     video_coding::FrameBuffer::ReturnReason res =
         frame_buffer_->NextFrame(kMaxWaitForFrameMs, &frame);
 
     if (res == video_coding::FrameBuffer::ReturnReason::kStopped)
       return;
 
     if (frame) {
       if (video_receiver_.Decode(frame.get()) == VCM_OK)
         rtp_stream_receiver_.FrameDecoded(frame->picture_id);
     } else {
       LOG(LS_WARNING) << "No decodable frame in " << kMaxWaitForFrameMs
                       << " ms, requesting keyframe.";
       RequestKeyFrame();
     }
   } else {
     video_receiver_.Decode(kMaxDecodeWaitTimeMs);
   }
 }
-
 }  // namespace internal
 }  // namespace webrtc