Start documenting the threading model

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 be623cef56ade6bd4c3f60357bdc7eb47db151ad..dd2faf3e68f867da03d2bbd834d174ed3f166201 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -20,6 +20,7 @@
 #include "webrtc/base/location.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/common_video/h264/profile_level_id.h"
 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
@@ -236,8 +237,12 @@ VideoReceiveStream::~VideoReceiveStream() {
   LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString();
   Stop();
 
+  LOG(LS_INFO) << "~VideoReceiveStream: DeRegisterModule rtp_stream_sync_.";
   process_thread_->DeRegisterModule(&rtp_stream_sync_);
+  LOG(LS_INFO) << "~VideoReceiveStream: DeRegisterModule video_receiver_.";
   process_thread_->DeRegisterModule(&video_receiver_);
+
+  LOG(LS_INFO) << "~VideoReceiveStream: Modules DeRegistered.";
 }
 
 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
@@ -310,29 +315,26 @@ void VideoReceiveStream::Start() {
   call_stats_->RegisterStatsObserver(video_stream_decoder_.get());
   // Start the decode thread
   decode_thread_.Start();
-#if !defined(WEBRTC_ANDROID)
-  // On android, the decoding happens on a different thread and frames
-  // are delivered on that thread (that in itself needs to be fixed).
-  // In any event, the actual decoding work is higher priority than the
-  // |decode_thread_| on Android, so we only raise the deocode priority on
-  // platforms other than Android.
-  decode_thread_.SetPriority(rtc::kHighestPriority);
-#endif
   rtp_stream_receiver_.StartReceive();
 }
 
 void VideoReceiveStream::Stop() {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: StopReceive";
   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.
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: TriggerDecoderShutdown";
   video_receiver_.TriggerDecoderShutdown();
 
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: frame_buffer_";
   frame_buffer_->Stop();
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: call_stats_";
   call_stats_->DeregisterStatsObserver(&rtp_stream_receiver_);
 
   if (decode_thread_.IsRunning()) {
+    LOG(LS_INFO) << "VideoReceiveStream::Stop: decode_thread_";
     decode_thread_.Stop();
     // Deregister external decoders so they are no longer running during
     // destruction. This effectively stops the VCM since the decoder thread is
@@ -342,10 +344,15 @@ void VideoReceiveStream::Stop() {
       video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type);
   }
 
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: call_stats_";
   call_stats_->DeregisterStatsObserver(video_stream_decoder_.get());
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: video_stream_decoder_";
   video_stream_decoder_.reset();
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: incoming_video_stream_";
   incoming_video_stream_.reset();
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: transport_adapter_";
   transport_adapter_.Disable();
+  LOG(LS_INFO) << "VideoReceiveStream::Stop: done";
 }
 
 VideoReceiveStream::Stats VideoReceiveStream::GetStats() const {
@@ -477,28 +484,48 @@ void VideoReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
   video_receiver_.SetMinimumPlayoutDelay(delay_ms);
 }
 
-bool VideoReceiveStream::DecodeThreadFunction(void* ptr) {
-  return static_cast<VideoReceiveStream*>(ptr)->Decode();
+void VideoReceiveStream::DecodeThreadFunction(void* ptr) {
+  while (static_cast<VideoReceiveStream*>(ptr)->Decode()) {
+  }
 }
 
 bool VideoReceiveStream::Decode() {
   TRACE_EVENT0("webrtc", "VideoReceiveStream::Decode");
   static const int kMaxWaitForFrameMs = 3000;
   std::unique_ptr<video_coding::FrameObject> frame;
-  video_coding::FrameBuffer::ReturnReason res =
-      frame_buffer_->NextFrame(kMaxWaitForFrameMs, &frame);
+
+  video_coding::FrameBuffer::ReturnReason res;
+#if defined(WEBRTC_ANDROID)
+  static const int kPollIntervalMs = 10;
+  int time_remaining = kMaxWaitForFrameMs;
+  do {
+    res = frame_buffer_->NextFrame(kPollIntervalMs, &frame);
+    if (res != video_coding::FrameBuffer::ReturnReason::kTimeout)
+      break;
+    time_remaining -= kPollIntervalMs;
+    video_receiver_.PollDecodedFrames();
+  } while (time_remaining > 0);
+#else
+  res = frame_buffer_->NextFrame(kMaxWaitForFrameMs, &frame);
+#endif
 
   if (res == video_coding::FrameBuffer::ReturnReason::kStopped)
     return false;
 
   if (frame) {
-    if (video_receiver_.Decode(frame.get()) == VCM_OK)
+    auto ret = video_receiver_.Decode(frame.get());
+    if (ret == VCM_OK) {
       rtp_stream_receiver_.FrameDecoded(frame->picture_id);
+    } else {
+      printf("ret = %i\n", ret);
+    }
   } else {
+    RTC_DCHECK_EQ(res, video_coding::FrameBuffer::ReturnReason::kTimeout);
     LOG(LS_WARNING) << "No decodable frame in " << kMaxWaitForFrameMs
                     << " ms, requesting keyframe.";
     RequestKeyFrame();
   }
+
   return true;
 }
 }  // namespace internal