Forward the SignalFirstPacketReceived to RtpReceiver.

webrtc/api/rtpreceiver.cc

Index: webrtc/api/rtpreceiver.cc
diff --git a/webrtc/api/rtpreceiver.cc b/webrtc/api/rtpreceiver.cc
index 1b52ce27def13f255c245d353aa239f4d64d1078..923350c06ec50bcd430f960aae21c63f52c326ad 100644
--- a/webrtc/api/rtpreceiver.cc
+++ b/webrtc/api/rtpreceiver.cc
@@ -29,12 +29,15 @@ AudioRtpReceiver::AudioRtpReceiver(MediaStreamInterface* stream,
           rtc::Thread::Current(),
           AudioTrack::Create(track_id,
                              RemoteAudioSource::Create(ssrc, provider)))),
-      cached_track_enabled_(track_->enabled()) {
+      cached_track_enabled_(track_->enabled()),
+      observer_(nullptr) {

[pthatcher1, 2016/06/08 17:35:55]

You don't need this if you use "= nullptr" in the other spot.

   RTC_DCHECK(track_->GetSource()->remote());
   track_->RegisterObserver(this);
   track_->GetSource()->RegisterAudioObserver(this);
   Reconfigure();
   stream->AddTrack(track_);
+  provider_->SignalFirstAudioPacketReceived.connect(
+      this, &AudioRtpReceiver::OnFirstAudioPacketReceived);
 }
 
 AudioRtpReceiver::~AudioRtpReceiver() {
@@ -83,6 +86,22 @@ void AudioRtpReceiver::Reconfigure() {
   provider_->SetAudioPlayout(ssrc_, track_->enabled());
 }
 
+void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
+  observer_ = observer;
+  // If received the first packet before setting the observer, call the
+  // observer.
+  if (received_first_packet_) {
+    observer_->OnFirstPacketReceived(cricket::MEDIA_TYPE_AUDIO);
+  }
+}
+
+void AudioRtpReceiver::OnFirstAudioPacketReceived() {
+  if (observer_) {
+    observer_->OnFirstPacketReceived(cricket::MEDIA_TYPE_AUDIO);
+  }
+  received_first_packet_ = true;

[pthatcher1, 2016/06/08 17:35:55]

Do you have a unit test that covers setting the observer after the signal has
fired?  You could do so by simply setting a new observer after it's fired and
then seeing if the second observer fired.  

+}
+
 VideoRtpReceiver::VideoRtpReceiver(MediaStreamInterface* stream,
                                    const std::string& track_id,
                                    rtc::Thread* worker_thread,
@@ -100,10 +119,13 @@ VideoRtpReceiver::VideoRtpReceiver(MediaStreamInterface* stream,
               track_id,
               VideoTrackSourceProxy::Create(rtc::Thread::Current(),
                                             worker_thread,
-                                            source_)))) {
+                                            source_)))),
+      observer_(nullptr) {
   source_->SetState(MediaSourceInterface::kLive);
   provider_->SetVideoPlayout(ssrc_, true, &broadcaster_);
   stream->AddTrack(track_);
+  provider_->SignalFirstVideoPacketReceived.connect(
+      this, &VideoRtpReceiver::OnFirstVideoPacketReceived);
 }
 
 VideoRtpReceiver::~VideoRtpReceiver() {
@@ -132,4 +154,20 @@ bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
   return provider_->SetVideoRtpReceiveParameters(ssrc_, parameters);
 }
 
+void VideoRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
+  observer_ = observer;
+  // If received the first packet before setting the observer, call the
+  // observer.
+  if (received_first_packet_) {
+    observer_->OnFirstPacketReceived(cricket::MEDIA_TYPE_VIDEO);
+  }
+}
+
+void VideoRtpReceiver::OnFirstVideoPacketReceived() {
+  if (observer_) {
+    observer_->OnFirstPacketReceived(cricket::MEDIA_TYPE_VIDEO);

[pthatcher1, 2016/06/08 17:35:55]

Might as well add a media_type() getter to the RtpReceiver so that you can just
do media_type() here instead of duplicating MEDIA_TYPE_VIDEO.

Also, this might allow having shared code with a common RtpReceiverBase, which
Taylor is planning on adding.

[Zhi Huang, 2016/06/09 00:37:36]

Done.

+  }
+  received_first_packet_ = true;
+}
+
 }  // namespace webrtc