Interface RTCRtpReceiver and RTCPeerConnection.getReceivers() added.

content/renderer/media/rtc_peer_connection_handler.cc

Index: content/renderer/media/rtc_peer_connection_handler.cc
diff --git a/content/renderer/media/rtc_peer_connection_handler.cc b/content/renderer/media/rtc_peer_connection_handler.cc
index 3240a7a2fbe2e9e97bb3f0bade3ce22d159fb618..554a3267124ec110e76b6b489b4459821efaa132 100644
--- a/content/renderer/media/rtc_peer_connection_handler.cc
+++ b/content/renderer/media/rtc_peer_connection_handler.cc
@@ -29,6 +29,7 @@
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/rtc_rtp_receiver.h"
 #include "content/renderer/media/webrtc/rtc_stats.h"
 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
@@ -1634,6 +1635,30 @@ void RTCPeerConnectionHandler::getStats(
           base::Passed(&callback)));
 }
 
+blink::WebVector<std::unique_ptr<blink::WebRTCRtpReceiver>>
+RTCPeerConnectionHandler::getReceivers() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::getReceivers");
+
+  std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>>
+      webrtc_receivers = native_peer_connection_->GetReceivers();
+  blink::WebVector<std::unique_ptr<blink::WebRTCRtpReceiver>> web_receivers(
+      webrtc_receivers.size());
+  for (size_t i = 0; i < webrtc_receivers.size(); ++i) {
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> webrtc_track =
+        webrtc_receivers[i]->track();
+    DCHECK(webrtc_track);
+    if (webrtc_track->kind() == webrtc::MediaStreamTrackInterface::kAudioKind) {
+      web_receivers[i].reset(new RTCRtpReceiver(

[Taylor_Brandstetter, 2017/03/24 18:13:07]

If I understand correctly: this method will create a new content::RTCRtpReceiver
every time the method is called, and the blink layer ensures that only one
blink::RTCRtpReceiver is created per webrtc::RtpReceiver. And it does this using
getId, which returns the value of the webrtc::RtpReceiver pointer. This is a
little confusing; can you describe it all in a comment somewhere?

[hbos_chromium, 2017/03/27 14:55:19]

You understand correctly.

There are comments in WebRTCRtpReceiver.
Added new comments here, in rtc_rtp_receiver.h, WebRTCPeerConnectionHandler.h
and rtc_peer_connection_handler.cc.

+          webrtc_receivers[i].get(), GetRemoteAudioTrack(webrtc_track->id())));
+    } else {
+      web_receivers[i].reset(new RTCRtpReceiver(
+          webrtc_receivers[i].get(), GetRemoteVideoTrack(webrtc_track->id())));
+    }
+  }
+  return web_receivers;
+}
+
 void RTCPeerConnectionHandler::CloseClientPeerConnection() {
   DCHECK(thread_checker_.CalledOnValidThread());
   if (!is_closed_)
@@ -1994,6 +2019,32 @@ void RTCPeerConnectionHandler::RunSynchronousClosureOnSignalingThread(
   }
 }
 
+blink::WebMediaStreamTrack RTCPeerConnectionHandler::GetRemoteAudioTrack(
+    const std::string& track_id) const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  blink::WebString id = blink::WebString::fromUTF8(track_id);
+  for (const auto& remote_stream_pair : remote_streams_) {

[Taylor_Brandstetter, 2017/03/24 18:13:07]

May be a corner case, but someone could remove the track from the remote stream,
so getting the tracks from the streams isn't 100% reliable.

Could there just be a map somewhere of webrtc:: tracks to blink:: tracks?

[hbos_chromium, 2017/03/27 14:55:19]

The case of adding or removing tracks to streams is already not properly
supported on the blink layer. The blink::RTCPeerConnection only knows of the
didAddRemoteStream and didRemoveRemoteStream cases:
https://cs.chromium.org/chromium/src/third_party/WebKit/Source/modules/peerco...

didAddRemoteStream is indirectly called from
content::RTCPeerConnectionHandler::Observer::OnAddStream which creates the
stream and its tracks:
https://cs.chromium.org/chromium/src/content/renderer/media/rtc_peer_connecti...

The stream/track problem needs to be addressed, but I think listening to events
of tracks being added and removed and decoupling tracks from streams should be
addressed separately as part of the add/removeTrack work.

I have a TODO about this in RTCPeerConnection.h, I added another TODO here. I
also added an if-statement to ignore receivers that don't have a known track
with a TODO to avoid a possible DCHECK crash.

[Taylor_Brandstetter, 2017/03/27 21:39:01]

Sounds like a reasonable plan. I'm happy as long as we document the limitations
and can't hit DCHECKs.

+    blink::WebMediaStreamTrack web_track =
+        remote_stream_pair.second->webkit_stream().getAudioTrack(id);
+    if (!web_track.isNull())
+      return web_track;
+  }
+  return blink::WebMediaStreamTrack();
+}
+
+blink::WebMediaStreamTrack RTCPeerConnectionHandler::GetRemoteVideoTrack(
+    const std::string& track_id) const {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  blink::WebString id = blink::WebString::fromUTF8(track_id);
+  for (const auto& remote_stream_pair : remote_streams_) {
+    blink::WebMediaStreamTrack web_track =
+        remote_stream_pair.second->webkit_stream().getVideoTrack(id);
+    if (!web_track.isNull())
+      return web_track;
+  }
+  return blink::WebMediaStreamTrack();
+}
+
 void RTCPeerConnectionHandler::ReportICEState(
     webrtc::PeerConnectionInterface::IceConnectionState new_state) {
   DCHECK(thread_checker_.CalledOnValidThread());