Interface RTCRtpReceiver and RTCPeerConnection.getReceivers() added.

content/renderer/media/rtc_peer_connection_handler.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/rtc_peer_connection_handler.h"
 
 #include <string.h>
 
 #include <string>
 #include <utility>
@@ skipping 11 matching lines @@
 #include "content/public/common/content_features.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_constraints_util.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #include "content/renderer/media/remote_media_stream_impl.h"
 #include "content/renderer/media/rtc_certificate.h"
 #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"
 #include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/base/media_switches.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebRTCAnswerOptions.h"
 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
@@ skipping 1585 matching lines @@
 
 void RTCPeerConnectionHandler::getStats(
     std::unique_ptr<blink::WebRTCStatsReportCallback> callback) {
   DCHECK(thread_checker_.CalledOnValidThread());
   signaling_thread()->PostTask(FROM_HERE,
       base::Bind(&GetRTCStatsOnSignalingThread,
           base::ThreadTaskRunnerHandle::Get(), native_peer_connection_,
           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_)
     client_->closePeerConnection();
 }
 
 void RTCPeerConnectionHandler::StartEventLog(IPC::PlatformFileForTransit file,
                                              int64_t max_file_size_bytes) {
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(file != IPC::InvalidPlatformFileForTransit());
@@ skipping 340 matching lines @@
     base::WaitableEvent event(base::WaitableEvent::ResetPolicy::AUTOMATIC,
                               base::WaitableEvent::InitialState::NOT_SIGNALED);
     thread->PostTask(FROM_HERE,
         base::Bind(&RunSynchronousClosure, closure,
                    base::Unretained(trace_event_name),
                    base::Unretained(&event)));
     event.Wait();
   }
 }
 
+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());
   if (ice_state_seen_[new_state])
     return;
   ice_state_seen_[new_state] = true;
   UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.ConnectionState", new_state,
                             webrtc::PeerConnectionInterface::kIceConnectionMax);
 }
 
 void RTCPeerConnectionHandler::ResetUMAStats() {
   DCHECK(thread_checker_.CalledOnValidThread());
   num_local_candidates_ipv6_ = 0;
   num_local_candidates_ipv4_ = 0;
   ice_connection_checking_start_ = base::TimeTicks();
   memset(ice_state_seen_, 0, sizeof(ice_state_seen_));
 }
 }  // namespace content