RTCRtpSender with track behind RuntimeEnabled flag

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 12 matching lines @@
 #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_rtp_sender.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"
 #include "third_party/WebKit/public/platform/WebRTCError.h"
 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
 #include "third_party/WebKit/public/platform/WebRTCLegacyStats.h"
 #include "third_party/WebKit/public/platform/WebRTCOfferOptions.h"
+#include "third_party/WebKit/public/platform/WebRTCRtpSender.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
 #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
 #include "third_party/WebKit/public/platform/WebURL.h"
 #include "third_party/webrtc/pc/mediasession.h"
 
 using webrtc::DataChannelInterface;
 using webrtc::IceCandidateInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
@@ skipping 1587 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::WebRTCRtpSender>>
+RTCPeerConnectionHandler::GetSenders() {
+  DCHECK(thread_checker_.CalledOnValidThread());
+  TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::getSenders");
+  std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> webrtc_senders =
+      native_peer_connection_->GetSenders();
+  blink::WebVector<std::unique_ptr<blink::WebRTCRtpSender>> web_senders(
+      webrtc_senders.size());
+  for (size_t i = 0; i < web_senders.size(); ++i) {
+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> webrtc_track =
+        webrtc_senders[i]->track();
+    std::unique_ptr<blink::WebMediaStreamTrack> web_track;
+
+    if (webrtc_track) {
+      std::string track_id = webrtc_track->id();
+      bool is_audio_track = (webrtc_track->kind() ==
+                             webrtc::MediaStreamTrackInterface::kAudioKind);
+      for (const auto& stream_adapter : local_streams_) {
+        blink::WebVector<blink::WebMediaStreamTrack> tracks;
+        if (is_audio_track)
+          stream_adapter->web_stream().AudioTracks(tracks);
+        else
+          stream_adapter->web_stream().VideoTracks(tracks);
+        for (const blink::WebMediaStreamTrack& track : tracks) {
+          if (track.Id() == track_id.c_str()) {
+            web_track.reset(new blink::WebMediaStreamTrack(track));
+            break;
+          }
+        }
+        if (web_track)
+          break;
+      }
+      DCHECK(web_track);
+    }
+
+    web_senders[i] = base::MakeUnique<RTCRtpSender>(webrtc_senders[i].get(),
+                                                    std::move(web_track));
+  }
+  return web_senders;
+}
+
 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();
   std::vector<std::unique_ptr<blink::WebRTCRtpReceiver>> web_receivers;
   for (size_t i = 0; i < webrtc_receivers.size(); ++i) {
     rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> webrtc_track =
@@ skipping 413 matching lines @@
 }
 
 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