Chromium Code Reviews| Index: content/renderer/media/webrtc/webrtc_media_stream_adapter.cc |
| diff --git a/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc b/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc |
| index 3c34d794dded57cd712436cbad564dca909e531b..e4692908f24258597e71158ed6858b98e794906d 100644 |
| --- a/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc |
| +++ b/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc |
| @@ -5,11 +5,13 @@ |
| #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h" |
| #include "base/logging.h" |
| -#include "content/renderer/media/media_stream_audio_source.h" |
| #include "content/renderer/media/media_stream_audio_track.h" |
| #include "content/renderer/media/media_stream_track.h" |
| #include "content/renderer/media/webrtc/media_stream_video_webrtc_sink.h" |
| #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" |
| +#include "content/renderer/media/webrtc/peer_connection_remote_audio_source.h" |
| +#include "content/renderer/media/webrtc/processed_local_audio_track.h" |
| +#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
| #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" |
| #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
| #include "third_party/WebKit/public/platform/WebString.h" |
| @@ -84,20 +86,25 @@ void WebRtcMediaStreamAdapter::CreateAudioTrack( |
| return; |
| } |
| - webrtc::AudioTrackInterface* audio_track = native_track->GetAudioAdapter(); |
| - if (!audio_track) { |
| - DLOG(ERROR) << "Audio track doesn't support webrtc."; |
| - return; |
| - } |
| - |
| - if (native_track->is_local_track()) { |
| - const blink::WebMediaStreamSource& source = track.source(); |
| - MediaStreamAudioSource* audio_source = MediaStreamAudioSource::From(source); |
| - if (audio_source && audio_source->audio_capturer()) |
| - audio_source->audio_capturer()->EnablePeerConnectionMode(); |
| + // If we have an instance of ProcessedLocalAudioTrack or |
| + // PeerConnectionRemoteAudioTrack, use its webrtc::AudioTrackInterface |
| + // implementation. Otherwise, create a place-holder instance for tracks |
| + // providing audio data from other sources. |
| + scoped_refptr<webrtc::AudioTrackInterface> track_interface; |
| + if (ProcessedLocalAudioTrack* local_rtc_track = |
| + ProcessedLocalAudioTrack::From(native_track)) { |
| + track_interface = local_rtc_track->adapter(); |
| + } else if (PeerConnectionRemoteAudioTrack* remote_pc_track = |
| + PeerConnectionRemoteAudioTrack::From(native_track)) { |
| + track_interface = remote_pc_track->track_interface(); |
| + } else { |
| + const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter = |
| + WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(), nullptr); |
| + adapter->SetMediaStreamAudioTrack(native_track->GetWeakPtr()); |
| + track_interface = adapter; |
| } |
| - webrtc_media_stream_->AddTrack(audio_track); |
| + webrtc_media_stream_->AddTrack(track_interface.get()); |
|
o1ka
2016/03/31 16:35:35
If track_interface was created at l.101, doesn't i
miu
2016/03/31 22:35:30
webrtc::AudioTrackInterface implements libjingle's
o1ka
2016/04/01 15:11:41
Ok, this is someting new for me, thanks!
But does
miu
2016/04/08 18:36:47
Regarding the code in Patch Set 3: Yes, webrtc_med
|
| } |
| void WebRtcMediaStreamAdapter::CreateVideoTrack( |