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1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h" | 5 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h" |
6 | 6 |
7 #include "base/logging.h" | 7 #include "base/logging.h" |
8 #include "content/renderer/media/media_stream_audio_source.h" | |
9 #include "content/renderer/media/media_stream_audio_track.h" | 8 #include "content/renderer/media/media_stream_audio_track.h" |
10 #include "content/renderer/media/media_stream_track.h" | 9 #include "content/renderer/media/media_stream_track.h" |
11 #include "content/renderer/media/webrtc/media_stream_video_webrtc_sink.h" | 10 #include "content/renderer/media/webrtc/media_stream_video_webrtc_sink.h" |
12 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" | 11 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h" |
| 12 #include "content/renderer/media/webrtc/peer_connection_remote_audio_source.h" |
| 13 #include "content/renderer/media/webrtc/processed_local_audio_track.h" |
| 14 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
13 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" | 15 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" |
14 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | 16 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
15 #include "third_party/WebKit/public/platform/WebString.h" | 17 #include "third_party/WebKit/public/platform/WebString.h" |
16 | 18 |
17 namespace content { | 19 namespace content { |
18 | 20 |
19 WebRtcMediaStreamAdapter::WebRtcMediaStreamAdapter( | 21 WebRtcMediaStreamAdapter::WebRtcMediaStreamAdapter( |
20 const blink::WebMediaStream& web_stream, | 22 const blink::WebMediaStream& web_stream, |
21 PeerConnectionDependencyFactory* factory) | 23 PeerConnectionDependencyFactory* factory) |
22 : web_stream_(web_stream), | 24 : web_stream_(web_stream), |
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77 const blink::WebMediaStreamTrack& track) { | 79 const blink::WebMediaStreamTrack& track) { |
78 DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeAudio); | 80 DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeAudio); |
79 // A media stream is connected to a peer connection, enable the | 81 // A media stream is connected to a peer connection, enable the |
80 // peer connection mode for the sources. | 82 // peer connection mode for the sources. |
81 MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::From(track); | 83 MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::From(track); |
82 if (!native_track) { | 84 if (!native_track) { |
83 DLOG(ERROR) << "No native track for blink audio track."; | 85 DLOG(ERROR) << "No native track for blink audio track."; |
84 return; | 86 return; |
85 } | 87 } |
86 | 88 |
87 webrtc::AudioTrackInterface* audio_track = native_track->GetAudioAdapter(); | 89 // If we have an instance of ProcessedLocalAudioTrack or |
88 if (!audio_track) { | 90 // PeerConnectionRemoteAudioTrack, use its webrtc::AudioTrackInterface |
89 DLOG(ERROR) << "Audio track doesn't support webrtc."; | 91 // implementation. Otherwise, create a place-holder instance for tracks |
90 return; | 92 // providing audio data from other sources. |
| 93 scoped_refptr<webrtc::AudioTrackInterface> track_interface; |
| 94 if (ProcessedLocalAudioTrack* local_rtc_track = |
| 95 ProcessedLocalAudioTrack::From(native_track)) { |
| 96 track_interface = local_rtc_track->adapter(); |
| 97 } else if (PeerConnectionRemoteAudioTrack* remote_pc_track = |
| 98 PeerConnectionRemoteAudioTrack::From(native_track)) { |
| 99 track_interface = remote_pc_track->track_interface(); |
| 100 } else { |
| 101 const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter = |
| 102 WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(), nullptr); |
| 103 adapter->SetMediaStreamAudioTrack(native_track->GetWeakPtr()); |
| 104 track_interface = adapter; |
91 } | 105 } |
92 | 106 |
93 if (native_track->is_local_track()) { | 107 webrtc_media_stream_->AddTrack(track_interface.get()); |
94 const blink::WebMediaStreamSource& source = track.source(); | |
95 MediaStreamAudioSource* audio_source = MediaStreamAudioSource::From(source); | |
96 if (audio_source && audio_source->audio_capturer()) | |
97 audio_source->audio_capturer()->EnablePeerConnectionMode(); | |
98 } | |
99 | |
100 webrtc_media_stream_->AddTrack(audio_track); | |
101 } | 108 } |
102 | 109 |
103 void WebRtcMediaStreamAdapter::CreateVideoTrack( | 110 void WebRtcMediaStreamAdapter::CreateVideoTrack( |
104 const blink::WebMediaStreamTrack& track) { | 111 const blink::WebMediaStreamTrack& track) { |
105 DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeVideo); | 112 DCHECK_EQ(track.source().getType(), blink::WebMediaStreamSource::TypeVideo); |
106 MediaStreamVideoWebRtcSink* adapter = | 113 MediaStreamVideoWebRtcSink* adapter = |
107 new MediaStreamVideoWebRtcSink(track, factory_); | 114 new MediaStreamVideoWebRtcSink(track, factory_); |
108 video_adapters_.push_back(adapter); | 115 video_adapters_.push_back(adapter); |
109 webrtc_media_stream_->AddTrack(adapter->webrtc_video_track()); | 116 webrtc_media_stream_->AddTrack(adapter->webrtc_video_track()); |
110 } | 117 } |
111 | 118 |
112 } // namespace content | 119 } // namespace content |
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