| /external/chromium_org/content/renderer/media/webrtc/ |
| D | webrtc_media_stream_adapter_unittest.cc | 35 blink::WebMediaStreamSource audio_source; in CreateBlinkMediaStream() local
36 audio_source.initialize("audio", in CreateBlinkMediaStream()
39 audio_source.setExtraData(new MediaStreamAudioSource()); in CreateBlinkMediaStream()
41 audio_track_vector[0].initialize(audio_source); in CreateBlinkMediaStream()
112 blink::WebMediaStreamSource audio_source; in TEST_F() local
113 audio_source.initialize("audio source", in TEST_F()
119 audio_tracks[0].initialize(audio_source.id(), audio_source); in TEST_F()
|
| D | webrtc_media_stream_adapter.cc | 94 MediaStreamAudioSource* audio_source = in CreateAudioTrack() local
96 if (audio_source && audio_source->GetAudioCapturer()) in CreateAudioTrack()
97 audio_source->GetAudioCapturer()->EnablePeerConnectionMode(); in CreateAudioTrack()
|
| D | mock_peer_connection_dependency_factory.cc | 526 MediaStreamAudioSource* audio_source) { in CreateAudioCapturer() argument
531 DCHECK(audio_source); in CreateAudioCapturer()
533 constraints, NULL, audio_source); in CreateAudioCapturer()
|
| D | peer_connection_dependency_factory.h | 170 MediaStreamAudioSource* audio_source); in NON_EXPORTED_BASE()
|
| D | peer_connection_dependency_factory.cc | 589 MediaStreamAudioSource* audio_source) { in CreateAudioCapturer() argument
599 audio_source); in CreateAudioCapturer()
|
| D | mock_peer_connection_dependency_factory.h | 213 MediaStreamAudioSource* audio_source) OVERRIDE;
|
| /external/chromium_org/content/renderer/media/ |
| D | webrtc_audio_capturer.cc | 136 MediaStreamAudioSource* audio_source) { in CreateCapturer() argument
138 render_view_id, device_info, constraints, audio_device, audio_source); in CreateCapturer()
228 MediaStreamAudioSource* audio_source) in WebRtcAudioCapturer() argument
241 audio_source_(audio_source), in WebRtcAudioCapturer()
451 void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source, in Capture() argument
498 DCHECK_EQ(audio_source->channels(), in Capture()
500 DCHECK_EQ(audio_source->frames(), in Capture()
514 audio_power_monitor_.Scan(*audio_source, audio_source->frames()); in Capture()
527 audio_processor_->PushCaptureData(audio_source); in Capture()
|
| D | webrtc_audio_capturer.h | 62 MediaStreamAudioSource* audio_source);
132 MediaStreamAudioSource* audio_source);
136 virtual void Capture(const media::AudioBus* audio_source,
|
| D | webrtc_local_audio_renderer.cc | 73 scoped_ptr<media::AudioBus> audio_source = media::AudioBus::Create( in OnData() local
75 audio_source->FromInterleaved(audio_data, in OnData()
76 audio_source->frames(), in OnData()
78 loopback_fifo_->Push(audio_source.get()); in OnData()
|
| D | webrtc_local_audio_source_provider_unittest.cc | 41 blink::WebMediaStreamSource audio_source; in SetUp() local
42 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), in SetUp()
46 audio_source); in SetUp()
|
| D | media_stream_audio_processor.cc | 89 void Push(const media::AudioBus* audio_source) { in Push() argument
95 fifo_->Push(audio_source); in Push()
218 const media::AudioBus* audio_source) { in PushCaptureData() argument
220 DCHECK_EQ(audio_source->channels(), in PushCaptureData()
222 DCHECK_EQ(audio_source->frames(), in PushCaptureData()
225 capture_converter_->Push(audio_source); in PushCaptureData()
|
| D | webrtc_local_audio_track_unittest.cc | 176 MediaStreamAudioSource* audio_source = new MediaStreamAudioSource(); in SetUp() local
177 blink_source_.setExtraData(audio_source); in SetUp()
183 audio_source); in SetUp()
184 audio_source->SetAudioCapturer(capturer_); in SetUp()
|
| D | media_stream_audio_processor.h | 71 void PushCaptureData(const media::AudioBus* audio_source); in NON_EXPORTED_BASE()
|
| D | rtc_peer_connection_handler_unittest.cc | 228 blink::WebMediaStreamSource audio_source; in CreateLocalMediaStream() local
229 audio_source.initialize(blink::WebString::fromUTF8(audio_track_label), in CreateLocalMediaStream()
232 audio_source.setExtraData(new MediaStreamAudioSource()); in CreateLocalMediaStream()
243 audio_tracks[0].initialize(audio_source.id(), audio_source); in CreateLocalMediaStream()
|
| D | media_stream_impl.cc | 421 MediaStreamAudioSource* audio_source( in InitializeSourceObject() local
427 webkit_source->setExtraData(audio_source); in InitializeSourceObject()
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| D | audiotrack.cc | 36 AudioSourceInterface* audio_source) in AudioTrack() argument
38 audio_source_(audio_source) { in AudioTrack()
|
| D | audiotrack.h | 62 AudioTrack(const std::string& label, AudioSourceInterface* audio_source);
|
| D | peerconnectionfactory.h | 73 AudioSourceInterface* audio_source);
|
| /external/robolectric/src/main/java/com/xtremelabs/robolectric/shadows/ |
| D | ShadowMediaRecorder.java | 74 public void setAudioSource(int audio_source) { in setAudioSource() argument
75 audioSource = audio_source; in setAudioSource()
|
| /external/chromium_org/media/base/ |
| D | audio_capturer_source.h | 27 virtual void Capture(const AudioBus* audio_source,
|
| /external/chromium_org/media/audio/android/ |
| D | opensles_input.cc | 217 SLDataSource audio_source = {&mic_locator, NULL}; in CreateRecorder() local
234 &audio_source, in CreateRecorder()
|
| D | opensles_output.cc | 225 SLDataSource audio_source = {&simple_buffer_queue, &format_}; in CreatePlayer() local
240 &audio_source, in CreatePlayer()
|
| /external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
| D | opensles_input.cc | 336 SLDataSource audio_source = { &micLocator, NULL }; in CreateAudioRecorder() local
356 &audio_source, in CreateAudioRecorder()
|
| D | opensles_output.cc | 390 SLDataSource audio_source = { &simple_buf_queue, &configuration }; in CreateAudioPlayer() local
407 &audio_source, &audio_sink, in CreateAudioPlayer()
|