| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| audiotrack.cc | 36 AudioSourceInterface* audio_source)
38 audio_source_(audio_source) {
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| audiotrack.h | 62 AudioTrack(const std::string& label, AudioSourceInterface* audio_source);
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| peerconnectionfactory.h | 73 AudioSourceInterface* audio_source);
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| /external/chromium_org/content/renderer/media/webrtc/ |
| webrtc_media_stream_adapter_unittest.cc | 41 blink::WebMediaStreamSource audio_source; local
42 audio_source.initialize("audio",
45 audio_source.setExtraData(new MediaStreamAudioSource());
47 audio_track_vector[0].initialize(audio_source);
118 blink::WebMediaStreamSource audio_source; local
119 audio_source.initialize("audio source",
125 audio_tracks[0].initialize(audio_source.id(), audio_source);
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| webrtc_media_stream_adapter.cc | 94 MediaStreamAudioSource* audio_source = local
96 if (audio_source && audio_source->GetAudioCapturer().get())
97 audio_source->GetAudioCapturer()->EnablePeerConnectionMode();
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| mock_peer_connection_dependency_factory.cc | 525 MediaStreamAudioSource* audio_source) {
530 DCHECK(audio_source);
532 constraints, NULL, audio_source);
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| peer_connection_dependency_factory.h | 170 MediaStreamAudioSource* audio_source);
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| mock_peer_connection_dependency_factory.h | 213 MediaStreamAudioSource* audio_source) OVERRIDE;
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| peer_connection_dependency_factory.cc | 598 MediaStreamAudioSource* audio_source) {
608 audio_source);
|
| /external/chromium_org/media/base/ |
| audio_capturer_source.h | 27 virtual void Capture(const AudioBus* audio_source,
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| /external/chromium_org/content/renderer/media/ |
| webrtc_audio_capturer.h | 62 MediaStreamAudioSource* audio_source);
132 MediaStreamAudioSource* audio_source);
136 virtual void Capture(const media::AudioBus* audio_source,
|
| webrtc_audio_capturer.cc | 33 // Method to check if any of the data in |audio_source| has energy.
34 bool HasDataEnergy(const media::AudioBus& audio_source) {
35 for (int ch = 0; ch < audio_source.channels(); ++ch) {
36 const float* channel_ptr = audio_source.channel(ch);
37 for (int frame = 0; frame < audio_source.frames(); ++frame) {
136 MediaStreamAudioSource* audio_source) {
138 render_view_id, device_info, constraints, audio_device, audio_source);
235 MediaStreamAudioSource* audio_source)
248 audio_source_(audio_source),
458 void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source,
[all...] |
| webrtc_local_audio_source_provider_unittest.cc | 42 blink::WebMediaStreamSource audio_source; variable
43 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
47 audio_source);
|
| media_stream_audio_processor.h | 71 // Pushes capture data in |audio_source| to the internal FIFO. Each call to
75 void PushCaptureData(const media::AudioBus* audio_source);
|
| webrtc_local_audio_track_unittest.cc | 177 MediaStreamAudioSource* audio_source = new MediaStreamAudioSource(); variable
178 blink_source_.setExtraData(audio_source);
184 audio_source);
185 audio_source->SetAudioCapturer(capturer_.get());
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| rtc_peer_connection_handler_unittest.cc | 237 blink::WebMediaStreamSource audio_source; local
238 audio_source.initialize(blink::WebString::fromUTF8(audio_track_label),
241 audio_source.setExtraData(new MediaStreamAudioSource());
252 audio_tracks[0].initialize(audio_source.id(), audio_source);
[all...] |
| media_stream_audio_processor.cc | 226 const media::AudioBus* audio_source) {
229 capture_fifo_->Push(audio_source);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| acm_send_test.h | 30 AcmSendTest(InputAudioFile* audio_source,
|
| acm_send_test_oldapi.h | 31 AcmSendTestOldApi(InputAudioFile* audio_source,
|
| acm_send_test.cc | 26 AcmSendTest::AcmSendTest(InputAudioFile* audio_source,
30 audio_source_(audio_source),
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| acm_send_test_oldapi.cc | 26 AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
31 audio_source_(audio_source),
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| /external/chromium_org/media/audio/android/ |
| opensles_input.cc | 222 SLDataSource audio_source = {&mic_locator, NULL}; local
239 &audio_source,
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| opensles_output.cc | 225 SLDataSource audio_source = {&simple_buffer_queue, &format_}; local
240 &audio_source,
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| /external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
| opensles_input.cc | 336 SLDataSource audio_source = { &micLocator, NULL }; local
356 &audio_source,
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| opensles_output.cc | 390 SLDataSource audio_source = { &simple_buf_queue, &configuration }; local
407 &audio_source, &audio_sink,
|
