/external/chromium_org/third_party/WebKit/Source/core/html/ |
H A D | HTMLAudioElement.cpp | 43 RefPtrWillBeRawPtr<HTMLAudioElement> audio = adoptRefWillBeNoop(new HTMLAudioElement(document)); local 44 audio->ensureUserAgentShadowRoot(); 45 audio->suspendIfNeeded(); 46 return audio.release(); 51 RefPtrWillBeRawPtr<HTMLAudioElement> audio = adoptRefWillBeNoop(new HTMLAudioElement(document)); local 52 audio->ensureUserAgentShadowRoot(); 53 audio->setPreload(AtomicString("auto", AtomicString::ConstructFromLiteral)); 55 audio->setSrc(src); 56 audio->suspendIfNeeded(); 57 return audio [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | audioframe.h | 41 AudioFrame(int16* audio, size_t audio_length, int sample_freq, bool stereo) argument 42 : audio10ms_(audio),
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_utility.h | 41 bool audio; member in struct:webrtc::RtpUtility::Payload
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H A D | rtp_payload_registry_unittest.cc | 41 bool audio = true; local 44 audio, 45 {// Initialize the audio struct in this case. 106 EXPECT_FALSE(retrieved_payload->audio);
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | transmit_mixer_unittest.cc | 23 int16_t audio[], int samples_per_channel, 22 Process(int channel, ProcessingTypes type, int16_t audio[], int samples_per_channel, int sample_rate_hz, bool is_stereo) argument
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/external/chromium_org/content/browser/renderer_host/media/ |
H A D | media_stream_ui_controller_unittest.cc | 58 void CreateDummyRequest(const std::string& label, bool audio, bool video) { argument 61 StreamOptions components(audio, video ); 115 // Create the first audio request. 123 // Create the third audio and video request. 143 // Create the first audio request. 151 // Create the third audio and video request.
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H A D | peer_connection_tracker_host.cc | 96 bool audio, 103 audio, 94 OnGetUserMedia( const std::string& origin, bool audio, bool video, const std::string& audio_constraints, const std::string& video_constraints) argument
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/external/chromium_org/ppapi/proxy/ |
H A D | plugin_dispatcher_unittest.cc | 29 PP_Resource GetCurrentConfig(PP_Resource audio) { argument 32 PP_Bool StartPlayback(PP_Resource audio) { argument 35 PP_Bool StopPlayback(PP_Resource audio) { argument
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/external/chromium_org/ppapi/thunk/ |
H A D | ppb_audio_thunk.cc | 53 PP_Resource GetCurrentConfig(PP_Resource audio) { argument 55 EnterResource<PPB_Audio_API> enter(audio, true); 61 PP_Bool StartPlayback(PP_Resource audio) { argument 63 EnterResource<PPB_Audio_API> enter(audio, true); 69 PP_Bool StopPlayback(PP_Resource audio) { argument 71 EnterResource<PPB_Audio_API> enter(audio, true);
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/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
H A D | level_estimator_impl.cc | 27 int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) { argument 33 for (int i = 0; i < audio->num_channels(); ++i) { 34 rms_level->Process(audio->data(i), audio->samples_per_channel());
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H A D | noise_suppression_impl.cc | 58 int NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { argument 63 assert(audio->samples_per_split_channel() <= 160); 64 assert(audio->num_channels() == num_handles()); 70 audio->low_pass_split_data_f(i)); 79 int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { argument 85 assert(audio->samples_per_split_channel() <= 160); 86 assert(audio->num_channels() == num_handles()); 92 audio->low_pass_split_data_f(i), 93 audio->high_pass_split_data_f(i), 94 audio [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
H A D | rtp_rtcp.h | 33 * audio - True for a audio version of the RTP/RTCP module 58 bool audio; member in struct:webrtc::RtpRtcp::Configuration 315 * Used by the codec module to deliver a video or audio frame for 622 * set audio packet size, used to determine when it's time to send a DTMF 667 * Store the audio level in dBov for header-extension-for-audio-level-
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/external/chromium_org/third_party/webrtc/tools/e2e_quality/audio/ |
H A D | audio_e2e_harness.cc | 11 // Sets up a simple VoiceEngine loopback call with the default audio devices 36 VoEAudioProcessing* audio = VoEAudioProcessing::GetInterface(voe); local 37 ASSERT_TRUE(audio != NULL); 87 // Disable all audio processing. 88 ASSERT_EQ(0, audio->SetAgcStatus(false)); 89 ASSERT_EQ(0, audio->SetEcStatus(false)); 90 ASSERT_EQ(0, audio->EnableHighPassFilter(false)); 91 ASSERT_EQ(0, audio->SetNsStatus(false));
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/external/chromium_org/third_party/webrtc/voice_engine/include/mock/ |
H A D | fake_voe_external_media.h | 51 // If |audio| is NULL, a zero array of the correct length will be forwarded. 52 void CallProcess(ProcessingTypes type, int16_t* audio, argument 57 if (!audio) { 60 audio = data.get(); 66 it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
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/external/qemu/distrib/sdl-1.2.15/src/audio/baudio/ |
H A D | SDL_beaudio.cc | 24 /* Allow access to the audio stream on BeOS */ 96 /* The BeOS callback for handling the audio buffer */ 100 SDL_AudioDevice *audio = (SDL_AudioDevice *)device; local 103 SDL_memset(stream, audio->spec.silence, len); 105 /* Only do soemthing if audio is enabled */ 106 if ( ! audio->enabled ) 109 if ( ! audio->paused ) { 110 if ( audio->convert.needed ) { 111 SDL_mutexP(audio->mixer_lock); 112 (*audio [all...] |
/external/chromium_org/content/renderer/media/ |
H A D | mock_media_stream_dispatcher.cc | 109 StreamDeviceInfo audio; local 110 audio.device.id = "audio_input_device_id" + base::IntToString(session_id_); 111 audio.device.name = "microphone"; 112 audio.device.type = MEDIA_DEVICE_AUDIO_CAPTURE; 113 audio.device.video_facing = MEDIA_VIDEO_FACING_NONE; 115 audio.device.matched_output_device_id = 118 audio.session_id = session_id_; 119 audio_input_array_.push_back(audio); 123 StreamDeviceInfo audio; local 124 audio [all...] |
/external/chromium_org/content/renderer/media/webrtc/ |
H A D | webrtc_media_stream_adapter_unittest.cc | 37 blink::WebMediaStream CreateBlinkMediaStream(bool audio, bool video) { argument 39 audio ? static_cast<size_t>(1) : 0); 40 if (audio) { 42 audio_source.initialize("audio", 44 "audio"); 113 // audio sources. This can happen if a MediaStream is created with 114 // remote audio track. 119 audio_source.initialize("audio source",
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/external/chromium_org/ppapi/shared_impl/ |
H A D | media_stream_buffer.h | 53 Audio audio; member in union:ppapi::MediaStreamBuffer
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/external/chromium_org/third_party/WebKit/Source/web/ |
H A D | WebUserMediaRequest.cpp | 60 bool WebUserMediaRequest::audio() const function in class:blink::WebUserMediaRequest 63 return m_private->audio();
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | mediamessages.h | 50 // A collection of audio and video and data streams. Most of the 66 const std::vector<StreamParams>& audio() const { return audio_; } function in struct:cricket::MediaStreams
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
H A D | coder.cc | 80 int32_t AudioCoder::Encode(const AudioFrame& audio, argument 84 // Fake a timestamp in case audio doesn't contain a correct timestamp. 85 // Make a local copy of the audio frame since audio is const 87 audioFrame.CopyFrom(audio);
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/external/qemu/distrib/sdl-1.2.15/src/audio/nds/ |
H A D | SDL_ndsaudio.c | 118 SDL_AudioDevice *audio = (SDL_AudioDevice *)sdl_nds_audiodevice; local 121 SDL_memset(stream, audio->spec.silence, len); 123 /* Only do soemthing if audio is enabled */ 124 if ( ! audio->enabled ) 127 if ( ! audio->paused ) { 128 if ( audio->convert.needed ) { 129 //fprintf(stderr,"converting audio\n"); 130 SDL_mutexP(audio->mixer_lock); 131 (*audio->spec.callback)(audio [all...] |
/external/webrtc/src/modules/audio_processing/ |
H A D | noise_suppression_impl.cc | 57 int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) { argument 63 assert(audio->samples_per_split_channel() <= 160); 64 assert(audio->num_channels() == num_handles()); 70 audio->low_pass_split_data(i), 71 audio->high_pass_split_data(i), 72 audio->low_pass_split_data(i), 73 audio->high_pass_split_data(i)); 76 audio->low_pass_split_data(i), 77 audio->high_pass_split_data(i), 78 audio [all...] |
/external/chromium_org/chrome/common/extensions/docs/examples/extensions/talking_alarm_clock/ |
H A D | common.js | 16 var audio = null; variable 60 if (audio) { 61 audio.pause(); 74 if (audio) { 75 audio.pause(); 76 document.body.removeChild(audio); 77 audio = null; 85 audio = document.createElement('audio'); 86 audio [all...] |
/external/chromium_org/content/browser/media/ |
H A D | webrtc_internals_unittest.cc | 78 const std::string& audio, 86 VerifyString(dict, "audio", audio); 74 VerifyGetUserMediaData(base::Value* actual_data, int rid, int pid, const std::string& origin, const std::string& audio, const std::string& video) argument
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