/external/qemu/distrib/sdl-1.2.15/src/audio/ |
H A D | SDL_audio.c | 37 /* Available audio drivers */ 127 SDL_AudioDevice *audio = (SDL_AudioDevice *)audiop; local 135 if ( audio->ThreadInit ) { 136 audio->ThreadInit(audio); 138 audio->threadid = SDL_ThreadID(); 141 fill = audio->spec.callback; 142 udata = audio->spec.userdata; 144 if ( audio->convert.needed ) { 145 if ( audio 239 SDL_LockAudio_Default(SDL_AudioDevice *audio) argument 247 SDL_UnlockAudio_Default(SDL_AudioDevice *audio) argument 302 SDL_AudioDevice *audio; local 399 SDL_AudioDevice *audio; local 574 SDL_AudioDevice *audio = current_audio; local 590 SDL_AudioDevice *audio = current_audio; local 599 SDL_AudioDevice *audio = current_audio; local 609 SDL_AudioDevice *audio = current_audio; local 624 SDL_AudioDevice *audio = current_audio; local [all...] |
/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/ppapi/c/ |
H A D | ppb_audio.h | 25 * realtime stereo audio streaming capabilities. 34 * <code>PPB_Audio_Callback</code> defines the type of an audio callback 35 * function used to fill the audio buffer with data. Please see the 39 * @param[in] sample_buffer A buffer to fill with audio data. 41 * @param[in] latency How long before the audio data is to be presented. 63 * for handling audio resources. Refer to the 64 * <a href="/native-client/devguide/coding/audio.html">Audio</a> 79 * ...Assume the application has cached the audio configuration interface in 80 * audio_config_interface and the audio interface in 96 * Create() creates an audio resourc [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/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/qemu/distrib/sdl-1.2.15/src/audio/mint/ |
H A D | SDL_mintaudio.c | 41 /* The audio device */ 45 unsigned long SDL_MintAudio_audiosize; /* Length of audio buffer=spec->size */ 63 SDL_AudioDevice *audio = SDL_MintAudio_device; local 66 SDL_memset(buffer, audio->spec.silence, audio->spec.size); 68 if (audio->paused) 71 if (audio->convert.needed) { 74 if ( audio->convert.src_format == AUDIO_U8 ) { 79 SDL_memset(audio->convert.buf, silence, audio [all...] |
/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/chromium_org/ppapi/proxy/ |
H A D | audio_buffer_resource.cc | 37 return buffer_->audio.timestamp; 45 buffer_->audio.timestamp = timestamp; 53 return buffer_->audio.sample_rate; 69 return buffer_->audio.number_of_channels; 77 return buffer_->audio.number_of_samples; 85 return buffer_->audio.data; 93 return buffer_->audio.data_size;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | output_audio_file.h | 37 virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE { 39 return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
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H A D | audio_sink.h | 21 // Interface class for an object receiving raw output audio from test 28 // Writes |num_samples| from |audio| to the AudioSink. Returns true if 30 virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0; 44 // Forks the output audio to two AudioSink objects. 50 virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE { 51 return left_sink_->WriteArray(audio, num_samples) && 52 right_sink_->WriteArray(audio, num_samples);
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H A D | audio_checksum.h | 30 virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE { 37 checksum_.Update(audio, num_samples * sizeof(*audio));
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/external/chromium_org/ppapi/api/ |
H A D | ppb_audio.idl | 8 * realtime stereo audio streaming capabilities. 17 * <code>PPB_Audio_Callback</code> defines the type of an audio callback 18 * function used to fill the audio buffer with data. Please see the 22 * @param[in] sample_buffer A buffer to fill with audio data. 24 * @param[in] latency How long before the audio data is to be presented. 35 * for handling audio resources. Refer to the 36 * <a href="/native-client/devguide/coding/audio.html">Audio</a> 51 * ...Assume the application has cached the audio configuration interface in 52 * audio_config_interface and the audio interface in 68 * Create() creates an audio resourc [all...] |
/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/webrtc/src/modules/audio_processing/ |
H A D | gain_control_impl.cc | 71 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) { argument 76 assert(audio->samples_per_split_channel() <= 160); 78 WebRtc_Word16* mixed_data = audio->low_pass_split_data(0); 79 if (audio->num_channels() > 1) { 80 audio->CopyAndMixLowPass(1); 81 mixed_data = audio->mixed_low_pass_data(0); 89 static_cast<WebRtc_Word16>(audio->samples_per_split_channel())); 99 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { argument 104 assert(audio->samples_per_split_channel() <= 160); 105 assert(audio 149 ProcessCaptureAudio(AudioBuffer* audio) argument [all...] |
H A D | level_estimator_impl.cc | 93 int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) { argument 99 if (audio->is_muted()) { 100 level->ProcessMuted(audio->samples_per_channel()); 104 int16_t* mixed_data = audio->data(0); 105 if (audio->num_channels() > 1) { 106 audio->CopyAndMix(1); 107 mixed_data = audio->mixed_data(0); 110 level->Process(mixed_data, audio->samples_per_channel());
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H A D | voice_detection_impl.cc | 58 int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) { argument 67 assert(audio->samples_per_split_channel() <= 160); 69 WebRtc_Word16* mixed_data = audio->low_pass_split_data(0); 70 if (audio->num_channels() > 1) { 71 audio->CopyAndMixLowPass(1); 72 mixed_data = audio->mixed_low_pass_data(0); 83 audio->set_activity(AudioFrame::kVadPassive); 86 audio->set_activity(AudioFrame::kVadActive);
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/external/chromium_org/ppapi/tests/ |
H A D | test_audio.cc | 140 // Test creating audio resources for all guaranteed sample rates and various 168 // Make a config, create the audio resource, and release the config. 175 PP_Resource audio = audio_interface_->Create( local 180 ASSERT_TRUE(audio); 181 ASSERT_TRUE(audio_interface_->IsAudio(audio)); 183 // Check that the config returned for |audio| matches what we gave it. 184 ac = audio_interface_->GetCurrentConfig(audio); 192 // Start and stop audio playback. The documentation indicates that 196 ASSERT_TRUE(audio_interface_->StartPlayback(audio)); 197 ASSERT_TRUE(audio_interface_->StopPlayback(audio)); 212 PP_Resource audio = audio_interface_->Create( local 240 PP_Resource audio = audio_interface_->Create( local 279 PP_Resource audio = audio_interface_->Create( local 309 PP_Resource audio = audio_interface_->Create( local 340 PP_Resource audio = audio_interface_->Create( local 380 PP_Resource audio = audio_interface_1_0_->Create( local 421 PP_Resource audio = audio_interface_->Create( local 455 PP_Resource audio = audio_interface_->Create( local [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
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...] |
H A D | gain_control_impl.cc | 55 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) { argument 60 assert(audio->samples_per_split_channel() <= 160); 66 audio->mixed_low_pass_data(), 67 static_cast<int16_t>(audio->samples_per_split_channel())); 77 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) { argument 82 assert(audio->samples_per_split_channel() <= 160); 83 assert(audio->num_channels() == num_handles()); 93 audio->low_pass_split_data(i), 94 audio->high_pass_split_data(i), 95 static_cast<int16_t>(audio 127 ProcessCaptureAudio(AudioBuffer* audio) argument [all...] |
/external/qemu/distrib/sdl-1.2.15/src/audio/macrom/ |
H A D | SDL_romaudio.c | 96 #ifdef __MACOSX__ /* Mac OS X uses threaded audio, so normal thread code is okay */ 121 static void mix_buffer(SDL_AudioDevice *audio, UInt8 *buffer) argument 123 if ( ! audio->paused ) { 125 SDL_mutexP(audio->mixer_lock); 127 if ( audio->convert.needed ) { 128 audio->spec.callback(audio->spec.userdata, 129 (Uint8 *)audio->convert.buf,audio->convert.len); 130 SDL_ConvertAudio(&audio 173 SDL_AudioDevice *audio = (SDL_AudioDevice *)chan->userInfo; local 336 SDL_AudioDevice *audio = (SDL_AudioDevice *)newbuf->dbUserInfo[0]; local [all...] |
/external/chromium_org/third_party/WebKit/Source/core/css/ |
H A D | mediaControlsAndroid.css | 27 /* WARNING: This css file can only style <audio> and <video> elements */ 29 audio { 33 audio::-webkit-media-controls-enclosure { 41 audio::-webkit-media-controls-panel, video::-webkit-media-controls-panel { 45 audio::-webkit-media-controls-mute-button, video::-webkit-media-controls-mute-button { 49 audio::-webkit-media-controls-play-button, video::-webkit-media-controls-play-button { 55 audio::-webkit-media-controls-current-time-display, video::-webkit-media-controls-current-time-display, 56 audio::-webkit-media-controls-time-remaining-display, video::-webkit-media-controls-time-remaining-display { 62 audio::-webkit-media-controls-volume-slider, video::-webkit-media-controls-volume-slider { 72 audio [all...] |
/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/bluetooth/bluedroid/audio_a2dp_hw/ |
H A D | Android.mk | 14 LOCAL_MODULE := audio.a2dp.default
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/external/chromium_org/chrome/browser/extensions/api/audio/ |
H A D | audio_api.cc | 5 #include "chrome/browser/extensions/api/audio/audio_api.h" 10 #include "chrome/common/extensions/api/audio.h" 15 namespace audio = api::audio; 43 audio::OnDeviceChanged::kEventName, 62 results_ = api::audio::GetInfo::Results::Create(output_info, input_info); 64 SetError("Error occurred when querying audio device information."); 69 scoped_ptr<api::audio::SetActiveDevices::Params> params( 70 api::audio::SetActiveDevices::Params::Create(*args_)); 82 scoped_ptr<api::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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