Searched refs:audio (Results 1 - 25 of 296) sorted by relevance

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/external/qemu/distrib/sdl-1.2.15/src/audio/
H A DSDL_audio.c37 /* 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
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/external/chromium_org/third_party/WebKit/Source/core/html/
H A DHTMLAudioElement.cpp43 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
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/external/chromium_org/ppapi/c/
H A Dppb_audio.h25 * 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
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/external/chromium_org/content/renderer/media/
H A Dmock_media_stream_dispatcher.cc109 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
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/external/qemu/distrib/sdl-1.2.15/src/audio/baudio/
H A DSDL_beaudio.cc24 /* 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
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/external/qemu/distrib/sdl-1.2.15/src/audio/mint/
H A DSDL_mintaudio.c41 /* 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
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/external/qemu/distrib/sdl-1.2.15/src/audio/nds/
H A DSDL_ndsaudio.c118 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
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/external/chromium_org/ppapi/proxy/
H A Daudio_buffer_resource.cc37 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;
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/
H A Doutput_audio_file.h37 virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
39 return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
H A Daudio_sink.h21 // 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);
H A Daudio_checksum.h30 virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
37 checksum_.Update(audio, num_samples * sizeof(*audio));
/external/chromium_org/ppapi/api/
H A Dppb_audio.idl8 * 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
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/external/chromium_org/ppapi/thunk/
H A Dppb_audio_thunk.cc53 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);
/external/webrtc/src/modules/audio_processing/
H A Dgain_control_impl.cc71 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
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H A Dlevel_estimator_impl.cc93 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());
H A Dvoice_detection_impl.cc58 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);
/external/chromium_org/ppapi/tests/
H A Dtest_audio.cc140 // 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
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/external/chromium_org/third_party/webrtc/modules/audio_processing/
H A Dnoise_suppression_impl.cc58 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
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H A Dgain_control_impl.cc55 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
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/external/qemu/distrib/sdl-1.2.15/src/audio/macrom/
H A DSDL_romaudio.c96 #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
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/external/chromium_org/third_party/WebKit/Source/core/css/
H A DmediaControlsAndroid.css27 /* 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
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/external/chromium_org/third_party/webrtc/tools/e2e_quality/audio/
H A Daudio_e2e_harness.cc11 // 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));
/external/bluetooth/bluedroid/audio_a2dp_hw/
H A DAndroid.mk14 LOCAL_MODULE := audio.a2dp.default
/external/chromium_org/chrome/browser/extensions/api/audio/
H A Daudio_api.cc5 #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
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/external/chromium_org/third_party/libjingle/source/talk/media/base/
H A Daudioframe.h41 AudioFrame(int16* audio, size_t audio_length, int sample_freq, bool stereo) argument
42 : audio10ms_(audio),

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