| /external/chromium_org/third_party/WebKit/Source/core/html/ |
| 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.release()
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_utility.h | 41 bool audio; member in struct:webrtc::RtpUtility::Payload
|
| 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);
|
| /external/chromium_org/content/renderer/media/ |
| mock_media_stream_dispatcher.cc | 35 // Audio and video share the same request so we use |audio_input_request_id_|
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
[all...] |
| /external/chromium_org/third_party/WebKit/Source/web/ |
| WebUserMediaRequest.cpp | 60 bool WebUserMediaRequest::audio() const function in class:blink::WebUserMediaRequest
63 return m_private->audio();
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| 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
617 * Audio
622 * set audio packet size, used to determine when it's time to send a DTMF
651 * Set payload type for Redundant Audio Data RFC 2198
659 * Get payload type for Redundant Audio Data RFC 2198
667 * Store the audio level in dBov for header-extension-for-audio-level
[all...] |
| /external/chromium_org/third_party/webrtc/tools/e2e_quality/audio/ |
| 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));
|
| /external/chromium_org/ppapi/shared_impl/ |
| media_stream_buffer.h | 21 struct Audio {
28 // Uses 8 bytes to make sure the Audio struct has consistent size between
48 PP_COMPILE_ASSERT_SIZE_IN_BYTES(Audio, 40);
53 Audio audio; member in union:ppapi::MediaStreamBuffer
|
| ppb_audio_shared.cc | 116 // Setup integer audio buffer for user audio data.
131 // Clear contents of shm buffer before starting audio thread. This will
132 // prevent a burst of static if for some reason the audio thread doesn't
157 // In general, the audio thread should not do Pepper calls, but it might
158 // anyway (for example, our Audio test does CallOnMainThread). If it did a
202 PPB_Audio_Shared* audio = static_cast<PPB_Audio_Shared*>(self); local
203 audio->Run();
217 TRACE_EVENT0("audio", "PPB_Audio_Shared::FireRenderCallback");
224 // Deinterleave the audio data into the shared memory as floats
[all...] |
| /external/chromium_org/third_party/WebKit/Source/modules/mediastream/ |
| UserMediaRequest.cpp | 70 WebMediaConstraints audio = parseOptions(options, "audio", exceptionState); local
78 if (audio.isNull() && video.isNull()) {
79 exceptionState.throwDOMException(SyntaxError, "At least one of audio and video must be requested");
83 return new UserMediaRequest(context, controller, audio, video, successCallback, errorCallback);
86 UserMediaRequest::UserMediaRequest(ExecutionContext* context, UserMediaController* controller, WebMediaConstraints audio, WebMediaConstraints video, NavigatorUserMediaSuccessCallback* successCallback, NavigatorUserMediaErrorCallback* errorCallback)
88 , m_audio(audio)
100 bool UserMediaRequest::audio() const function in class:blink::UserMediaRequest
|
| /external/chromium_org/third_party/WebKit/Source/platform/exported/ |
| WebMediaStream.cpp | 142 MediaStreamComponentVector audio, video; local
145 audio.append(component);
151 m_private = MediaStreamDescriptor::create(label, audio, video);
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| 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
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| opus_test.cc | 214 const int kBufferSizeSamples = 480 * 12 * 2; // Can hold 120 ms stereo audio.
215 int16_t audio[kBufferSizeSamples]; local
244 // Get 10 msec of audio.
257 // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
264 &audio[written_samples]));
267 // Sometimes we need to loop over the audio vector to produce the right
279 opus_mono_encoder_, &audio[read_samples],
284 opus_stereo_encoder_, &audio[read_samples],
|
| target_delay_unittest.cc | 43 rtp_info_.type.Audio.channel = 1;
44 rtp_info_.type.Audio.isCNG = false;
47 int16_t audio[kFrameSizeSamples]; local
50 audio[n] = (rand() & kRange) - kRange / 2;
51 WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_);
150 // Pull audio equivalent to the amount of audio in one RTP packet.
|
| /external/chromium_org/chrome/common/extensions/docs/examples/extensions/talking_alarm_clock/ |
| 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.addEventListener('ended', function(evt)
[all...] |
| /external/chromium_org/media/audio/ |
| audio_parameters.h | 22 // size as sizeof(AudioInputBufferParameters) + #(bytes in audio buffer) without
26 int8 audio[1]; member in struct:media::AudioInputBuffer
41 // Telephone quality sample rate, mostly for speech-only audio.
47 // Bitmasks to determine whether certain platform (typically hardware) audio
75 // Returns size of audio buffer in bytes.
78 // Returns the number of bytes representing one second of audio.
81 // Returns the number of bytes representing a frame of audio.
|
| /external/bluetooth/bluedroid/main/ |
| bte_main.c | 516 * Definitions for audio state structure, this type needs to match to
527 bt_hc_audio_state_t audio; member in struct:bt_audio_state_tag
534 ** Description Sets audio state on controller state for SCO (PCM, WBS, FM)
556 p_msg->audio.handle = handle;
557 p_msg->audio.peer_codec = codec;
558 p_msg->audio.state = state;
561 p_msg->hdr.len = sizeof(p_msg->audio);
568 bt_hc_if->tx_cmd((TRANSAC)p_msg, (char *)(&p_msg->audio), sizeof(*p_msg));
|
| /external/chromium_org/chrome/browser/media/ |
| media_stream_capture_indicator.cc | 353 void MediaStreamCaptureIndicator::MaybeCreateStatusTrayIcon(bool audio,
371 GetStatusTrayIconInfo(audio, video, &image, &tool_tip);
414 bool audio = false; local
421 // Check if any audio and video devices have been used.
428 // The audio/video icon is shown only for non-whitelisted extensions or on
437 audio = audio || usage.IsCapturingAudio();
459 MaybeCreateStatusTrayIcon(audio, video);
466 bool audio,
471 DCHECK(audio || video)
[all...] |
| /external/chromium_org/media/filters/ |
| ffmpeg_demuxer_unittest.cc | 254 // Open a file containing streams but none of which are audio/video streams.
284 // Audio stream should be present.
285 stream = demuxer_->GetStream(DemuxerStream::AUDIO);
287 EXPECT_EQ(DemuxerStream::AUDIO, stream->type());
305 // Stream #1: Audio (Vorbis)
308 // Stream #4: Audio (16-bit signed little endian PCM)
310 // We should only pick the first audio/video streams we come across.
320 // Audio stream should be Vorbis.
321 stream = demuxer_->GetStream(DemuxerStream::AUDIO);
323 EXPECT_EQ(DemuxerStream::AUDIO, stream->type())
375 DemuxerStream* audio = demuxer_->GetStream(DemuxerStream::AUDIO); local
432 DemuxerStream* audio = demuxer_->GetStream(DemuxerStream::AUDIO); local
492 DemuxerStream* audio = demuxer_->GetStream(DemuxerStream::AUDIO); local
538 DemuxerStream* audio = demuxer_->GetStream(DemuxerStream::AUDIO); local
644 DemuxerStream* audio = demuxer_->GetStream(DemuxerStream::AUDIO); local
688 DemuxerStream* audio = demuxer_->GetStream(DemuxerStream::AUDIO); local
745 DemuxerStream* audio = demuxer_->GetStream(DemuxerStream::AUDIO); local
770 DemuxerStream* audio = demuxer_->GetStream(DemuxerStream::AUDIO); local
[all...] |
| /external/chromium_org/ppapi/tests/ |
| test_audio.cc | 96 REGISTER_TEST_CASE(Audio);
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));
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_coding/codecs/opus/ |
| opus_interface.c | 40 /* Default to VoIP application for mono, and AUDIO for stereo. */
68 opus_int16* audio = (opus_int16*) audio_in; local
76 res = opus_encode(inst->encoder, audio, samples, coded,
229 opus_int16* audio = (opus_int16*) decoded; local
231 int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 0);
246 opus_int16* audio = (opus_int16*) decoded; local
248 int res = opus_decode(inst, coded, encoded_bytes, audio, frame_size, 1);
|
| /external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
| opensles_input.cc | 268 // On average half the current buffer will have been filled with audio.
315 // no audio can have been returned yet meaning fifo must be empty. Any other
346 // Interfaces for recording android audio data and Android are needed.
404 WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, id_, "Audio overrun");
411 // All buffers passed to OpenSL have been flushed. Restart the audio from
441 int8_t* audio = rec_buf_[active_queue_].get(); local
443 fifo_->Push(audio);
514 // If the fifo_ has audio data process it.
516 int8_t* audio = fifo_->Pop(); local
517 audio_buffer_->SetRecordedBuffer(audio, buffer_size_samples())
[all...] |
| opensles_output.cc | 326 // |num_fifo_buffers_needed_| is a multiple of 10ms of buffered up audio.
398 // Interfaces for streaming audio data, setting volume and Android are needed.
453 WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, id_, "Audio underrun");
460 // All buffers have been flushed. Restart the audio from scratch.
489 int8_t* audio = fifo_->Pop(); local
490 if (audio)
493 audio,
557 int8_t* audio = play_buf_[active_queue_].get(); local
558 fine_buffer_->GetBufferData(audio);
559 fifo_->Push(audio);
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/ |
| stream_synchronization_unittest.cc | 85 // the audio and video delays needed to get the two streams in sync.
88 // |current_audio_delay_ms| is the number of milliseconds which audio is
101 StreamSynchronization::Measurements audio; local
104 audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
112 audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency,
121 // Capture an audio and a video frame at the same time.
122 audio.latest_timestamp = send_time_->NowRtp(audio_frequency,
128 // Audio later than video.
132 audio.latest_receive_time_ms = receive_time_->time_now_ms();
134 // Video later than audio
[all...] |
| /external/chromium_org/content/browser/speech/ |
| google_streaming_remote_engine.cc | 368 const AudioChunk& audio = *(event_args.audio_data.get()); local
370 DCHECK_EQ(audio.bytes_per_sample(), config_.audio_num_bits_per_sample / 8);
371 encoder_->Encode(audio);
|
