Searched defs:audio (Results 1 - 25 of 65) sorted by relevance

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/external/autotest/client/site_tests/webrtc_PausePlayPeerConnections/
H A Dpause-play.js43 * - localConnection is fed video/audio from source
50 * @param {!HTMLMediaElement} element - And 'audio' or 'video' element.
123 const constraints = {audio: true};
128 } else if (elementType === 'audio') {
131 throw new Error('elementType must be one of "audio" or "video"');
/external/ltp/testcases/kernel/device-drivers/v4l/user_space/
H A Dtest_VIDIOC_ENUMAUDIO.c41 struct v4l2_audio audio; local
47 memset(&audio, 0xff, sizeof(audio));
48 audio.index = i;
49 ret_enum = ioctl(get_video_fd(), VIDIOC_ENUMAUDIO, &audio);
58 CU_ASSERT_EQUAL(audio.index, i);
60 CU_ASSERT(0 < strlen((char *)audio.name));
62 ((char *)audio.name, sizeof(audio.name)));
64 //CU_ASSERT_EQUAL(audio
110 struct v4l2_audio audio; local
130 struct v4l2_audio audio; local
150 struct v4l2_audio audio; local
171 struct v4l2_audio audio; local
[all...]
H A Dtest_VIDIOC_AUDIO.c67 struct v4l2_audio audio; local
70 memset(&audio, 0xff, sizeof(audio));
71 ret_get = ioctl(get_video_fd(), VIDIOC_G_AUDIO, &audio);
80 //CU_ASSERT_EQUAL(audio.index, ?);
82 CU_ASSERT(0 < strlen((char *)audio.name));
83 CU_ASSERT(valid_string((char *)audio.name, sizeof(audio.name)));
85 CU_ASSERT(valid_audio_capability(audio.capability));
86 CU_ASSERT(valid_audio_mode(audio
129 struct v4l2_audio audio; local
166 struct v4l2_audio audio; local
296 struct v4l2_audio audio; local
353 struct v4l2_audio audio; local
410 struct v4l2_audio audio; local
[all...]
/external/webrtc/talk/media/base/
H A Daudioframe.h41 AudioFrame(int16_t* audio, size_t audio_length, int sample_freq, bool stereo) argument
42 : audio10ms_(audio),
/external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/
H A Daudio_encoder_pcm16b.cc19 size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio, argument
22 return WebRtcPcm16b_Encode(audio, input_len, encoded);
/external/webrtc/webrtc/modules/audio_processing/
H A Dlevel_estimator_impl.cc31 void LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) { argument
32 RTC_DCHECK(audio);
38 for (size_t i = 0; i < audio->num_channels(); i++) {
39 rms_->Process(audio->channels_const()[i], audio->num_frames());
H A Dhigh_pass_filter_impl.cc104 void HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) { argument
105 RTC_DCHECK(audio);
111 RTC_DCHECK_GE(160u, audio->num_frames_per_band());
112 RTC_DCHECK_EQ(filters_.size(), audio->num_channels());
114 filters_[i]->Process(audio->split_bands(i)[kBand0To8kHz],
115 audio->num_frames_per_band());
/external/webrtc/webrtc/modules/audio_processing/vad/
H A Dvoice_activity_detector.cc37 void VoiceActivityDetector::ProcessChunk(const int16_t* audio, argument
43 const int16_t* resampled_ptr = audio;
48 resampler_.Push(audio, length, resampled_, kLength10Ms, length);
54 // buffers the audio and processes it all at once when GetActivity() is
/external/webrtc/webrtc/modules/rtp_rtcp/source/
H A Drtp_utility.h36 bool audio; member in struct:webrtc::RtpUtility::Payload
/external/webrtc/webrtc/voice_engine/
H A Dtransmit_mixer_unittest.cc23 int16_t audio[], size_t samples_per_channel,
22 Process(int channel, ProcessingTypes type, int16_t audio[], size_t samples_per_channel, int sample_rate_hz, bool is_stereo) argument
/external/webrtc/webrtc/modules/audio_coding/codecs/
H A Daudio_encoder.cc28 rtc::ArrayView<const int16_t> audio,
32 RTC_CHECK_EQ(audio.size(),
35 EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded);
26 Encode( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
/external/webrtc/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/webrtc/talk/app/webrtc/
H A Dremoteaudiosource.cc63 void OnData(const AudioSinkInterface::Data& audio) override {
65 source_->OnData(audio);
154 void RemoteAudioSource::OnData(const AudioSinkInterface::Data& audio) { argument
155 // Called on the externally-owned audio callback thread, via/from webrtc.
158 sink->OnData(audio.data, 16, audio.sample_rate, audio.channels,
159 audio.samples_per_channel);
H A Dpeerconnectionendtoend_unittest.cc96 void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints, argument
98 caller_->GetAndAddUserMedia(audio, audio_constraints,
100 callee_->GetAndAddUserMedia(audio, audio_constraints,
/external/webrtc/webrtc/modules/audio_coding/acm2/
H A Daudio_coding_module_impl.h65 // Add 10 ms of raw (PCM) audio data to the encoder.
157 // Get 10 milliseconds of raw audio data to play out, and
190 const int16_t* audio; member in struct:webrtc::acm2::final::InputData
224 // Preprocessing of input audio, including resampling and down-mixing if
225 // required, before pushing audio into encoder's buffer.
227 // in_frame: input audio-frame
H A Drent_a_codec_unittest.cc117 int16_t audio[kPacketSizeSamples] = {0}; local
125 EncodeInternal(0, rtc::ArrayView<const int16_t>(audio),
132 EncodeInternal(2, rtc::ArrayView<const int16_t>(audio),
138 info = rac.GetEncoderStack()->Encode(0, audio, arraysize(encoded), encoded);
150 info = rac.GetEncoderStack()->Encode(1, audio, arraysize(encoded), encoded);
156 info = rac.GetEncoderStack()->Encode(2, audio, arraysize(encoded), encoded);
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/
H A Daudio_encoder_pcm.cc82 rtc::ArrayView<const int16_t> audio,
88 speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
110 size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio, argument
113 return WebRtcG711_EncodeA(audio, input_len, encoded);
123 size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio, argument
126 return WebRtcG711_EncodeU(audio, input_len, encoded);
80 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/
H A Daudio_encoder_ilbc.cc94 rtc::ArrayView<const int16_t> audio,
104 RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size());
105 std::copy(audio.cbegin(), audio.cend(),
92 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/
H A Daudio_encoder_isac_t_impl.h118 rtc::ArrayView<const int16_t> audio,
130 int r = T::Encode(isac_state_, audio.data(), encoded);
116 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
/external/webrtc/webrtc/modules/audio_processing/agc/
H A Dagc.cc42 float Agc::AnalyzePreproc(const int16_t* audio, size_t length) { argument
46 if (audio[i] == 32767 || audio[i] == -32768)
52 int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) { argument
53 vad_.ProcessChunk(audio, length, sample_rate_hz);
/external/webrtc/webrtc/modules/rtp_rtcp/include/
H A Drtp_rtcp.h37 * audio - True for a audio version of the RTP/RTCP module
59 bool audio; member in struct:webrtc::RtpRtcp::Configuration
286 * Used by the codec module to deliver a video or audio frame for
556 * set audio packet size, used to determine when it's time to send a DTMF
592 * Store the audio level in dBov for header-extension-for-audio-level-
/external/webrtc/webrtc/modules/utility/source/
H A Dcoder.cc77 int32_t AudioCoder::Encode(const AudioFrame& audio, argument
81 // Fake a timestamp in case audio doesn't contain a correct timestamp.
82 // Make a local copy of the audio frame since audio is const
84 audioFrame.CopyFrom(audio);
/external/vboot_reference/firmware/lib/
H A Dvboot_audio.c62 static void VbGetDevMusicNotes(VbAudioContext *audio, int use_short) argument
85 if (!audio->background_beep)
192 audio->music_notes = notebuf;
193 audio->note_count = count;
194 audio->free_notes_when_done = 1;
200 audio->music_notes = builtin;
201 audio->note_count = count;
202 audio->free_notes_when_done = 0;
212 VbAudioContext *audio = &au; local
218 /* Calibrate audio dela
256 VbAudioLooping(VbAudioContext *audio) argument
295 VbAudioClose(VbAudioContext *audio) argument
[all...]
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/
H A Daudio_encoder_g722.cc96 rtc::ArrayView<const int16_t> audio,
108 encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
94 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument
/external/webrtc/webrtc/modules/audio_coding/codecs/red/
H A Daudio_encoder_copy_red.cc57 rtc::ArrayView<const int16_t> audio,
61 speech_encoder_->Encode(rtp_timestamp, audio, max_encoded_bytes, encoded);
55 EncodeInternal( uint32_t rtp_timestamp, rtc::ArrayView<const int16_t> audio, size_t max_encoded_bytes, uint8_t* encoded) argument

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