Searched defs:audio (Results 1 - 25 of 75) sorted by relevance

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/external/autotest/client/site_tests/audio_WebRtcAudioLoopback/
H A Daudio_loopback_test.js31 var audio = document.createElement('audio'); variable
32 audio.autoplay = false;
33 newCell.appendChild(audio);
35 return audio;
50 .getUserMedia({audio: true, video: true})
133 // Finished, pause the audio.
/external/tensorflow/tensorflow/examples/wav_to_spectrogram/
H A Dwav_to_spectrogram_test.cc29 float audio[8] = {-1.0f, 0.0f, 1.0f, 0.0f, -1.0f, 0.0f, 1.0f, 0.0f}; local
32 tensorflow::wav::EncodeAudioAsS16LEWav(audio, 44100, 1, 8, &wav_string));
/external/tensorflow/tensorflow/core/kernels/
H A Dencode_wav_op.cc29 // Encode a tensor as audio samples into the contents of a WAV format file.
35 const Tensor& audio = context->input(0); variable
36 OP_REQUIRES(context, audio.dims() == 2,
37 errors::InvalidArgument("audio must be 2-dimensional",
38 audio.shape().DebugString()));
47 FastBoundsCheck(audio.NumElements(), std::numeric_limits<int32>::max()),
49 "Cannot encode audio with >= max int32 elements"));
51 const int32 channel_count = static_cast<int32>(audio.dim_size(1));
52 const int32 sample_count = static_cast<int32>(audio.dim_size(0));
54 // Encode audio t
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H A Ddecode_wav_op_test.cc71 {decode_wav_op.audio, decode_wav_op.sample_rate},
74 const Tensor& audio = outputs[0]; local
77 EXPECT_EQ(2, audio.dims());
78 EXPECT_EQ(1, audio.dim_size(1));
79 EXPECT_EQ(4, audio.dim_size(0));
80 EXPECT_NEAR(0.0f, audio.flat<float>()(0), 1e-4f);
81 EXPECT_NEAR(0.5f, audio.flat<float>()(1), 1e-4f);
82 EXPECT_NEAR(1.0f, audio.flat<float>()(2), 1e-4f);
83 EXPECT_NEAR(-1.0f, audio.flat<float>()(3), 1e-4f);
91 // audio shap
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H A Dencode_wav_op_test.cc60 {decode_wav_op.audio, decode_wav_op.sample_rate},
63 const Tensor& audio = outputs[0]; local
66 EXPECT_EQ(2, audio.dims());
67 EXPECT_EQ(2, audio.dim_size(1));
68 EXPECT_EQ(4, audio.dim_size(0));
69 EXPECT_NEAR(0.0f, audio.flat<float>()(0), 1e-4f);
70 EXPECT_NEAR(0.5f, audio.flat<float>()(1), 1e-4f);
71 EXPECT_NEAR(1.0f, audio.flat<float>()(2), 1e-4f);
72 EXPECT_NEAR(-1.0f, audio.flat<float>()(3), 1e-4f);
73 EXPECT_NEAR(0.25f, audio
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/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
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/external/tensorflow/tensorflow/core/lib/wav/
H A Dwav_io_test.cc29 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; local
35 EncodeAudioAsS16LEWav(audio, 44100, 2, 3, nullptr).code());
42 EncodeAudioAsS16LEWav(audio, 0, 2, 3, &result).code());
44 EncodeAudioAsS16LEWav(audio, 44100, 0, 3, &result).code());
46 EncodeAudioAsS16LEWav(audio, 44100, 2, 0, &result).code());
51 EncodeAudioAsS16LEWav(audio, kuint32max_plus_one, 2, 3, &result).code());
55 EncodeAudioAsS16LEWav(audio, 44100, kuint16max_plus_one, 3, &result)
60 EncodeAudioAsS16LEWav(audio, 44100, 2, 1073741813, &result).code());
64 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; local
66 TF_EXPECT_OK(EncodeAudioAsS16LEWav(audio, 4410
75 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f}; local
82 float audio[] = {0.0f, 0.1f, 0.2f, 0.3f, 0.4f, 0.5f}; local
123 float audio[] = {0.0f, 1.0f, 0.0f, -1.0f}; local
152 float audio[] = {0.0f, 1.0f, 0.0f, -1.0f}; local
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/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/tensorflow/tensorflow/contrib/summary/
H A Dsummary.py66 from tensorflow.contrib.summary.summary_ops import audio namespace
/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);
/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
/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);

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