/external/autotest/client/site_tests/audio_WebRtcAudioLoopback/ |
H A D | audio_loopback_test.js | 31 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.
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/external/tensorflow/tensorflow/examples/wav_to_spectrogram/ |
H A D | wav_to_spectrogram_test.cc | 29 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));
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/external/tensorflow/tensorflow/core/kernels/ |
H A D | encode_wav_op.cc | 29 // 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 [all...] |
H A D | decode_wav_op_test.cc | 71 {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 [all...] |
H A D | encode_wav_op_test.cc | 60 {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 [all...] |
/external/ltp/testcases/kernel/device-drivers/v4l/user_space/ |
H A D | test_VIDIOC_ENUMAUDIO.c | 41 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...] |
/external/tensorflow/tensorflow/core/lib/wav/ |
H A D | wav_io_test.cc | 29 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 [all...] |
/external/webrtc/talk/media/base/ |
H A D | audioframe.h | 41 AudioFrame(int16_t* audio, size_t audio_length, int sample_freq, bool stereo) argument 42 : audio10ms_(audio),
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/external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
H A D | audio_encoder_pcm16b.cc | 19 size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio, argument 22 return WebRtcPcm16b_Encode(audio, input_len, encoded);
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/external/webrtc/webrtc/modules/audio_processing/ |
H A D | level_estimator_impl.cc | 31 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());
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H A D | high_pass_filter_impl.cc | 104 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());
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/external/webrtc/webrtc/modules/audio_processing/vad/ |
H A D | voice_activity_detector.cc | 37 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
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_utility.h | 36 bool audio; member in struct:webrtc::RtpUtility::Payload
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/external/webrtc/webrtc/voice_engine/ |
H A D | transmit_mixer_unittest.cc | 23 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
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/external/webrtc/webrtc/modules/audio_coding/codecs/ |
H A D | audio_encoder.cc | 28 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
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/external/webrtc/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/tensorflow/tensorflow/contrib/summary/ |
H A D | summary.py | 66 from tensorflow.contrib.summary.summary_ops import audio namespace
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/external/webrtc/talk/app/webrtc/ |
H A D | remoteaudiosource.cc | 63 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);
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | audio_coding_module_impl.h | 65 // 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
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/external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
H A D | audio_encoder_pcm.cc | 82 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
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/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | audio_encoder_ilbc.cc | 94 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
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
H A D | audio_encoder_isac_t_impl.h | 118 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
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/external/webrtc/webrtc/modules/audio_processing/agc/ |
H A D | agc.cc | 42 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);
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
H A D | rtp_rtcp.h | 37 * 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-
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/external/webrtc/webrtc/modules/utility/source/ |
H A D | coder.cc | 77 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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