/external/webrtc/webrtc/audio/ |
audio_sink.h | 32 int sample_rate, 37 sample_rate(sample_rate), 43 int sample_rate; // Sample rate in Hz. member in struct:webrtc::AudioSinkInterface::Data
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/external/webrtc/webrtc/modules/audio_device/include/ |
audio_device_defines.h | 87 int sample_rate, 105 int sample_rate, 118 int sample_rate, 127 int sample_rate, 152 AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer) 153 : sample_rate_(sample_rate), 156 frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {} 157 void reset(int sample_rate, size_t channels, size_t frames_per_buffer) { 158 sample_rate_ = sample_rate; 161 frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100) 171 int sample_rate() const { return sample_rate_; } function in class:webrtc::AudioParameters [all...] |
/external/webrtc/webrtc/common_audio/ |
wav_header.h | 36 int sample_rate, 47 int sample_rate, 57 int* sample_rate,
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wav_file.h | 29 virtual int sample_rate() const = 0; 42 WavWriter(const std::string& filename, int sample_rate, size_t num_channels); 53 int sample_rate() const override { return sample_rate_; } 81 int sample_rate() const override { return sample_rate_; } 104 int sample_rate,
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wav_header.cc | 63 int sample_rate, 67 // num_channels, sample_rate, and bytes_per_sample must be positive, must fit 70 if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0) 72 if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max()) 79 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample > 138 static inline uint32_t ByteRate(size_t num_channels, int sample_rate, 140 return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample); 150 int sample_rate, 154 RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format, 168 WriteLE32(&header.fmt.SampleRate, sample_rate); [all...] |
wav_header_unittest.cc | 95 int sample_rate = 0; local 122 ReadWavHeader(&r, &num_channels, &sample_rate, &format, 143 ReadWavHeader(&r, &num_channels, &sample_rate, &format, 164 ReadWavHeader(&r, &num_channels, &sample_rate, &format, 186 ReadWavHeader(&r, &num_channels, &sample_rate, &format, 209 ReadWavHeader(&r, &num_channels, &sample_rate, &format, 228 ReadWavHeader(&r, &num_channels, &sample_rate, &format, 240 ReadWavHeader(&r, &num_channels, &sample_rate, &format, 272 int sample_rate = 0; local 278 ReadWavHeader(&r, &num_channels, &sample_rate, &format 308 int sample_rate = 0; local [all...] |
/external/webrtc/talk/media/base/ |
audiorenderer.h | 43 int sample_rate,
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/external/webrtc/webrtc/modules/audio_processing/logging/ |
aec_logging_file_handling.h | 29 int sample_rate,
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aec_logging_file_handling.cc | 25 int sample_rate, 28 if (rtc_WavSampleRate(*wav_file) == sample_rate) 41 *wav_file = rtc_WavOpen(filename, sample_rate, 1);
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/external/webrtc/webrtc/modules/audio_device/android/ |
opensles_common.cc | 21 SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) { 29 configuration.samplesPerSec = sample_rate * 1000;
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/external/autotest/server/site_tests/brillo_PlaybackAudioTest/ |
brillo_PlaybackAudioTest.py | 64 def test_playback(self, fb_query, playback_cmd, sample_width, sample_rate, 72 @param sample_rate: Sample rate to test playback at. 76 sample_rate=sample_rate, 91 sample_rate=_DEFAULT_SAMPLE_RATE, 103 @param sample_rate: Sample rate to test playback at. 113 sample_rate=sample_rate, 131 sample_rate, host_filename, 148 sample_rate=sample_rate [all...] |
/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
nonlinear_beamformer_test.cc | 46 WavWriter out_file(FLAGS_o, in_file.sample_rate(), 1); 54 bf.Initialize(kChunkSizeMs, in_file.sample_rate()); 57 FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate()); 59 FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate()); 62 rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond), 65 rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
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covariance_matrix_generator.h | 36 int sample_rate, 45 int sample_rate,
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/external/flac/libFLAC/include/protected/ |
stream_decoder.h | 47 unsigned sample_rate; /* in Hz */ member in struct:FLAC__StreamDecoderProtected
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/external/webrtc/webrtc/common_audio/resampler/ |
sinusoidal_linear_chirp_source.h | 29 SinusoidalLinearChirpSource(int sample_rate, size_t samples,
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/prebuilts/gdb/darwin-x86/lib/python2.7/ |
sunaudio.py | 26 sample_rate = get_long_be(fp.read(4)) 35 return (data_size, encoding, sample_rate, channels, info) 41 data_size, encoding, sample_rate, channels, info = hdr 47 print 'Sample rate:', sample_rate
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/prebuilts/gdb/linux-x86/lib/python2.7/ |
sunaudio.py | 26 sample_rate = get_long_be(fp.read(4)) 35 return (data_size, encoding, sample_rate, channels, info) 41 data_size, encoding, sample_rate, channels, info = hdr 47 print 'Sample rate:', sample_rate
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/prebuilts/python/darwin-x86/2.7.5/lib/python2.7/ |
sunaudio.py | 26 sample_rate = get_long_be(fp.read(4)) 35 return (data_size, encoding, sample_rate, channels, info) 41 data_size, encoding, sample_rate, channels, info = hdr 47 print 'Sample rate:', sample_rate
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/prebuilts/python/linux-x86/2.7.5/lib/python2.7/ |
sunaudio.py | 26 sample_rate = get_long_be(fp.read(4)) 35 return (data_size, encoding, sample_rate, channels, info) 41 data_size, encoding, sample_rate, channels, info = hdr 47 print 'Sample rate:', sample_rate
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/device/google/dragon/audio/hal/ |
cras_dsp.c | 29 int sample_rate; member in struct:cras_dsp_context 68 if (cras_dsp_pipeline_instantiate(pipeline, ctx->sample_rate) != 0) { 73 if (cras_dsp_pipeline_get_sample_rate(pipeline) != ctx->sample_rate) { 76 ctx->sample_rate); 141 struct cras_dsp_context *cras_dsp_context_new(int sample_rate, 147 ctx->sample_rate = sample_rate;
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/frameworks/av/services/audioflinger/ |
SpdifStreamOut.cpp | 56 mApplicationSampleRate = config->sample_rate; 75 customConfig.sample_rate = config->sample_rate * mRateMultiplier; 85 config->sample_rate, 90 customConfig.sample_rate,
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/hardware/qcom/audio/msm8909/hal/audio_extn/ |
utils.c | 160 uint32_t sample_rate = 48000; local 168 sample_rate = (uint32_t)strtol(str, (char **)NULL, 10); 169 ALOGV("%s: sample_rate - %d", __func__, sample_rate); 170 if (0 != sample_rate) { 175 ss_info->sample_rate = sample_rate; 226 so_info->app_type_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE; 299 app_type_cfg[length++] = so_info->app_type_cfg.sample_rate; 350 __func__, so_info->flags, so_info->app_type_cfg.sample_rate, 491 int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE; local [all...] |
/external/autotest/server/site_tests/brillo_RecordingAudioTest/ |
brillo_RecordingAudioTest.py | 68 sample_rate, num_channels): 75 @param sample_rate: Recording sample rate in hertz. 103 sample_rate=sample_rate, 109 sample_rate=_DEFAULT_SAMPLE_RATE, 120 @param sample_rate: Recording sample rate in hertz. 133 sample_rate=sample_rate,
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/external/webrtc/webrtc/voice_engine/ |
voe_base_impl.h | 78 int sample_rate, 88 int sample_rate, 94 int sample_rate, 98 int sample_rate, 128 const void* audio_data, uint32_t sample_rate, size_t number_of_channels, 132 void GetPlayoutData(int sample_rate, size_t number_of_channels,
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/device/google/dragon/audio/hal/dsp/ |
drc.h | 111 float sample_rate; member in struct:drc 154 struct drc *drc_new(float sample_rate);
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