/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 | 69 def test_playback(self, fb_query, playback_cmd, sample_width, sample_rate, 77 @param sample_rate: Sample rate to test playback at. 81 sample_rate=sample_rate, 94 def test_audio(self, fb_client, playback_method, sample_rate, sample_width, 101 @param sample_rate: Sample rate to test playback at. 109 sample_rate=sample_rate, 117 self.host, num_channels, sample_rate, sample_width, 126 sample_rate=sample_rate [all...] |
/external/autotest/server/site_tests/brillo_RecordingAudioTest/ |
brillo_RecordingAudioTest.py | 41 sample_rate, num_channels, rec_file): 48 @param sample_rate: Recording sample rate in hertz. 59 '--duration_secs=%d --num_channels=%d --sample_rate=%d ' 61 (duration_secs, num_channels, sample_rate, sample_width, 65 '--duration_secs=%d --num_channels=%d --sample_rate=%d ' 67 (duration_secs, num_channels, sample_rate, rec_file)) 70 (num_channels, duration_secs, sample_rate, sample_width, 77 sample_rate, num_channels, duration_secs): 84 @param sample_rate: Recording sample rate in hertz. 96 sample_rate=sample_rate [all...] |
/hardware/qcom/audio/hal/audio_extn/ |
utils.c | 40 int acdb_dev_id, int sample_rate, 67 app_type_cfg[len++] = sample_rate; 74 __func__, stream_type, app_type, acdb_dev_id, sample_rate, 124 int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE; local 160 sample_rate= CODEC_BACKEND_DEFAULT_SAMPLE_RATE; 162 acdb_dev_id, sample_rate, 172 acdb_dev_id, sample_rate, 190 acdb_dev_id, sample_rate, PCM_CAPTURE, 200 acdb_dev_id, sample_rate, 216 int *sample_rate) 347 int sample_rate; local [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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/system/bt/stack/a2dp/ |
a2dp_vendor_ldac.cc | 491 result->sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_44100; 493 result->sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_48000; 495 result->sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_88200; 497 result->sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_96000; 499 result->sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_176400; 501 result->sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_192000; 519 codec_local_capability_.sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_44100; 522 codec_local_capability_.sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_48000; 525 codec_local_capability_.sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_88200; 528 codec_local_capability_.sample_rate |= BTAV_A2DP_CODEC_SAMPLE_RATE_96000 [all...] |
/external/flac/libFLAC/include/protected/ |
stream_decoder.h | 47 unsigned sample_rate; /* in Hz */ member in struct:FLAC__StreamDecoderProtected
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/external/python/cpython2/Lib/ |
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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/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 | 55 mApplicationSampleRate = config->sample_rate; 74 customConfig.sample_rate = config->sample_rate * mRateMultiplier; 84 config->sample_rate, 89 customConfig.sample_rate,
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