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  /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
  /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,
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,
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,
  /external/webrtc/webrtc/modules/audio_processing/logging/
aec_logging_file_handling.h 29 int sample_rate,
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);
  /external/webrtc/webrtc/modules/audio_device/android/
opensles_common.cc 21 SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) {
29 configuration.samplesPerSec = sample_rate * 1000;
  /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),
covariance_matrix_generator.h 36 int sample_rate,
45 int sample_rate,
  /external/flac/libFLAC/include/protected/
stream_decoder.h 47 unsigned sample_rate; /* in Hz */ member in struct:FLAC__StreamDecoderProtected
  /external/webrtc/webrtc/common_audio/resampler/
sinusoidal_linear_chirp_source.h 29 SinusoidalLinearChirpSource(int sample_rate, size_t samples,
  /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
  /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
  /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
  /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
  /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;
  /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,
  /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,
  /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,
  /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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