/external/webrtc/webrtc/modules/audio_device/include/ |
D | audio_device_defines.h | 87 int sample_rate, in OnDataAvailable() argument 105 int sample_rate, in OnData() argument 118 int sample_rate, in PushCaptureData() argument 127 int sample_rate, in PullRenderData() argument 152 AudioParameters(int sample_rate, size_t channels, size_t frames_per_buffer) in AudioParameters() argument 153 : sample_rate_(sample_rate), in AudioParameters() 156 frames_per_10ms_buffer_(static_cast<size_t>(sample_rate / 100)) {} in AudioParameters() 157 void reset(int sample_rate, size_t channels, size_t frames_per_buffer) { in reset() argument 158 sample_rate_ = sample_rate; in reset() 161 frames_per_10ms_buffer_ = static_cast<size_t>(sample_rate / 100); in reset() [all …]
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/external/autotest/server/site_tests/brillo_PlaybackAudioTest/ |
D | brillo_PlaybackAudioTest.py | 70 def test_playback(self, fb_query, playback_cmd, sample_width, sample_rate, argument 82 sample_rate=sample_rate, 95 def test_audio(self, fb_client, playback_method, sample_rate, sample_width, argument 110 sample_rate=sample_rate, 118 self.host, num_channels, sample_rate, sample_width, 127 sample_rate=sample_rate, 158 for sample_rate in sample_rates_arr: 162 logging.info('Sample rate: %d', sample_rate) 169 sample_rate=sample_rate, 174 failed_params.append((sample_rate, sample_width,
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/external/autotest/server/site_tests/brillo_RecordingAudioTest/ |
D | brillo_RecordingAudioTest.py | 41 sample_rate, num_channels, rec_file): argument 61 (duration_secs, num_channels, sample_rate, sample_width, 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): argument 96 sample_rate=sample_rate, 102 sample_rate=sample_rate, 150 for sample_rate in sample_rates_arr: 156 logging.info('Sample rate: %d', sample_rate) 164 sample_rate=sample_rate, [all …]
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/external/webrtc/webrtc/common_audio/ |
D | wav_header_unittest.cc | 95 int sample_rate = 0; in TEST() local 122 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 143 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 164 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 186 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 209 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 228 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 240 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() 272 int sample_rate = 0; in TEST() local 278 ReadWavHeader(&r, &num_channels, &sample_rate, &format, in TEST() [all …]
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D | wav_header.cc | 63 int sample_rate, in CheckWavParameters() argument 70 if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0) in CheckWavParameters() 72 if (static_cast<uint64_t>(sample_rate) > std::numeric_limits<uint32_t>::max()) in CheckWavParameters() 79 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample > in CheckWavParameters() 138 static inline uint32_t ByteRate(size_t num_channels, int sample_rate, in ByteRate() argument 140 return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample); in ByteRate() 150 int sample_rate, in WriteWavHeader() argument 154 RTC_CHECK(CheckWavParameters(num_channels, sample_rate, format, in WriteWavHeader() 168 WriteLE32(&header.fmt.SampleRate, sample_rate); in WriteWavHeader() 169 WriteLE32(&header.fmt.ByteRate, ByteRate(num_channels, sample_rate, in WriteWavHeader() [all …]
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D | 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_; } in sample_rate() function 81 int sample_rate() const override { return sample_rate_; } in sample_rate() function 104 int sample_rate,
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D | wav_file.cc | 43 s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels() in FormatAsString() 45 << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s"; in FormatAsString() 101 WavWriter::WavWriter(const std::string& filename, int sample_rate, in WavWriter() argument 103 : sample_rate_(sample_rate), in WavWriter() 155 int sample_rate, in rtc_WavOpen() argument 158 new webrtc::WavWriter(filename, sample_rate, num_channels)); in rtc_WavOpen() 172 return reinterpret_cast<const webrtc::WavWriter*>(wf)->sample_rate(); in rtc_WavSampleRate()
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D | wav_header.h | 36 int sample_rate, 47 int sample_rate, 57 int* sample_rate,
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/external/autotest/server/brillo/feedback/ |
D | closed_loop_audio_client.py | 151 sample_rate=_DEFAULT_SAMPLE_RATE, argument 162 self.sample_rate = sample_rate 170 (num_channels, duration_secs, sample_rate, sample_width, 192 sample_rate=self.sample_rate, 227 sample_rate=self.sample_rate, 275 sample_rate=_DEFAULT_SAMPLE_RATE, argument 291 self.sample_rate = sample_rate 304 self.sample_rate, self.sample_width, 326 sample_rate=self.sample_rate,
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/external/autotest/server/brillo/ |
D | audio_utils.py | 67 def check_wav_file(filename, num_channels=None, sample_rate=None, argument 87 if sample_rate is not None and chk_file.getframerate() != sample_rate: 89 sample_rate, chk_file.getframerate()) 106 def generate_sine_file(host, num_channels, sample_rate, sample_width, argument 130 sample_width * _BITS_PER_BYTE, sample_rate, 175 sample_rate): argument 207 1.0 / sample_rate) 208 fft_freqs_rec = numpy.fft.rfftfreq(len(rec_data), 1.0 / sample_rate)
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/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
D | nonlinear_beamformer_test.cc | 46 WavWriter out_file(FLAGS_o, in_file.sample_rate(), 1); in main() 54 bf.Initialize(kChunkSizeMs, in_file.sample_rate()); in main() 57 FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate()); in main() 59 FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate()); in main() 62 rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond), in main() 65 rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond), in main()
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D | covariance_matrix_generator.cc | 68 int sample_rate, in AngledCovarianceMatrix() argument 78 sample_rate, in AngledCovarianceMatrix() 92 int sample_rate, in PhaseAlignmentMasks() argument 101 (static_cast<float>(frequency_bin) / fft_size) * sample_rate; in PhaseAlignmentMasks()
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/external/webrtc/webrtc/audio/ |
D | audio_sink.h | 32 int sample_rate, in Data() 37 sample_rate(sample_rate), in Data() 43 int sample_rate; // Sample rate in Hz. member
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/external/webrtc/webrtc/voice_engine/ |
D | 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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D | voe_base_impl.cc | 114 const int16_t* audio_data, int sample_rate, in OnDataAvailable() argument 123 voe_channels, number_of_voe_channels, audio_data, sample_rate, in OnDataAvailable() 134 PushCaptureData(voe_channels[i], audio_data, 16, sample_rate, in OnDataAvailable() 143 int bits_per_sample, int sample_rate, in OnData() argument 145 PushCaptureData(voe_channel, audio_data, bits_per_sample, sample_rate, in OnData() 150 int bits_per_sample, int sample_rate, in PushCaptureData() argument 159 sample_rate, number_of_frames, number_of_channels); in PushCaptureData() 160 channel_ptr->PrepareEncodeAndSend(sample_rate); in PushCaptureData() 166 int sample_rate, in PullRenderData() argument 172 assert(number_of_frames == static_cast<size_t>(sample_rate / 100)); in PullRenderData() [all …]
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/external/flac/libFLAC/ |
D | format.c | 200 FLAC_API FLAC__bool FLAC__format_sample_rate_is_valid(unsigned sample_rate) in FLAC__format_sample_rate_is_valid() argument 202 if(sample_rate == 0 || sample_rate > FLAC__MAX_SAMPLE_RATE) { in FLAC__format_sample_rate_is_valid() 209 FLAC_API FLAC__bool FLAC__format_blocksize_is_subset(unsigned blocksize, unsigned sample_rate) in FLAC__format_blocksize_is_subset() argument 213 else if(sample_rate <= 48000 && blocksize > 4608) in FLAC__format_blocksize_is_subset() 219 FLAC_API FLAC__bool FLAC__format_sample_rate_is_subset(unsigned sample_rate) in FLAC__format_sample_rate_is_subset() argument 222 !FLAC__format_sample_rate_is_valid(sample_rate) || in FLAC__format_sample_rate_is_subset() 224 sample_rate >= (1u << 16) && in FLAC__format_sample_rate_is_subset() 225 !(sample_rate % 1000 == 0 || sample_rate % 10 == 0) in FLAC__format_sample_rate_is_subset()
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D | stream_encoder_framing.c | 84 FLAC__ASSERT(FLAC__format_sample_rate_is_valid(metadata->data.stream_info.sample_rate)); in FLAC__add_metadata_block() 85 …if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.sample_rate, FLAC__STREAM_META… in FLAC__add_metadata_block() 259 FLAC__ASSERT(FLAC__format_sample_rate_is_valid(header->sample_rate)); in FLAC__frame_add_header() 261 switch(header->sample_rate) { in FLAC__frame_add_header() 274 if(header->sample_rate <= 255000 && header->sample_rate % 1000 == 0) in FLAC__frame_add_header() 276 else if(header->sample_rate % 10 == 0) in FLAC__frame_add_header() 278 else if(header->sample_rate <= 0xffff) in FLAC__frame_add_header() 340 if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate / 1000, 8)) in FLAC__frame_add_header() 344 if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate, 16)) in FLAC__frame_add_header() 348 if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate / 10, 16)) in FLAC__frame_add_header()
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/external/webrtc/webrtc/modules/audio_device/android/ |
D | audio_manager.cc | 177 jint sample_rate, in CacheAudioParameters() argument 189 env, sample_rate, channels, hardware_aec, hardware_agc, hardware_ns, in CacheAudioParameters() 194 jint sample_rate, in OnCacheAudioParameters() argument 207 ALOGD("sample_rate: %d", sample_rate); in OnCacheAudioParameters() 217 playout_parameters_.reset(sample_rate, static_cast<size_t>(channels), in OnCacheAudioParameters() 219 record_parameters_.reset(sample_rate, static_cast<size_t>(channels), in OnCacheAudioParameters()
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D | audio_manager_unittest.cc | 84 PRINT("%ssample rate: %d Hz\n", kTag, playout_parameters_.sample_rate()); in TEST_F() 91 PRINT("%ssample rate: %d Hz\n", kTag, record_parameters_.sample_rate()); in TEST_F() 121 EXPECT_EQ(0, params.sample_rate()); in TEST_F() 141 EXPECT_EQ(kSampleRate, params.sample_rate()); in TEST_F()
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D | audio_device_unittest.cc | 105 size_t num_callbacks, const std::string& file_name, int sample_rate) in FileAudioStream() argument 107 sample_rate_(sample_rate), in FileAudioStream() 110 sample_rate_ = sample_rate; in FileAudioStream() 519 return playout_parameters_.sample_rate(); in playout_sample_rate() 522 return record_parameters_.sample_rate(); in record_sample_rate() 570 std::string GetFileName(int sample_rate) { in GetFileName() argument 571 EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100); in GetFileName() 576 sample_rate / 1000); in GetFileName() 585 static_cast<int>(bytes / (sample_rate * kBytesPerSample)); in GetFileName() 755 EXPECT_EQ(playout_parameters_.sample_rate(), in TEST_F() [all …]
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/external/webrtc/webrtc/modules/audio_processing/logging/ |
D | aec_logging_file_handling.cc | 25 int sample_rate, in WebRtcAec_ReopenWav() argument 28 if (rtc_WavSampleRate(*wav_file) == sample_rate) in WebRtcAec_ReopenWav() 41 *wav_file = rtc_WavOpen(filename, sample_rate, 1); in WebRtcAec_ReopenWav()
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D | aec_logging.h | 28 sample_rate, wav_file) \ argument 30 WebRtcAec_ReopenWav(name, instance_index, process_rate, sample_rate, \ 78 sample_rate) \ argument
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | neteq_stereo_unittest.cc | 29 int sample_rate; member 51 sample_rate_hz_(GetParam().sample_rate), in NetEqStereoTest() 383 int sample_rate = sample_rates[rate_index]; in GetTestParameters() local 388 p.sample_rate = sample_rate; in GetTestParameters() 391 if (sample_rate == 8000) { in GetTestParameters() 405 ", sample_rate = " << p.sample_rate << "}"; in PrintTo()
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/external/webrtc/webrtc/modules/audio_processing/test/ |
D | audio_file_processor.cc | 32 return StreamConfig(file.sample_rate(), file.num_channels()); in GetStreamConfig() 38 CheckedDivExact(file.sample_rate(), AudioFileProcessor::kChunksPerSecond), in GetChannelBuffer() 116 CheckedDivExact(msg.sample_rate(), kChunksPerSecond), in HandleMessage() 120 : msg.sample_rate(); in HandleMessage() 124 input_config_ = StreamConfig(msg.sample_rate(), msg.num_input_channels()); in HandleMessage()
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/external/webrtc/webrtc/modules/audio_device/ios/ |
D | audio_device_unittest_ios.cc | 109 int sample_rate) in FileAudioStream() argument 110 : file_size_in_bytes_(0), sample_rate_(sample_rate), file_pos_(0) { in FileAudioStream() 112 sample_rate_ = sample_rate; in FileAudioStream() 514 int playout_sample_rate() const { return playout_parameters_.sample_rate(); } in playout_sample_rate() 515 int record_sample_rate() const { return record_parameters_.sample_rate(); } in record_sample_rate() 545 std::string GetFileName(int sample_rate) { in GetFileName() argument 546 EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100 || in GetFileName() 547 sample_rate == 16000); in GetFileName() 550 sample_rate / 1000); in GetFileName() 559 static_cast<int>(bytes / (sample_rate * kBytesPerSample)); in GetFileName()
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