| /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()
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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()
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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()
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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/site_tests/brillo_PlaybackAudioTest/ |
| D | brillo_PlaybackAudioTest.py | 64 def test_playback(self, fb_query, playback_cmd, sample_width, sample_rate, argument
76 sample_rate=sample_rate,
91 sample_rate=_DEFAULT_SAMPLE_RATE, argument
113 sample_rate=sample_rate,
131 sample_rate, host_filename,
148 sample_rate=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/autotest/server/site_tests/brillo_RecordingAudioTest/ |
| D | brillo_RecordingAudioTest.py | 68 sample_rate, num_channels): argument
103 sample_rate=sample_rate,
109 sample_rate=_DEFAULT_SAMPLE_RATE, argument
133 sample_rate=sample_rate,
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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/autotest/server/brillo/feedback/ |
| D | closed_loop_audio_client.py | 230 sample_rate=_DEFAULT_SAMPLE_RATE, argument
241 self.sample_rate = sample_rate
250 (num_channels, duration_secs, sample_rate, sample_width,
272 sample_rate=self.sample_rate,
307 sample_rate=self.sample_rate,
451 sample_rate=None, num_channels=None, argument
466 sample_rate=sample_rate, sample_width=sample_width)
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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()
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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 | opensles_common.cc | 21 SLDataFormat_PCM CreatePcmConfiguration(int sample_rate) { in CreatePcmConfiguration() argument
29 configuration.samplesPerSec = sample_rate * 1000; in CreatePcmConfiguration()
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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/autotest/site_utils/tester_feedback/ |
| D | audio_query_delegate_impl.py | 183 sample_rate=None, num_channels=None, peak_percent_min=1, argument
200 sample_rate=sample_rate, sample_width=sample_width)
207 if sample_rate is not None:
208 props.append('has sample rate of %d' % sample_rate)
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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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