/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
dtmf_buffer_unittest.cc | 30 static int sample_rate_hz = 8000; member in namespace:webrtc 57 DtmfBuffer* buffer = new DtmfBuffer(sample_rate_hz); 97 DtmfBuffer buffer(sample_rate_hz); 132 DtmfBuffer buffer(sample_rate_hz); 158 DtmfBuffer buffer(sample_rate_hz); 202 DtmfBuffer buffer(sample_rate_hz); 245 DtmfBuffer buffer(sample_rate_hz); 279 DtmfBuffer buffer(sample_rate_hz);
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neteq_impl.cc | 96 int fs = config.sample_rate_hz; 213 const int sample_rate_hz = AudioDecoder::CodecSampleRateHz(codec); local 215 sample_rate_hz, decoder); [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
delay_test.cc | 32 DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz."); 47 int sample_rate_hz; member in struct:webrtc::__anon20462::CodecSettings 141 ++test_cntr_, config.codec.name, config.codec.sample_rate_hz, 153 config.name, &my_codec_param, config.sample_rate_hz, 257 test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
neteq_rtpplay.cc | 168 int sample_rate_hz = 16000; local 170 config.sample_rate_hz = sample_rate_hz; 194 input_frame_size_timestamps = 30 * sample_rate_hz / 1000; 237 packet->time_ms() * sample_rate_hz / 1000); 276 sample_rate_hz = 1000 * samples_per_channel / kOutputBlockSizeMs;
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/external/webrtc/src/modules/audio_processing/ |
audio_processing_impl.cc | 205 int AudioProcessingImpl::sample_rate_hz() const { function in class:webrtc::AudioProcessingImpl
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
neteq.h | 72 : sample_rate_hz(16000), 79 int sample_rate_hz; // Initial vale. Will change with input data. member in struct:webrtc::NetEq::Config
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/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
audio_processing_impl.cc | 311 int AudioProcessingImpl::sample_rate_hz() const { function in class:webrtc::AudioProcessingImpl 517 int sample_rate_hz, 527 sample_rate_hz,
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/external/webrtc/src/modules/audio_processing/test/ |
process_test.cc | 87 printf(" -fs SAMPLE_RATE_HZ\n"); 155 int32_t sample_rate_hz = 16000; local 162 int samples_per_channel = sample_rate_hz / 100; 198 ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz)); 199 samples_per_channel = sample_rate_hz / 100; 202 apm->set_sample_rate_hz(sample_rate_hz)); 379 printf("Sample rate: %d Hz\n", sample_rate_hz); 703 far_frame._frequencyInHz = sample_rate_hz; 706 near_frame._frequencyInHz = sample_rate_hz; 711 fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file)) [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
process_test.cc | 65 printf(" -fs SAMPLE_RATE_HZ\n"); 159 int32_t sample_rate_hz = 16000; local 165 int samples_per_channel = sample_rate_hz / 100; 202 ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz)); 203 samples_per_channel = sample_rate_hz / 100; 448 printf("Sample rate: %d Hz\n", sample_rate_hz); 844 far_frame.sample_rate_hz_ = sample_rate_hz; 847 near_frame.sample_rate_hz_ = sample_rate_hz; 852 fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file)) [all...] |
audio_processing_unittest.cc | 236 std::string ResourceFilePath(std::string name, int sample_rate_hz) { 239 ss << name << sample_rate_hz / 1000 << "_stereo"; local 295 void Init(int sample_rate_hz, 409 void ApmTest::Init(int sample_rate_hz, 416 SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_); 427 std::string filename = ResourceFilePath("far", sample_rate_hz); 435 filename = ResourceFilePath("near", sample_rate_hz); 445 sample_rate_hz, [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
audio_coding_module_impl.cc | 2056 int sample_rate_hz; local 2101 int sample_rate_hz; local [all...] |