| /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);
|
| neteq_impl.cc | 211 const int sample_rate_hz = AudioDecoder::CodecSampleRateHz(codec); local
213 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::__anon19752::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;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_rtpplay.cc | 178 int sample_rate_hz = 16000; local
180 config.sample_rate_hz = sample_rate_hz;
212 input_frame_size_timestamps = 30 * sample_rate_hz / 1000;
251 rtp->time() * sample_rate_hz / 1000);
283 sample_rate_hz = 1000 * samples_per_channel / kOutputBlockSizeMs;
|
| /external/webrtc/src/modules/audio_processing/ |
| audio_processing_impl.cc | 205 int AudioProcessingImpl::sample_rate_hz() const { function in class:webrtc::AudioProcessingImpl
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/interface/ |
| neteq.h | 72 : sample_rate_hz(16000),
78 int sample_rate_hz; // Initial vale. Will change with input data. member in struct:webrtc::NetEq::Config
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| audio_processing_impl.cc | 315 int AudioProcessingImpl::sample_rate_hz() const { function in class:webrtc::AudioProcessingImpl
518 int sample_rate_hz,
528 sample_rate_hz,
|
| /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");
158 int32_t sample_rate_hz = 16000; local
164 int samples_per_channel = sample_rate_hz / 100;
200 ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz));
201 samples_per_channel = sample_rate_hz / 100;
441 printf("Sample rate: %d Hz\n", sample_rate_hz);
820 far_frame.sample_rate_hz_ = sample_rate_hz;
823 near_frame.sample_rate_hz_ = sample_rate_hz;
828 fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file))
[all...] |
| audio_processing_unittest.cc | 239 std::string ResourceFilePath(std::string name, int sample_rate_hz) {
242 ss << name << sample_rate_hz / 1000 << "_stereo"; local
298 void Init(int sample_rate_hz,
412 void ApmTest::Init(int sample_rate_hz,
419 SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_);
430 std::string filename = ResourceFilePath("far", sample_rate_hz);
438 filename = ResourceFilePath("near", sample_rate_hz);
448 sample_rate_hz,
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
| audio_coding_module_impl.cc | 1619 int sample_rate_hz; local
[all...] |
