| /external/webrtc/webrtc/modules/audio_coding/neteq/include/ |
| neteq.h | 82 : sample_rate_hz(16000),
94 int sample_rate_hz; // Initial value. Will change with input data. member in struct:webrtc::NetEq::Config
188 // produce samples at the rate |sample_rate_hz|. Returns kOK on success, kFail
196 int sample_rate_hz) = 0;
263 // (Config::sample_rate_hz) is returned.
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_record_jni.cc | 185 const int sample_rate_hz = audio_parameters_.sample_rate(); local
186 ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
187 audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
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| audio_track_jni.cc | 202 const int sample_rate_hz = audio_parameters_.sample_rate(); local
203 ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
204 audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
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| opensles_player.cc | 179 const int sample_rate_hz = audio_parameters_.sample_rate(); local
180 ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
181 audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
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| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| transient_suppression_test.cc | 47 DEFINE_int32(sample_rate_hz,
65 "num_channels and sample_rate_hz, the detection signal from the\n"
67 "signal from the reference_file_name with sample_rate_hz, divides them\n"
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| transient_detector.h | 34 TransientDetector(int sample_rate_hz);
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| transient_suppressor.h | 32 int Initialize(int sample_rate_hz, int detector_rate_hz, int num_channels);
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| transient_suppressor.cc | 68 int TransientSuppressor::Initialize(int sample_rate_hz,
71 switch (sample_rate_hz) {
102 data_length_ = sample_rate_hz * ts::kChunkSizeMs / 1000;
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| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_encoder_pcm.h | 51 AudioEncoderPcm(const Config& config, int sample_rate_hz);
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
| audio_encoder_isac_t.h | 36 int sample_rate_hz = 16000; member in struct:webrtc::final::Config
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| /external/webrtc/webrtc/modules/audio_processing/agc/ |
| agc_manager_direct.h | 60 void Process(const int16_t* audio, size_t length, int sample_rate_hz);
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| /external/webrtc/webrtc/voice_engine/ |
| output_mixer.h | 66 int GetMixedAudio(int sample_rate_hz, size_t num_channels,
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| output_mixer.cc | 465 int OutputMixer::GetMixedAudio(int sample_rate_hz,
470 "OutputMixer::GetMixedAudio(sample_rate_hz=%d, num_channels=%" PRIuS ")",
471 sample_rate_hz, num_channels);
481 frame->sample_rate_hz_ = sample_rate_hz;
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| /external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
| audio_encoder_cng.cc | 23 int sample_rate_hz,
29 WebRtcCng_InitEnc(cng_inst.get(), sample_rate_hz,
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_rtpplay.cc | 460 int sample_rate_hz = CodecSampleRate(packet->header().payloadType); local
461 if (sample_rate_hz <= 0) {
476 new webrtc::test::OutputWavFile(output_file_name, sample_rate_hz));
492 config.sample_rate_hz = sample_rate_hz;
508 input_frame_size_timestamps = 30 * sample_rate_hz / 1000;
562 static_cast<uint32_t>(packet->time_ms() * sample_rate_hz / 1000));
621 sample_rate_hz = rtc::checked_cast<int>(
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| neteq_performance_test.cc | 42 config.sample_rate_hz = kSampRateHz;
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| /external/webrtc/webrtc/modules/audio_processing/ |
| audio_processing_performance_unittest.cc | 78 SimulationConfig(int sample_rate_hz, SettingsType simulation_settings)
79 : sample_rate_hz(sample_rate_hz),
163 int sample_rate_hz = 16000; member in struct:webrtc::__anon27027::SimulationConfig
273 "_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
406 simulation_config_->sample_rate_hz);
410 (simulation_config_->sample_rate_hz *
416 simulation_config_->sample_rate_hz);
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| /external/webrtc/webrtc/voice_engine/test/auto_test/ |
| voe_conference_test.cc | 27 void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) {
30 for (int i = 0; i < sample_rate_hz; ++i) {
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_receiver.h | 48 int sample_rate_hz; member in struct:webrtc::acm2::AcmReceiver::Decoder
97 // - sample_rate_hz : sample rate.
120 int sample_rate_hz,
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| acm_receive_test_oldapi.cc | 146 int sample_rate_hz,
150 sample_rate_hz, num_channels, name);
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| /external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
| intelligibility_enhancer_unittest.cc | 91 config_.sample_rate_hz = kSampleRate;
96 config_.sample_rate_hz = kSampleRate;
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| isac.c | [all...] |
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| audio_decoder_unittest.cc | 322 config.sample_rate_hz = codec_input_rate_hz_;
324 static_cast<int>(frame_size_ / (config.sample_rate_hz / 1000));
372 config.sample_rate_hz = codec_input_rate_hz_;
389 config.sample_rate_hz = codec_input_rate_hz_;
406 config.sample_rate_hz = codec_input_rate_hz_;
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| delay_manager.cc | 74 int sample_rate_hz) {
75 if (sample_rate_hz <= 0) {
99 packet_len_ms = (1000 * packet_len_samp) / sample_rate_hz;
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| /external/webrtc/webrtc/modules/audio_processing/aec/ |
| system_delay_unittest.cc | 27 // Initialization of AEC handle with respect to |sample_rate_hz|. Since the
29 void Init(int sample_rate_hz);
98 void SystemDelayTest::Init(int sample_rate_hz) {
100 EXPECT_EQ(0, WebRtcAec_Init(handle_, sample_rate_hz, 48000));
104 samples_per_frame_ = static_cast<size_t>(sample_rate_hz / 100);
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