/external/webrtc/webrtc/modules/audio_coding/neteq/ |
audio_decoder_impl.cc | 70 int sample_rate_hz,
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delay_manager.h | 49 int sample_rate_hz);
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dsp_helper.h | 120 // Downsamples |input| from |sample_rate_hz| to 4 kHz sample rate. The input
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nack.h | 75 void UpdateSampleRate(int sample_rate_hz);
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neteq_impl_unittest.cc | 84 config_.sample_rate_hz = 8000; 119 mock_dtmf_buffer_ = new MockDtmfBuffer(config_.sample_rate_hz); 122 dtmf_buffer_ = new DtmfBuffer(config_.sample_rate_hz); 437 int /* sample_rate_hz */, [all...] |
neteq_stereo_unittest.cc | 64 config.sample_rate_hz = sample_rate_hz_;
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neteq_external_decoder_unittest.cc | 177 config.sample_rate_hz =
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neteq_impl.h | 121 int sample_rate_hz) override;
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neteq_impl.cc | 102 int fs = config.sample_rate_hz; 208 int sample_rate_hz) { 219 rtp_payload_type, codec, codec_name, sample_rate_hz, decoder); [all...] |
/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
nonlinear_beamformer.cc | 201 void NonlinearBeamformer::Initialize(int chunk_size_ms, int sample_rate_hz) { 203 static_cast<size_t>(sample_rate_hz / (1000.f / chunk_size_ms)); 204 sample_rate_hz_ = sample_rate_hz; 208 hold_target_blocks_ = kHoldTargetSeconds * 2 * sample_rate_hz / kFftSize;
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nonlinear_beamformer.h | 47 void Initialize(int chunk_size_ms, int sample_rate_hz) override;
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/external/webrtc/webrtc/modules/audio_processing/intelligibility/test/ |
intelligibility_proc.cc | 111 config.sample_rate_hz = FLAGS_sample_rate;
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/external/webrtc/webrtc/modules/audio_processing/include/ |
audio_processing.h | 498 // sample_rate_hz: The sampling rate of the stream. 510 StreamConfig(int sample_rate_hz = 0, 513 : sample_rate_hz_(sample_rate_hz), 516 num_frames_(calculate_frames(sample_rate_hz)) {} 525 int sample_rate_hz() const { return sample_rate_hz_; } function in class:webrtc::StreamConfig 544 static size_t calculate_frames(int sample_rate_hz) { 546 AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000); [all...] |
/external/webrtc/webrtc/modules/audio_processing/agc/ |
agc_manager_direct.cc | 234 int sample_rate_hz) { 246 if (agc_->Process(audio, length, sample_rate_hz) != 0) {
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
audio_coding_module_impl.h | 126 int sample_rate_hz,
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audio_coding_module_unittest_oldapi.cc | 893 int sample_rate_hz; member in struct:webrtc::AcmReceiverBitExactnessOldApi::ExternalDecoder [all...] |
audio_coding_module_impl.cc | 625 int sample_rate_hz, 643 sample_rate_hz, external_decoder, name);
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/external/webrtc/webrtc/modules/include/ |
module_common_types.h | 509 size_t samples_per_channel, int sample_rate_hz, 574 int sample_rate_hz, 582 sample_rate_hz_ = sample_rate_hz;
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/external/webrtc/webrtc/voice_engine/ |
transmit_mixer.cc | [all...] |
/external/webrtc/webrtc/modules/audio_processing/ |
audio_processing_impl.h | 92 int sample_rate_hz,
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/include/ |
isac.h | 528 uint16_t sample_rate_hz); 716 void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz);
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/external/webrtc/webrtc/modules/audio_processing/test/ |
audio_processing_unittest.cc | 245 std::string ResourceFilePath(std::string name, int sample_rate_hz) { 248 ss << name << sample_rate_hz / 1000 << "_stereo"; local 357 void Init(int sample_rate_hz, 482 void ApmTest::Init(int sample_rate_hz, 489 SetContainerFormat(sample_rate_hz, num_input_channels, frame_, &float_cb_); 500 std::string filename = ResourceFilePath("far", sample_rate_hz); 508 filename = ResourceFilePath("near", sample_rate_hz); 518 "out", sample_rate_hz, output_sample_rate_hz, reverse_sample_rate_hz, [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
neteq_quality_test.cc | 248 config.sample_rate_hz = out_sampling_khz_ * 1000;
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/external/webrtc/talk/media/webrtc/ |
fakewebrtcvoiceengine.h | 104 int sample_rate_hz,
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/external/webrtc/webrtc/modules/audio_coding/include/ |
audio_coding_module.h | 480 int sample_rate_hz,
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