/external/webrtc/webrtc/modules/audio_coding/neteq/ |
time_stretch.h | 40 : sample_rate_hz_(sample_rate_hz), 46 assert(sample_rate_hz_ == 8000 || 47 sample_rate_hz_ == 16000 || 48 sample_rate_hz_ == 32000 || 49 sample_rate_hz_ == 48000); 92 const int sample_rate_hz_; 93 const int fs_mult_; // Sample rate multiplier = sample_rate_hz_ / 8000.
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time_stretch_unittest.cc | 64 sample_rate_hz_(32000), 65 block_size_(30 * sample_rate_hz_ / 1000), // 30 ms 79 Accelerate accelerate(sample_rate_hz_, kNumChannels, background_noise_); 104 const int sample_rate_hz_; member in class:webrtc::TimeStretchTest
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neteq_stereo_unittest.cc | 51 sample_rate_hz_(GetParam().sample_rate), 52 samples_per_ms_(sample_rate_hz_ / 1000), 64 config.sample_rate_hz = sample_rate_hz_; 91 switch (sample_rate_hz_) { 243 const int sample_rate_hz_; member in class:webrtc::NetEqStereoTest
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
neteq_external_decoder_test.cc | 24 sample_rate_hz_(CodecSampleRateHz(codec_)), 27 config.sample_rate_hz = sample_rate_hz_; 35 kPayloadType, sample_rate_hz_)); 59 EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000), 61 EXPECT_EQ(sample_rate_hz_, neteq_->last_output_sample_rate_hz());
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neteq_external_decoder_test.h | 56 int sample_rate_hz_; member in class:webrtc::test::NetEqExternalDecoderTest
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/external/webrtc/webrtc/modules/audio_processing/ |
voice_detection_impl.cc | 43 sample_rate_hz_ = sample_rate_hz; 51 static_cast<size_t>(frame_size_ms_ * sample_rate_hz_) / 1000; 67 int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_, 85 Initialize(sample_rate_hz_); 146 Initialize(sample_rate_hz_);
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noise_suppression_impl.h | 46 int sample_rate_hz_ GUARDED_BY(crit_) = 0;
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voice_detection_impl.h | 51 int sample_rate_hz_ GUARDED_BY(crit_) = 0;
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noise_suppression_impl.cc | 58 sample_rate_hz_ = sample_rate_hz; 115 Initialize(channels_, sample_rate_hz_);
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/external/webrtc/webrtc/voice_engine/ |
utility_unittest.cc | 27 src_frame_.sample_rate_hz_ = 16000; 28 src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100; 51 frame->sample_rate_hz_ = sample_rate_hz; 60 SetMonoFrame(frame, data, frame->sample_rate_hz_); 69 frame->sample_rate_hz_ = sample_rate_hz; 79 SetStereoFrame(frame, left, right, frame->sample_rate_hz_); 85 EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
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utility.cc | 28 src_frame.num_channels_, src_frame.sample_rate_hz_, 53 if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_, 56 << sample_rate_hz << ", dst_frame->sample_rate_hz_ = " 57 << dst_frame->sample_rate_hz_
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output_mixer.cc | 481 frame->sample_rate_hz_ = sample_rate_hz; 491 if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz) 495 "mixing frequency = %d", _audioFrame.sample_rate_hz_); 496 _mixingFrequencyHz = _audioFrame.sample_rate_hz_; 527 frame.sample_rate_hz_ = _audioProcessingModulePtr->input_sample_rate_hz(); 550 _audioFrame.sample_rate_hz_, 571 if (sampleRate != _audioFrame.sample_rate_hz_) 575 (uint16_t)(_audioFrame.sample_rate_hz_));
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/external/webrtc/webrtc/modules/utility/source/ |
file_recorder_impl.cc | 149 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; 167 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; 207 _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_,
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/external/webrtc/webrtc/modules/audio_coding/test/ |
PacketLossTest.h | 57 int sample_rate_hz_; member in class:webrtc::PacketLossTest
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PacketLossTest.cc | 117 sample_rate_hz_(32000), 144 sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
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/external/webrtc/webrtc/modules/audio_processing/include/ |
audio_processing.h | 314 // The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_| 358 // The |sample_rate_hz_|, |num_channels_|, and |samples_per_channel_| 359 // members of |frame| must be valid. |sample_rate_hz_| must correspond to 513 : sample_rate_hz_(sample_rate_hz), 519 sample_rate_hz_ = value; 525 int sample_rate_hz() const { return sample_rate_hz_; } 536 return sample_rate_hz_ == other.sample_rate_hz_ && 549 int sample_rate_hz_; member in class:webrtc::StreamConfig [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/cng/ |
audio_encoder_cng_unittest.cc | 40 sample_rate_hz_(8000) { 62 num_audio_samples_10ms_ = static_cast<size_t>(10 * sample_rate_hz_ / 1000); 65 .WillRepeatedly(Return(sample_rate_hz_)); 149 VoiceActivity(_, expected_first_block_size_ms * sample_rate_hz_ / 1000, 150 sample_rate_hz_)).WillOnce(Return(Vad::kPassive)); 154 _, expected_second_block_size_ms * sample_rate_hz_ / 1000, 155 sample_rate_hz_)).WillOnce(Return(Vad::kPassive)); 196 int sample_rate_hz_; member in class:webrtc::AudioEncoderCngTest
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/external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
audio_encoder_pcm.cc | 39 : sample_rate_hz_(sample_rate_hz), 60 return sample_rate_hz_;
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audio_encoder_pcm.h | 60 const int sample_rate_hz_; member in class:webrtc::AudioEncoderPcm
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/external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
intelligibility_enhancer.cc | 78 sample_rate_hz_(config.sample_rate_hz), 133 RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz); 151 RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz); 273 center_freqs_[i] *= 0.5f * sample_rate_hz_ / last_center_freq; 285 (0.5f * sample_rate_hz_))); 287 center_freqs_[max(kOne, i) - 1] * freqs_ / (0.5f * sample_rate_hz_))); 293 (0.5f * sample_rate_hz_))); 296 (0.5f * sample_rate_hz_)));
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/external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
external_media_test.cc | 85 EXPECT_GT(frame.sample_rate_hz_, 0); 103 EXPECT_EQ(f, frame.sample_rate_hz_);
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/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
nonlinear_beamformer.cc | 204 sample_rate_hz_ = sample_rate_hz; 221 float freq_hz = (static_cast<float>(i) / kFftSize) * sample_rate_hz_; 241 low_mean_start_bin_ = Round(kLowMeanStartHz * kFftSize / sample_rate_hz_); 242 low_mean_end_bin_ = Round(kLowMeanEndHz * kFftSize / sample_rate_hz_); 253 sample_rate_hz_ / 2.f); 255 sample_rate_hz_ / 2.f); 256 high_mean_start_bin_ = Round(kHighMeanStartHz * kFftSize / sample_rate_hz_); 257 high_mean_end_bin_ = Round(kHighMeanEndHz * kFftSize / sample_rate_hz_); 303 f_ix, kFftSize, sample_rate_hz_, kSpeedOfSoundMeterSeconds, 346 sample_rate_hz_, [all...] |
/external/webrtc/webrtc/modules/audio_coding/acm2/ |
audio_coding_module_impl.cc | 290 if (audio_frame.sample_rate_hz_ > 48000) { 298 if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) != 366 const bool resample = in_frame.sample_rate_hz_ != enc->SampleRateHz(); 383 static_cast<double>(in_frame.sample_rate_hz_)); 415 preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_; 422 src_ptr_audio, in_frame.sample_rate_hz_, enc->SampleRateHz(), 433 preprocess_frame_.sample_rate_hz_ = enc->SampleRateHz();
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/external/webrtc/webrtc/modules/audio_processing/agc/ |
agc_unittest.cc | 59 frame.sample_rate_hz_ = 16000; 61 frame.samples_per_channel_ = frame.sample_rate_hz_ / 100;
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/external/webrtc/webrtc/modules/audio_coding/codecs/red/ |
audio_encoder_copy_red_unittest.cc | 37 sample_rate_hz_(16000), 38 num_audio_samples_10ms(sample_rate_hz_ / 100), 47 .WillRepeatedly(Return(sample_rate_hz_)); 74 const int sample_rate_hz_; member in class:webrtc::AudioEncoderCopyRedTest
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