/external/chromium_org/third_party/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); 87 const int sample_rate_hz_; member in class:webrtc::TimeStretch 88 const int fs_mult_; // Sample rate multiplier = sample_rate_hz_ / 8000.
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neteq_external_decoder_unittest.cc | 43 : sample_rate_hz_(kSampleRateHz), 44 samples_per_ms_(sample_rate_hz_ / 1000), 54 config.sample_rate_hz = sample_rate_hz_; 75 assert(sample_rate_hz_ == 32000); 186 const int sample_rate_hz_; member in class:webrtc::NetEqExternalDecoderTest
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neteq_stereo_unittest.cc | 52 sample_rate_hz_(GetParam().sample_rate), 53 samples_per_ms_(sample_rate_hz_ / 1000), 64 config.sample_rate_hz = sample_rate_hz_; 91 switch (sample_rate_hz_) { 242 const int sample_rate_hz_; member in class:webrtc::NetEqStereoTest
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time_stretch.cc | 57 sample_rate_hz_, true /* compensate delay*/,
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/external/chromium_org/third_party/webrtc/voice_engine/ |
utility_unittest.cc | 31 src_frame_.sample_rate_hz_ = 16000; 32 src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100; 54 frame->sample_rate_hz_ = sample_rate_hz; 63 SetMonoFrame(frame, data, frame->sample_rate_hz_); 72 frame->sample_rate_hz_ = sample_rate_hz; 82 SetStereoFrame(frame, left, right, frame->sample_rate_hz_); 88 EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_); 176 src_frame_.sample_rate_hz_, 178 dst_frame_.sample_rate_hz_, [all...] |
utility.cc | 43 if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_, 44 dst_frame->sample_rate_hz_, 46 LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_, 47 dst_frame->sample_rate_hz_, audio_ptr_num_channels); 120 dst_af->sample_rate_hz_ = destination_rate;
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output_mixer.cc | 504 frame->sample_rate_hz_ = sample_rate_hz; 514 if (_audioFrame.sample_rate_hz_ != _mixingFrequencyHz) 518 "mixing frequency = %d", _audioFrame.sample_rate_hz_); 519 _mixingFrequencyHz = _audioFrame.sample_rate_hz_; 561 _audioFrame.sample_rate_hz_, 582 frame.sample_rate_hz_ = _audioProcessingModulePtr->input_sample_rate_hz(); 596 if (sampleRate != _audioFrame.sample_rate_hz_) 600 (uint16_t)(_audioFrame.sample_rate_hz_));
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transmit_mixer.cc | 347 _audioFrame.sample_rate_hz_, 384 MixOrReplaceAudioWithFile(_audioFrame.sample_rate_hz_); 395 RecordAudioToFile(_audioFrame.sample_rate_hz_); 404 _audioFrame.sample_rate_hz_, 428 channelPtr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_); 443 channel_ptr->PrepareEncodeAndSend(_audioFrame.sample_rate_hz_); [all...] |
voe_external_media_impl.cc | 181 frame->sample_rate_hz_ = desired_sample_rate_hz == 0 ? -1 :
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/external/webrtc/src/modules/audio_processing/ |
audio_processing_impl.cc | 74 sample_rate_hz_(kSampleRate16kHz), 76 samples_per_channel_(sample_rate_hz_ / 100), 193 sample_rate_hz_ = rate; 196 if (sample_rate_hz_ == kSampleRate32kHz) { 199 split_sample_rate_hz_ = sample_rate_hz_; 206 return sample_rate_hz_; 264 if (frame->_frequencyInHz != sample_rate_hz_) { 395 if (frame->_frequencyInHz != sample_rate_hz_) { 425 if (sample_rate_hz_ == kSampleRate32kHz) { 593 return (stream_data_changed && sample_rate_hz_ == kSampleRate32kHz) [all...] |
audio_processing_impl.h | 114 int sample_rate_hz_; member in class:webrtc::AudioProcessingImpl
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
PacketLossTest.h | 56 int sample_rate_hz_; member in class:webrtc::PacketLossTest
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PacketLossTest.cc | 117 sample_rate_hz_(32000), 143 sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
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PCMFile.cc | 136 audio_frame.sample_rate_hz_ = frequency_;
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initial_delay_unittest.cc | 132 in_audio_frame.sample_rate_hz_ = codec.plfreq;
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target_delay_unittest.cc | 156 ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
external_media_test.cc | 84 EXPECT_LT(0, frame.sample_rate_hz_); 102 EXPECT_EQ(f, frame.sample_rate_hz_);
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/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
audio_processing_impl.cc | 407 if (frame->sample_rate_hz_ != kSampleRate8kHz && 408 frame->sample_rate_hz_ != kSampleRate16kHz && 409 frame->sample_rate_hz_ != kSampleRate32kHz) { 413 frame->sample_rate_hz_ > kSampleRate16kHz) { 420 RETURN_ON_ERR(MaybeInitializeLocked(frame->sample_rate_hz_, 421 frame->sample_rate_hz_, 557 if (frame->sample_rate_hz_ != kSampleRate8kHz && 558 frame->sample_rate_hz_ != kSampleRate16kHz && 559 frame->sample_rate_hz_ != kSampleRate32kHz) { 563 if (frame->sample_rate_hz_ != fwd_in_format_.rate()) [all...] |
/external/chromium_org/third_party/webrtc/modules/utility/source/ |
file_recorder_impl.cc | 186 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; 204 tempAudioFrame.sample_rate_hz_ = incomingAudioFrame.sample_rate_hz_; 247 _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_, 256 _audioResampler.ResetIfNeeded(ptrAudioFrame->sample_rate_hz_, 274 uint16_t(ptrAudioFrame->sample_rate_hz_ / 1000);
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/external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
test_utils.h | 38 frame->sample_rate_hz_ = sample_rate_hz;
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process_test.cc | 605 far_frame.sample_rate_hz_ = msg.sample_rate(); 608 near_frame.sample_rate_hz_ = msg.sample_rate(); 656 far_frame.sample_rate_hz_, 728 near_frame.sample_rate_hz_, 820 far_frame.sample_rate_hz_ = sample_rate_hz; 823 near_frame.sample_rate_hz_ = sample_rate_hz; 846 far_frame.sample_rate_hz_ = sample_rate_hz; 849 near_frame.sample_rate_hz_ = sample_rate_hz; [all...] |
/external/chromium_org/third_party/webrtc/modules/interface/ |
module_common_types.h | 701 int sample_rate_hz_; member in class:webrtc::AudioFrame 729 sample_rate_hz_ = 0; 746 sample_rate_hz_ = sample_rate_hz; 769 sample_rate_hz_ = src.sample_rate_hz_;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
acm_receiver.cc | 468 audio_frame->sample_rate_hz_ = samples_per_channel * 100; 801 frame->sample_rate_hz_ = desired_sample_rate_hz; 803 frame->sample_rate_hz_ = current_sample_rate_hz_; 806 frame->samples_per_channel_ = frame->sample_rate_hz_ / 100; // Always 10 ms.
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audio_coding_module_impl.cc | [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_device/linux/ |
audio_device_pulse_linux.cc | 53 sample_rate_hz_(0), 1205 playSampleSpec.rate = sample_rate_hz_; 1225 _ptrAudioBuffer->SetPlayoutSampleRate(sample_rate_hz_); 1274 _playbackBufferSize = sample_rate_hz_ / 100 * 2 * _playChannels; 1324 recSampleSpec.rate = sample_rate_hz_; 1343 _ptrAudioBuffer->SetRecordingSampleRate(sample_rate_hz_); [all...] |