| /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
|
| 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_);
|
| /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);
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
| test_utils.h | 38 frame->sample_rate_hz_ = sample_rate_hz;
|
| 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.
|
| 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...] |
