Searched refs:frame_size_samples_ (Results 1 – 6 of 6) sorted by relevance
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
D | neteq_stereo_unittest.cc | 55 frame_size_samples_(frame_size_ms_ * samples_per_ms_), in NetEqStereoTest() 67 input_ = new int16_t[frame_size_samples_]; in NetEqStereoTest() 68 encoded_ = new uint8_t[2 * frame_size_samples_]; in NetEqStereoTest() 69 input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_]; in NetEqStereoTest() 70 encoded_multi_channel_ = new uint8_t[frame_size_samples_ * 2 * in NetEqStereoTest() 140 if (!input_file_->Read(frame_size_samples_, input_)) { in GetNewPackets() 143 payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_, in GetNewPackets() 145 if (frame_size_samples_ * 2 != payload_size_bytes_) { in GetNewPackets() 149 frame_size_samples_, in GetNewPackets() 151 test::InputAudioFile::DuplicateInterleaved(input_, frame_size_samples_, in GetNewPackets() [all …]
|
D | neteq_external_decoder_unittest.cc | 48 frame_size_samples_(frame_size_ms_ * samples_per_ms_), in NetEqExternalDecoderTest() 58 input_ = new int16_t[frame_size_samples_]; in NetEqExternalDecoderTest() 59 encoded_ = new uint8_t[2 * frame_size_samples_]; in NetEqExternalDecoderTest() 90 if (!input_file_->Read(frame_size_samples_, input_)) { in GetNewPackets() 93 payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_, in GetNewPackets() 95 if (frame_size_samples_ * 2 != payload_size_bytes_) { in GetNewPackets() 99 kPayloadType, frame_size_samples_, &rtp_header_); in GetNewPackets() 206 int frame_size_samples_; member in webrtc::NetEqExternalDecoderTest 241 frame_size_samples_ = frame_size_ms_ * samples_per_ms_; in LargeTimestampJumpTest()
|
/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
D | voice_detection_impl.cc | 49 frame_size_samples_(0) {} in VoiceDetectionImpl() 69 frame_size_samples_); in ProcessCaptureAudio() 143 frame_size_samples_ = frame_size_ms_ * in Initialize()
|
D | voice_detection_impl.h | 60 int frame_size_samples_; variable
|
/external/webrtc/src/modules/audio_processing/ |
D | voice_detection_impl.cc | 54 frame_size_samples_(0) {} in VoiceDetectionImpl() 80 frame_size_samples_); in ProcessCaptureAudio() 154 frame_size_samples_ = frame_size_ms_ * (apm_->split_sample_rate_hz() / 1000); in Initialize()
|
D | voice_detection_impl.h | 59 int frame_size_samples_; variable
|