/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
H A D | common.h | 44 samples_per_channel_(samples_per_channel), 52 samples_per_channel_(samples_per_channel), 62 samples_per_channel_(samples_per_channel), 73 memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T)); 92 int samples_per_channel() const { return samples_per_channel_; } 94 int length() const { return samples_per_channel_ * num_channels_; } 100 channels_[i] = &data_[i * samples_per_channel_]; 105 const int samples_per_channel_; member in class:webrtc::ChannelBuffer
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H A D | audio_processing_impl.h | 47 samples_per_channel_(AudioProcessing::kChunkSizeMs * rate_ / 1000) {} 52 samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000; 56 int samples_per_channel() const { return samples_per_channel_; } 60 int samples_per_channel_; member in class:webrtc::AudioRate
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
H A D | audio_frame_operations.cc | 29 if ((frame->samples_per_channel_ * 2) >= AudioFrame::kMaxDataSizeSamples) { 36 sizeof(int16_t) * frame->samples_per_channel_); 37 MonoToStereo(data_copy, frame->samples_per_channel_, frame->data_); 56 StereoToMono(frame->data_, frame->samples_per_channel_, frame->data_); 65 for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) { 74 frame.samples_per_channel_ * frame.num_channels_); 82 for (int i = 0; i < frame.samples_per_channel_; i++) { 95 for (int i = 0; i < frame.samples_per_channel_ * frame.num_channels_;
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H A D | audio_frame_operations_unittest.cc | 23 frame_.samples_per_channel_ = 320; 31 for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) { 38 for (int i = 0; i < frame->samples_per_channel_; i++) { 45 EXPECT_EQ(frame1.samples_per_channel_, 46 frame2.samples_per_channel_); 48 for (int i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_; 57 frame_.samples_per_channel_ = AudioFrame::kMaxDataSizeSamples; 70 stereo_frame.samples_per_channel_ = 320; 77 frame_.samples_per_channel_, 95 mono_frame.samples_per_channel_ [all...] |
H A D | file_recorder_impl.cc | 180 tempAudioFrame.samples_per_channel_ = 0; 187 tempAudioFrame.samples_per_channel_ = 188 incomingAudioFrame.samples_per_channel_; 190 i < (incomingAudioFrame.samples_per_channel_); i++) 205 tempAudioFrame.samples_per_channel_ = 206 incomingAudioFrame.samples_per_channel_; 208 i < (incomingAudioFrame.samples_per_channel_); i++) 219 if(tempAudioFrame.samples_per_channel_ != 0) 251 ptrAudioFrame->samples_per_channel_ * 260 ptrAudioFrame->samples_per_channel_, [all...] |
H A D | coder.cc | 89 _encodeTimestamp += audioFrame.samples_per_channel_;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | PCMFile.cc | 135 audio_frame.samples_per_channel_ = samples_10ms_; 147 audio_frame.samples_per_channel_, pcm_file_) != 148 static_cast<size_t>(audio_frame.samples_per_channel_)) { 152 int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_]; 154 for (k = 0; k < audio_frame.samples_per_channel_; k++) { 159 2 * audio_frame.samples_per_channel_, pcm_file_) != 160 static_cast<size_t>(2 * audio_frame.samples_per_channel_)) { 167 audio_frame.num_channels_ * audio_frame.samples_per_channel_, 170 audio_frame.samples_per_channel_)) {
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H A D | initial_delay_unittest.cc | 35 int samples = frame.num_channels_ * frame.samples_per_channel_; 134 in_audio_frame.samples_per_channel_ = codec.plfreq / 100; // 10 ms. 136 in_audio_frame.samples_per_channel_; 147 timestamp += in_audio_frame.samples_per_channel_;
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H A D | SpatialAudio.cc | 162 for (int n = 0; n < audioFrame.samples_per_channel_; n++) { 168 for (int n = 0; n < audioFrame.samples_per_channel_; n++) {
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/external/chromium_org/third_party/webrtc/modules/audio_conference_mixer/source/ |
H A D | audio_frame_manipulator.cc | 45 for(int position = 0; position < audioFrame.samples_per_channel_; 56 assert(rampSize <= audioFrame.samples_per_channel_); 66 assert(rampSize <= audioFrame.samples_per_channel_); 74 (audioFrame.samples_per_channel_ - rampSize) *
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | utility_unittest.cc | 32 src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100; 55 frame->samples_per_channel_ = sample_rate_hz / 100; 56 for (int i = 0; i < frame->samples_per_channel_; i++) { 73 frame->samples_per_channel_ = sample_rate_hz / 100; 74 for (int i = 0; i < frame->samples_per_channel_; i++) { 87 EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_); 102 for (int i = 0; i < ref_frame.samples_per_channel_ * 123 for (int i = 0; i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; 174 src_frame_.samples_per_channel_, [all...] |
H A D | level_indicator.cc | 55 audioFrame.samples_per_channel_*audioFrame.num_channels_);
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H A D | utility.cc | 37 src_frame.samples_per_channel_, 51 const int src_length = src_frame.samples_per_channel_ * 59 dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels; 119 dst_af->samples_per_channel_ = out_length / num_channels;
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/external/webrtc/src/modules/audio_processing/ |
H A D | audio_buffer.cc | 71 samples_per_channel_(samples_per_channel), 89 if (samples_per_channel_ == kSamplesPer32kHzChannel) { 182 return samples_per_channel_; 192 assert(frame->_payloadDataLengthInSamples == samples_per_channel_); 215 for (int j = 0; j < samples_per_channel_; j++) { 224 assert(frame->_payloadDataLengthInSamples == samples_per_channel_); 235 sizeof(int16_t) * samples_per_channel_); 248 for (int j = 0; j < samples_per_channel_; j++) { 266 samples_per_channel_); 280 samples_per_channel_); [all...] |
H A D | audio_buffer.h | 66 const int samples_per_channel_; member in class:webrtc::AudioBuffer
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H A D | audio_processing_impl.cc | 76 samples_per_channel_(sample_rate_hz_ / 100), 158 samples_per_channel_); 160 samples_per_channel_); 194 samples_per_channel_ = rate / 100; 272 if (frame->_payloadDataLengthInSamples != samples_per_channel_) { 403 if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
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H A D | audio_processing_impl.h | 116 int samples_per_channel_; member in class:webrtc::AudioProcessingImpl
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | audio_sink.h | 37 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
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/external/chromium_org/third_party/webrtc/modules/interface/ |
H A D | module_common_types.h | 634 * samples_per_channel_ * num_channels_ 692 int samples_per_channel_; member in class:webrtc::AudioFrame 720 samples_per_channel_ = 0; 737 samples_per_channel_ = samples_per_channel; 760 samples_per_channel_ = src.samples_per_channel_; 768 const int length = samples_per_channel_ * num_channels_; 774 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t)); 781 for (int i = 0; i < samples_per_channel_ * num_channels_; i++) { 803 int offset = samples_per_channel_ * num_channels [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
H A D | external_media_test.cc | 85 EXPECT_LT(0, frame.samples_per_channel_); 103 EXPECT_EQ(f / 100, frame.samples_per_channel_);
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/external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
H A D | test_utils.h | 101 frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs * 112 cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
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H A D | process_test.cc | 126 int num_samples = frame->samples_per_channel_ * frame->num_channels_; 614 far_frame.samples_per_channel_ = reverse_sample_rate / 100; 617 near_frame.samples_per_channel_ = samples_per_channel; 620 far_frame.samples_per_channel_, 652 ASSERT_EQ(sizeof(int16_t) * far_frame.samples_per_channel_ * 672 far_frame.samples_per_channel_, 746 near_frame.samples_per_channel_, 845 far_frame.samples_per_channel_ = samples_per_channel; 848 near_frame.samples_per_channel_ = samples_per_channel; 871 far_frame.samples_per_channel_ [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_send_test.cc | 45 input_frame_.samples_per_channel_ = input_block_size_samples_;
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H A D | acm_send_test_oldapi.cc | 42 input_frame_.samples_per_channel_ = input_block_size_samples_;
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H A D | audio_coding_module_impl.cc | 79 if (length_out_buff < frame.samples_per_channel_) { 82 for (int n = 0; n < frame.samples_per_channel_; ++n) 89 if (length_out_buff < frame.samples_per_channel_) { 92 for (int n = frame.samples_per_channel_ - 1; n >= 0; --n) { 1222 if (audio_frame.samples_per_channel_ <= 0) { 1239 != audio_frame.samples_per_channel_) { 1296 ptr_frame->timestamp_, ptr_audio, ptr_frame->samples_per_channel_, 1307 ptr_frame->timestamp_, ptr_audio, ptr_frame->samples_per_channel_, 1355 expected_in_ts_ += in_frame.samples_per_channel_; 1356 expected_codec_ts_ += in_frame.samples_per_channel_; [all...] |