/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
common.h | 43 samples_per_channel_(samples_per_channel), 53 memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T)); 63 int samples_per_channel() { return samples_per_channel_; } 65 int length() { return samples_per_channel_ * num_channels_; } 70 int samples_per_channel_;
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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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audio_processing_impl.cc | 426 if (frame->samples_per_channel_ != fwd_in_format_.samples_per_channel()) { 435 frame->samples_per_channel_ * 449 frame->samples_per_channel_ * 573 if (frame->samples_per_channel_ != rev_in_format_.samples_per_channel()) { 582 frame->samples_per_channel_ *
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
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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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_ = 320 [all...] |
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...] |
coder.cc | 89 _encodeTimestamp += audioFrame.samples_per_channel_;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
audio_sink.h | 37 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
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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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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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/ |
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/ |
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...] |
level_indicator.cc | 55 audioFrame.samples_per_channel_*audioFrame.num_channels_);
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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/ |
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...] |
audio_buffer.h | 66 const int samples_per_channel_; member in class:webrtc::AudioBuffer
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audio_processing_impl.h | 116 int samples_per_channel_; member in class:webrtc::AudioProcessingImpl
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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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/external/chromium_org/third_party/webrtc/modules/interface/ |
module_common_types.h | 642 * samples_per_channel_ * num_channels_ 700 int samples_per_channel_; member in class:webrtc::AudioFrame 728 samples_per_channel_ = 0; 745 samples_per_channel_ = samples_per_channel; 768 samples_per_channel_ = src.samples_per_channel_; 776 const int length = samples_per_channel_ * num_channels_; 782 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t)); 789 for (int i = 0; i < samples_per_channel_ * num_channels_; i++) { 811 int offset = samples_per_channel_ * num_channels_ [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
test_utils.h | 39 frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs * 50 cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
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audio_processing_unittest.cc | 129 for (int i = 0; i < frame->samples_per_channel_ * frame->num_channels_; ++i) { 136 for (int i = 0; i < frame->samples_per_channel_ * 2; i += 2) { 143 for (int i = 0; i < frame->samples_per_channel_ * frame->num_channels_; ++i) { 149 if (frame1.samples_per_channel_ != frame2.samples_per_channel_) { 156 frame1.samples_per_channel_ * frame1.num_channels_ * 195 const int length = frame.samples_per_channel_ * frame.num_channels_; 467 size_t frame_size = frame->samples_per_channel_ * 2; 480 frame->samples_per_channel_); 520 frame_->samples_per_channel_, [all...] |
process_test.cc | 125 int num_samples = frame->samples_per_channel_ * frame->num_channels_; 606 far_frame.samples_per_channel_ = samples_per_channel; 609 near_frame.samples_per_channel_ = samples_per_channel; 655 far_frame.samples_per_channel_, 727 near_frame.samples_per_channel_, 821 far_frame.samples_per_channel_ = samples_per_channel; 824 near_frame.samples_per_channel_ = samples_per_channel; 847 far_frame.samples_per_channel_ = samples_per_channel; 850 near_frame.samples_per_channel_ = samples_per_channel; [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
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_coding/main/acm2/ |
audio_coding_module_impl.cc | 78 if (length_out_buff < frame.samples_per_channel_) { 81 for (int n = 0; n < frame.samples_per_channel_; ++n) 88 if (length_out_buff < frame.samples_per_channel_) { 91 for (int n = frame.samples_per_channel_ - 1; n >= 0; --n) { [all...] |