| /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_;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| audio_sink.h | 37 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
|
| /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_)) {
|
| 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_;
|
| SpatialAudio.cc | 162 for (int n = 0; n < audioFrame.samples_per_channel_; n++) {
168 for (int n = 0; n < audioFrame.samples_per_channel_; n++) {
|
| /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) *
|
| /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_);
|
| 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;
|
| /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
|
| audio_processing_impl.h | 116 int samples_per_channel_; member in class:webrtc::AudioProcessingImpl
|
| 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_) {
|
| /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));
|
| 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_);
|
| /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...] |
