| /external/webrtc/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 (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
74 frame.samples_per_channel_ * frame.num_channels_);
82 for (size_t i = 0; i < frame.samples_per_channel_; i++) {
95 for (size_t 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 (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
38 for (size_t i = 0; i < frame->samples_per_channel_; i++) {
45 EXPECT_EQ(frame1.samples_per_channel_,
46 frame2.samples_per_channel_);
48 for (size_t 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 | 143 tempAudioFrame.samples_per_channel_ = 0;
150 tempAudioFrame.samples_per_channel_ =
151 incomingAudioFrame.samples_per_channel_;
153 i < (incomingAudioFrame.samples_per_channel_); i++)
168 tempAudioFrame.samples_per_channel_ =
169 incomingAudioFrame.samples_per_channel_;
171 i < (incomingAudioFrame.samples_per_channel_); i++)
182 if(tempAudioFrame.samples_per_channel_ != 0)
211 ptrAudioFrame->samples_per_channel_ *
|
| coder.cc | 86 _encodeTimestamp += static_cast<uint32_t>(audioFrame.samples_per_channel_);
|
| file_player_impl.cc | 130 unresampledAudioFrame.samples_per_channel_ = lengthInBytes >> 1;
170 unresampledAudioFrame.samples_per_channel_,
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| /external/webrtc/webrtc/modules/audio_conference_mixer/source/ |
| audio_frame_manipulator.cc | 45 for(size_t 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/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| audio_sink.h | 37 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
|
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| PCMFile.cc | 135 audio_frame.samples_per_channel_ = samples_10ms_;
150 audio_frame.samples_per_channel_, pcm_file_) !=
151 static_cast<size_t>(audio_frame.samples_per_channel_)) {
155 int16_t* stereo_audio = new int16_t[2 * audio_frame.samples_per_channel_];
156 for (size_t k = 0; k < audio_frame.samples_per_channel_; k++) {
161 2 * audio_frame.samples_per_channel_, pcm_file_) !=
162 static_cast<size_t>(2 * audio_frame.samples_per_channel_)) {
169 audio_frame.num_channels_ * audio_frame.samples_per_channel_,
172 audio_frame.samples_per_channel_)) {
|
| SpatialAudio.cc | 162 for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
168 for (size_t n = 0; n < audioFrame.samples_per_channel_; n++) {
|
| target_delay_unittest.cc | 157 ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_);
|
| /external/webrtc/webrtc/voice_engine/ |
| utility_unittest.cc | 28 src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100;
52 frame->samples_per_channel_ = sample_rate_hz / 100;
53 for (size_t i = 0; i < frame->samples_per_channel_; i++) {
70 frame->samples_per_channel_ = sample_rate_hz / 100;
71 for (size_t i = 0; i < frame->samples_per_channel_; i++) {
84 EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_);
99 for (size_t i = 0; i < ref_frame.samples_per_channel_ *
121 i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
|
| level_indicator.cc | 55 audioFrame.samples_per_channel_*audioFrame.num_channels_);
|
| utility.cc | 27 RemixAndResample(src_frame.data_, src_frame.samples_per_channel_,
71 dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
|
| output_mixer.cc | 549 _audioFrame.samples_per_channel_,
599 for (size_t i = 0; i < _audioFrame.samples_per_channel_; i++)
605 assert(_audioFrame.samples_per_channel_ == toneSamples);
|
| /external/webrtc/webrtc/modules/include/ |
| module_common_types.h | 475 * samples_per_channel_ * num_channels_
533 size_t samples_per_channel_; member in class:webrtc::AudioFrame
561 samples_per_channel_ = 0;
581 samples_per_channel_ = samples_per_channel;
604 samples_per_channel_ = src.samples_per_channel_;
612 const size_t length = samples_per_channel_ * num_channels_;
618 memset(data_, 0, samples_per_channel_ * num_channels_ * sizeof(int16_t));
625 for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
647 size_t offset = samples_per_channel_ * num_channels_
[all...] |
| /external/webrtc/webrtc/tools/agc/ |
| activity_metric.cc | 62 frame->samples_per_channel_;
64 for (size_t n = 0; n < frame->samples_per_channel_; n++)
67 for (size_t n = 0; n < frame->samples_per_channel_; n++)
99 frame.samples_per_channel_ !=
106 frame.data_, frame.samples_per_channel_, &features);
109 frame.samples_per_channel_);
229 frame.samples_per_channel_ = frame.sample_rate_hz_ / 100;
231 frame.samples_per_channel_;
|
| test_utils.cc | 31 frame->samples_per_channel_ * frame->num_channels_;
|
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| audio_coding_module_impl.cc | 51 if (length_out_buff < frame.samples_per_channel_) {
54 for (size_t n = 0; n < frame.samples_per_channel_; ++n)
61 if (length_out_buff < frame.samples_per_channel_) {
64 for (size_t n = frame.samples_per_channel_; n != 0; --n) {
283 if (audio_frame.samples_per_channel_ == 0) {
299 audio_frame.samples_per_channel_) {
352 input_data->length_per_channel = ptr_frame->samples_per_channel_;
390 expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
391 expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
414 preprocess_frame_.samples_per_channel_ = in_frame.samples_per_channel_
[all...] |
| acm_receiver_unittest_oldapi.cc | 102 frame.samples_per_channel_ = codec.plfreq / 100; // 10 ms.
104 memset(frame.data_, 0, frame.samples_per_channel_ * frame.num_channels_ *
110 timestamp_ += frame.samples_per_channel_;
|
| acm_receive_test_oldapi.cc | 163 EXPECT_EQ(samples_per_block, output_frame.samples_per_channel_);
|
| acm_send_test_oldapi.cc | 44 input_frame_.samples_per_channel_ = input_block_size_samples_;
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
| external_media_test.cc | 86 EXPECT_GT(frame.samples_per_channel_, 0U);
104 EXPECT_EQ(static_cast<size_t>(f / 100), frame.samples_per_channel_);
|
| /external/webrtc/webrtc/modules/audio_processing/agc/ |
| agc_unittest.cc | 61 frame.samples_per_channel_ = frame.sample_rate_hz_ / 100;
62 const size_t length = frame.samples_per_channel_ * frame.num_channels_;
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| process_test.cc | 128 int num_samples = frame->samples_per_channel_ * frame->num_channels_;
620 far_frame.samples_per_channel_ = reverse_sample_rate / 100;
623 near_frame.samples_per_channel_ = samples_per_channel;
626 far_frame.samples_per_channel_,
658 ASSERT_EQ(sizeof(int16_t) * far_frame.samples_per_channel_ *
681 far_frame.samples_per_channel_,
758 near_frame.samples_per_channel_,
857 far_frame.samples_per_channel_ = samples_per_channel;
860 near_frame.samples_per_channel_ = samples_per_channel;
[all...] |
| /external/webrtc/webrtc/modules/audio_conference_mixer/test/ |
| audio_conference_mixer_unittest.cc | 127 participants[i].fake_frame()->samples_per_channel_ = kSampleRateHz / 100;
|
