| /external/webrtc/webrtc/audio/ |
| audio_sink.h | 31 size_t samples_per_channel,
36 samples_per_channel(samples_per_channel),
42 size_t samples_per_channel; // Number of frames in the buffer. member in struct:webrtc::AudioSinkInterface::Data
|
| /external/webrtc/webrtc/modules/utility/include/ |
| audio_frame_operations.h | 29 static void MonoToStereo(const int16_t* src_audio, size_t samples_per_channel,
38 static void StereoToMono(const int16_t* src_audio, size_t samples_per_channel,
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_external_decoder_test.cc | 50 size_t samples_per_channel; local
55 &samples_per_channel,
60 samples_per_channel);
62 return samples_per_channel;
|
| neteq_performance_test.cc | 113 size_t samples_per_channel; local
114 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
119 assert(samples_per_channel == static_cast<size_t>(kSampRateHz * 10 / 1000));
|
| neteq_rtpplay.cc | 613 size_t samples_per_channel; local
614 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
622 1000 * samples_per_channel / kOutputBlockSizeMs);
627 size_t write_len = samples_per_channel * num_channels;
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
| audio_encoder_g722.cc | 49 const size_t samples_per_channel = local
52 encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
53 encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
118 const size_t samples_per_channel = SamplesPerChannel(); local
122 samples_per_channel, encoders_[i].encoded_buffer.data());
123 RTC_CHECK_EQ(encoded, samples_per_channel / 2);
129 for (size_t i = 0; i < samples_per_channel / 2; ++i) {
140 info.encoded_bytes = samples_per_channel / 2 * num_channels_;
|
| /external/webrtc/webrtc/voice_engine/ |
| utility.h | 39 // |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
42 size_t samples_per_channel,
|
| utility.cc | 36 size_t samples_per_channel,
47 AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
62 const size_t src_length = samples_per_channel * audio_ptr_num_channels;
|
| transmit_mixer_unittest.cc | 23 int16_t audio[], size_t samples_per_channel,
|
| /external/webrtc/webrtc/modules/utility/source/ |
| audio_frame_operations.cc | 17 size_t samples_per_channel,
19 for (size_t i = 0; i < samples_per_channel; i++) {
44 size_t samples_per_channel,
46 for (size_t i = 0; i < samples_per_channel; i++) {
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| neteq_impl_unittest.cc | 468 size_t samples_per_channel; local
474 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
475 ASSERT_EQ(kMaxOutputSize, samples_per_channel);
488 EXPECT_EQ(rtp_header.header.timestamp + output[samples_per_channel - 1],
500 EXPECT_EQ(kPayloadLengthSamples - output[samples_per_channel - 1],
547 size_t samples_per_channel; local
553 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
554 ASSERT_EQ(kMaxOutputSize, samples_per_channel);
584 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
585 ASSERT_EQ(kMaxOutputSize, samples_per_channel);
624 size_t samples_per_channel; local
736 size_t samples_per_channel; local
876 size_t samples_per_channel; local
982 size_t samples_per_channel; local
1079 size_t samples_per_channel; local
1200 size_t samples_per_channel; local
[all...] |
| neteq_unittest.cc | 952 size_t samples_per_channel; local
986 size_t samples_per_channel; local
1042 size_t samples_per_channel = 0; local
1276 size_t samples_per_channel; local
1352 size_t samples_per_channel; local
1419 size_t samples_per_channel; local
[all...] |
| neteq_external_decoder_unittest.cc | 190 size_t samples_per_channel; variable
196 &samples_per_channel,
201 samples_per_channel);
204 samples_per_channel = GetOutputAudio(kMaxBlockSize, output_, &output_type);
206 for (size_t i = 0; i < samples_per_channel; ++i) {
|
| neteq_stereo_unittest.cc | 216 size_t samples_per_channel; local
220 &samples_per_channel, &num_channels,
223 EXPECT_EQ(output_size_samples_, samples_per_channel);
228 &samples_per_channel, &num_channels,
231 EXPECT_EQ(output_size_samples_, samples_per_channel);
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| test_utils.cc | 78 size_t samples_per_channel,
82 size_t length = num_channels * samples_per_channel;
84 Interleave(data, samples_per_channel, num_channels, buffer.get());
|
| process_test.cc | 167 int samples_per_channel = sample_rate_hz / 100; local
206 samples_per_channel = sample_rate_hz / 100;
618 samples_per_channel = msg.sample_rate() / 100;
623 near_frame.samples_per_channel_ = samples_per_channel;
628 primary_cb.reset(new ChannelBuffer<float>(samples_per_channel,
708 ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
801 const size_t samples_per_channel = output_sample_rate / 100; local
807 apm->num_output_channels() * samples_per_channel,
815 samples_per_channel,
857 far_frame.samples_per_channel_ = samples_per_channel;
[all...] |
| test_utils.h | 81 size_t samples_per_channel,
|
| /external/webrtc/webrtc/common_audio/include/ |
| audio_util.h | 85 // |deinterleaved| buffers (|num_channel| buffers with |samples_per_channel|
89 size_t samples_per_channel,
95 for (size_t j = 0; j < samples_per_channel; ++j) {
104 // (|samples_per_channel| * |num_channels|).
107 size_t samples_per_channel,
113 for (size_t j = 0; j < samples_per_channel; ++j) {
122 // |interleaved| (|samples_per_channel| * |num_channels|).
|
| /external/webrtc/webrtc/modules/audio_processing/agc/ |
| agc_manager_direct.h | 59 size_t samples_per_channel);
|
| agc_manager_direct.cc | 191 size_t samples_per_channel) {
192 size_t length = num_channels * samples_per_channel;
|
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_receiver.cc | 215 size_t samples_per_channel; local
224 &samples_per_channel,
248 samples_per_channel = static_cast<size_t>(samples_per_channel_int);
263 samples_per_channel = static_cast<size_t>(samples_per_channel_int);
270 samples_per_channel * num_channels * sizeof(int16_t));
278 audio_frame->samples_per_channel_ = samples_per_channel;
279 audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/include/ |
| neteq.h | 169 // |samples_per_channel| elements. If more than one channel is written,
174 size_t* samples_per_channel, size_t* num_channels,
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| audio_processing_impl.h | 69 size_t samples_per_channel,
91 size_t samples_per_channel,
|
| /external/webrtc/talk/app/webrtc/ |
| remoteaudiosource.cc | 159 audio.samples_per_channel);
|
| /external/webrtc/webrtc/modules/include/ |
| module_common_types.h | 509 size_t samples_per_channel, int sample_rate_hz,
573 size_t samples_per_channel,
581 samples_per_channel_ = samples_per_channel;
588 const size_t length = samples_per_channel * num_channels;
|
