Searched defs:samples_per_channel (Results 1 - 25 of 25) sorted by relevance

/external/webrtc/webrtc/audio/
H A Daudio_sink.h31 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
30 Data(int16_t* data, size_t samples_per_channel, int sample_rate, size_t channels, uint32_t timestamp) argument
/external/webrtc/webrtc/modules/utility/source/
H A Daudio_frame_operations.cc17 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++) {
16 MonoToStereo(const int16_t* src_audio, size_t samples_per_channel, int16_t* dst_audio) argument
43 StereoToMono(const int16_t* src_audio, size_t samples_per_channel, int16_t* dst_audio) argument
/external/webrtc/webrtc/voice_engine/
H A Dtransmit_mixer_unittest.cc23 int16_t audio[], size_t samples_per_channel,
22 Process(int channel, ProcessingTypes type, int16_t audio[], size_t samples_per_channel, int sample_rate_hz, bool is_stereo) argument
H A Dutility.cc36 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;
35 RemixAndResample(const int16_t* src_data, size_t samples_per_channel, size_t num_channels, int sample_rate_hz, PushResampler<int16_t>* resampler, AudioFrame* dst_frame) argument
H A Dtransmit_mixer.cc1135 size_t samples_per_channel,
1158 RemixAndResample(audio, samples_per_channel, num_channels, sample_rate_hz,
1134 GenerateAudioFrame(const int16_t* audio, size_t samples_per_channel, size_t num_channels, int sample_rate_hz) argument
/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
H A Dneteq_external_decoder_test.cc50 size_t samples_per_channel; local
55 &samples_per_channel,
60 samples_per_channel);
62 return samples_per_channel;
H A Dneteq_performance_test.cc113 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));
H A Dneteq_rtpplay.cc613 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/common_audio/include/
H A Daudio_util.h85 // |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|).
88 Deinterleave(const T* interleaved, size_t samples_per_channel, size_t num_channels, T* const* deinterleaved) argument
106 Interleave(const T* const* deinterleaved, size_t samples_per_channel, size_t num_channels, T* interleaved) argument
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/
H A Daudio_encoder_g722.cc49 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/modules/audio_processing/test/
H A Dtest_utils.cc78 size_t samples_per_channel,
82 size_t length = num_channels * samples_per_channel;
84 Interleave(data, samples_per_channel, num_channels, buffer.get());
77 WriteFloatData(const float* const* data, size_t samples_per_channel, size_t num_channels, WavWriter* wav_file, RawFile* raw_file) argument
H A Dprocess_test.cc167 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...]
H A Daudio_processing_unittest.cc106 size_t samples_per_channel) {
107 for (size_t i = 0; i < samples_per_channel; ++i)
112 size_t samples_per_channel) {
113 for (size_t i = 0; i < samples_per_channel; ++i)
117 void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) { argument
118 for (size_t i = 0; i < samples_per_channel; i++) {
123 void VerifyChannelsAreEqual(int16_t* stereo, size_t samples_per_channel) { argument
124 for (size_t i = 0; i < samples_per_channel; i++) {
1945 const size_t samples_per_channel = static_cast<size_t>( local
1952 ChannelBuffer<int16_t> output_cb(samples_per_channel, num_input_channel
105 MixStereoToMono(const float* stereo, float* mono, size_t samples_per_channel) argument
111 MixStereoToMono(const int16_t* stereo, int16_t* mono, size_t samples_per_channel) argument
[all...]
/external/webrtc/webrtc/modules/audio_coding/neteq/
H A Daudio_decoder_unittest.cc65 size_t samples_per_channel,
69 ASSERT_LE(samples_per_channel * channels, output.size());
70 for (unsigned int n = 0; n < samples_per_channel; ++n)
64 CompareTwoChannels(const std::vector<int16_t>& output, size_t samples_per_channel, size_t channels, int tolerance) argument
H A Dneteq_external_decoder_unittest.cc190 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) {
H A Dneteq_stereo_unittest.cc216 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);
H A Dneteq_impl_unittest.cc468 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...]
H A Dneteq_impl.cc154 size_t* samples_per_channel, size_t* num_channels,
158 int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
168 rtc::checked_cast<int>(*samples_per_channel * 100);
746 size_t* samples_per_channel,
888 *samples_per_channel = output_size_samples_;
891 *samples_per_channel = output_size_samples_;
153 GetAudio(size_t max_length, int16_t* output_audio, size_t* samples_per_channel, size_t* num_channels, NetEqOutputType* type) argument
744 GetAudioInternal(size_t max_length, int16_t* output, size_t* samples_per_channel, size_t* num_channels) argument
H A Dneteq_unittest.cc952 size_t samples_per_channel; local
955 &samples_per_channel, &num_channels, &type));
986 size_t samples_per_channel; local
988 &samples_per_channel,
1042 size_t samples_per_channel = 0; local
1053 samples_per_channel = 0;
1060 &samples_per_channel,
1064 ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
1074 samples_per_channel = 0;
1082 &samples_per_channel,
1276 size_t samples_per_channel; local
1352 size_t samples_per_channel; local
1419 size_t samples_per_channel; local
[all...]
/external/webrtc/webrtc/modules/audio_processing/agc/
H A Dagc_manager_direct.cc191 size_t samples_per_channel) {
192 size_t length = num_channels * samples_per_channel;
189 AnalyzePreProcess(int16_t* audio, int num_channels, size_t samples_per_channel) argument
/external/webrtc/webrtc/modules/audio_coding/acm2/
H A Dacm_receiver.cc215 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);
H A Daudio_coding_module_impl.cc421 int samples_per_channel = resampler_.Resample10Msec( local
426 if (samples_per_channel < 0) {
432 static_cast<size_t>(samples_per_channel);
/external/webrtc/webrtc/modules/audio_processing/
H A Daudio_processing_impl.cc558 size_t samples_per_channel,
583 if (samples_per_channel != input_stream.num_frames()) {
835 size_t samples_per_channel,
843 if (samples_per_channel != reverse_config.num_frames()) {
557 ProcessStream(const float* const* src, size_t samples_per_channel, int input_sample_rate_hz, ChannelLayout input_layout, int output_sample_rate_hz, ChannelLayout output_layout, float* const* dest) argument
834 AnalyzeReverseStream(const float* const* data, size_t samples_per_channel, int rev_sample_rate_hz, ChannelLayout layout) argument
H A Daudio_processing_impl_locking_unittest.cc464 size_t samples_per_channel,
467 for (size_t k = 0; k < samples_per_channel; k++) {
461 PopulateAudioFrame(float** frame, float amplitude, size_t num_channels, size_t samples_per_channel, RandomGenerator* rand_gen) argument
/external/webrtc/webrtc/modules/include/
H A Dmodule_common_types.h509 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;
570 UpdateFrame(int id, uint32_t timestamp, const int16_t* data, size_t samples_per_channel, int sample_rate_hz, SpeechType speech_type, VADActivity vad_activity, size_t num_channels, uint32_t energy) argument

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