/external/webrtc/webrtc/audio/ |
H A D | 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 30 Data(int16_t* data, size_t samples_per_channel, int sample_rate, size_t channels, uint32_t timestamp) argument
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/external/webrtc/webrtc/modules/utility/source/ |
H A D | 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++) { 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
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/external/webrtc/webrtc/voice_engine/ |
H A D | transmit_mixer_unittest.cc | 23 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
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H A D | 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; 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
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H A D | transmit_mixer.cc | 1135 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
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
H A D | 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;
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H A D | 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));
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H A D | 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;
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/external/webrtc/webrtc/common_audio/include/ |
H A D | 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|). 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
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/external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
H A D | 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_;
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/external/webrtc/webrtc/modules/audio_processing/test/ |
H A D | 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()); 77 WriteFloatData(const float* const* data, size_t samples_per_channel, size_t num_channels, WavWriter* wav_file, RawFile* raw_file) argument
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H A D | 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...] |
H A D | audio_processing_unittest.cc | 106 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 D | audio_decoder_unittest.cc | 65 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
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H A D | 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) {
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H A D | 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);
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H A D | 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...] |
H A D | neteq_impl.cc | 154 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
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H A D | neteq_unittest.cc | 952 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 D | agc_manager_direct.cc | 191 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
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | 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);
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H A D | audio_coding_module_impl.cc | 421 int samples_per_channel = resampler_.Resample10Msec( local 426 if (samples_per_channel < 0) { 432 static_cast<size_t>(samples_per_channel);
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/external/webrtc/webrtc/modules/audio_processing/ |
H A D | audio_processing_impl.cc | 558 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
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H A D | audio_processing_impl_locking_unittest.cc | 464 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
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/external/webrtc/webrtc/modules/include/ |
H A D | 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; 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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