/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | neteq_impl_unittest.cc | 468 size_t samples_per_channel; in TEST_F() local 474 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type)); in TEST_F() 475 ASSERT_EQ(kMaxOutputSize, samples_per_channel); in TEST_F() 488 EXPECT_EQ(rtp_header.header.timestamp + output[samples_per_channel - 1], in TEST_F() 500 EXPECT_EQ(kPayloadLengthSamples - output[samples_per_channel - 1], in TEST_F() 547 size_t samples_per_channel; in TEST_F() local 553 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type)); in TEST_F() 554 ASSERT_EQ(kMaxOutputSize, samples_per_channel); in TEST_F() 584 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type)); in TEST_F() 585 ASSERT_EQ(kMaxOutputSize, samples_per_channel); in TEST_F() [all …]
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D | neteq_unittest.cc | 952 size_t samples_per_channel; in TEST_F() local 955 &samples_per_channel, &num_channels, &type)); in TEST_F() 986 size_t samples_per_channel; in TEST_F() local 988 &samples_per_channel, in TEST_F() 1042 size_t samples_per_channel = 0; in CheckBgn() local 1053 samples_per_channel = 0; in CheckBgn() 1060 &samples_per_channel, in CheckBgn() 1064 ASSERT_EQ(expected_samples_per_channel, samples_per_channel); in CheckBgn() 1074 samples_per_channel = 0; in CheckBgn() 1082 &samples_per_channel, in CheckBgn() [all …]
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D | neteq_stereo_unittest.cc | 216 size_t samples_per_channel; in RunTest() local 220 &samples_per_channel, &num_channels, in RunTest() 223 EXPECT_EQ(output_size_samples_, samples_per_channel); in RunTest() 228 &samples_per_channel, &num_channels, in RunTest() 231 EXPECT_EQ(output_size_samples_, samples_per_channel); in RunTest()
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D | neteq_external_decoder_unittest.cc | 190 size_t samples_per_channel; in GetAndVerifyOutput() local 196 &samples_per_channel, in GetAndVerifyOutput() 201 samples_per_channel); in GetAndVerifyOutput() 204 samples_per_channel = GetOutputAudio(kMaxBlockSize, output_, &output_type); in GetAndVerifyOutput() 206 for (size_t i = 0; i < samples_per_channel; ++i) { in GetAndVerifyOutput()
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D | neteq_impl.h | 109 size_t* samples_per_channel, 222 size_t* samples_per_channel,
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/external/webrtc/webrtc/modules/audio_coding/codecs/g722/ |
D | audio_encoder_g722.cc | 49 const size_t samples_per_channel = in AudioEncoderG722() local 52 encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]); in AudioEncoderG722() 53 encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2); in AudioEncoderG722() 118 const size_t samples_per_channel = SamplesPerChannel(); in EncodeInternal() local 122 samples_per_channel, encoders_[i].encoded_buffer.data()); in EncodeInternal() 123 RTC_CHECK_EQ(encoded, samples_per_channel / 2); in EncodeInternal() 129 for (size_t i = 0; i < samples_per_channel / 2; ++i) { in EncodeInternal() 140 info.encoded_bytes = samples_per_channel / 2 * num_channels_; in EncodeInternal()
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/external/webrtc/webrtc/audio/ |
D | audio_sink.h | 31 size_t samples_per_channel, in Data() 36 samples_per_channel(samples_per_channel), in Data() 42 size_t samples_per_channel; // Number of frames in the buffer. member
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
D | neteq_external_decoder_test.cc | 50 size_t samples_per_channel; in GetOutputAudio() local 55 &samples_per_channel, in GetOutputAudio() 60 samples_per_channel); in GetOutputAudio() 62 return samples_per_channel; in GetOutputAudio()
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D | neteq_performance_test.cc | 113 size_t samples_per_channel; in Run() local 114 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel, in Run() 119 assert(samples_per_channel == static_cast<size_t>(kSampRateHz * 10 / 1000)); in Run()
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D | neteq_rtpplay.cc | 613 size_t samples_per_channel; in main() local 614 int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel, in main() 622 1000 * samples_per_channel / kOutputBlockSizeMs); in main() 627 size_t write_len = samples_per_channel * num_channels; in main()
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/external/webrtc/webrtc/modules/utility/source/ |
D | audio_frame_operations.cc | 17 size_t samples_per_channel, in MonoToStereo() argument 19 for (size_t i = 0; i < samples_per_channel; i++) { in MonoToStereo() 44 size_t samples_per_channel, in StereoToMono() argument 46 for (size_t i = 0; i < samples_per_channel; i++) { in StereoToMono()
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/external/webrtc/webrtc/common_audio/include/ |
D | audio_util.h | 89 size_t samples_per_channel, in Deinterleave() argument 95 for (size_t j = 0; j < samples_per_channel; ++j) { in Deinterleave() 107 size_t samples_per_channel, in Interleave() argument 113 for (size_t j = 0; j < samples_per_channel; ++j) { in Interleave()
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/external/webrtc/webrtc/modules/utility/include/ |
D | 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,
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/external/webrtc/webrtc/voice_engine/ |
D | utility.cc | 36 size_t samples_per_channel, in RemixAndResample() argument 47 AudioFrameOperations::StereoToMono(src_data, samples_per_channel, in RemixAndResample() 62 const size_t src_length = samples_per_channel * audio_ptr_num_channels; in RemixAndResample()
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D | utility.h | 42 size_t samples_per_channel,
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D | transmit_mixer_unittest.cc | 23 int16_t audio[], size_t samples_per_channel, in Process() argument
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/external/webrtc/webrtc/modules/audio_processing/test/ |
D | process_test.cc | 167 int samples_per_channel = sample_rate_hz / 100; in void_main() local 206 samples_per_channel = sample_rate_hz / 100; in void_main() 618 samples_per_channel = msg.sample_rate() / 100; in void_main() 623 near_frame.samples_per_channel_ = samples_per_channel; in void_main() 628 primary_cb.reset(new ChannelBuffer<float>(samples_per_channel, in void_main() 708 ASSERT_EQ(sizeof(int16_t) * samples_per_channel * in void_main() 801 const size_t samples_per_channel = output_sample_rate / 100; in void_main() local 807 apm->num_output_channels() * samples_per_channel, in void_main() 815 samples_per_channel, in void_main() 857 far_frame.samples_per_channel_ = samples_per_channel; in void_main() [all …]
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D | test_utils.cc | 78 size_t samples_per_channel, in WriteFloatData() argument 82 size_t length = num_channels * samples_per_channel; in WriteFloatData() 84 Interleave(data, samples_per_channel, num_channels, buffer.get()); in WriteFloatData()
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D | audio_processing_unittest.cc | 106 size_t samples_per_channel) { in MixStereoToMono() argument 107 for (size_t i = 0; i < samples_per_channel; ++i) in MixStereoToMono() 112 size_t samples_per_channel) { in MixStereoToMono() argument 113 for (size_t i = 0; i < samples_per_channel; ++i) in MixStereoToMono() 117 void CopyLeftToRightChannel(int16_t* stereo, size_t samples_per_channel) { in CopyLeftToRightChannel() argument 118 for (size_t i = 0; i < samples_per_channel; i++) { in CopyLeftToRightChannel() 123 void VerifyChannelsAreEqual(int16_t* stereo, size_t samples_per_channel) { in VerifyChannelsAreEqual() argument 124 for (size_t i = 0; i < samples_per_channel; i++) { in VerifyChannelsAreEqual() 1945 const size_t samples_per_channel = static_cast<size_t>( in TEST_F() local 1952 ChannelBuffer<int16_t> output_cb(samples_per_channel, num_input_channels); in TEST_F() [all …]
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
D | acm_receiver.cc | 215 size_t samples_per_channel; in GetAudio() local 224 &samples_per_channel, in GetAudio() 248 samples_per_channel = static_cast<size_t>(samples_per_channel_int); in GetAudio() 263 samples_per_channel = static_cast<size_t>(samples_per_channel_int); in GetAudio() 270 samples_per_channel * num_channels * sizeof(int16_t)); in GetAudio() 278 audio_frame->samples_per_channel_ = samples_per_channel; in GetAudio() 279 audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100); in GetAudio()
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/external/webrtc/webrtc/modules/audio_processing/agc/ |
D | agc_manager_direct.h | 59 size_t samples_per_channel);
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D | agc_manager_direct.cc | 191 size_t samples_per_channel) { in AnalyzePreProcess() argument 192 size_t length = num_channels * samples_per_channel; in AnalyzePreProcess()
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/external/webrtc/webrtc/modules/include/ |
D | module_common_types.h | 509 size_t samples_per_channel, int sample_rate_hz, 573 size_t samples_per_channel, in UpdateFrame() argument 581 samples_per_channel_ = samples_per_channel; in UpdateFrame() 588 const size_t length = samples_per_channel * num_channels; in UpdateFrame()
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/external/webrtc/webrtc/modules/audio_processing/ |
D | audio_processing_impl.h | 69 size_t samples_per_channel, 91 size_t samples_per_channel,
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/external/webrtc/webrtc/modules/audio_coding/neteq/include/ |
D | neteq.h | 174 size_t* samples_per_channel, size_t* num_channels,
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