/external/webrtc/modules/audio_processing/include/ |
D | audio_frame_proxies.cc | 23 StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_, in ProcessAudioFrame() 25 StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_, in ProcessAudioFrame() 49 if (frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate8kHz && in ProcessReverseAudioFrame() 50 frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate16kHz && in ProcessReverseAudioFrame() 51 frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate32kHz && in ProcessReverseAudioFrame() 52 frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate48kHz) { in ProcessReverseAudioFrame() 60 StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_, in ProcessReverseAudioFrame() 62 StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_, in ProcessReverseAudioFrame()
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/external/webrtc/modules/audio_processing/agc2/ |
D | down_sampler.cc | 59 sample_rate_hz_ = sample_rate_hz; in Initialize() 60 down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000); in Initialize() 64 if (sample_rate_hz_ == kSampleRate16kHz) { in Initialize() 66 } else if (sample_rate_hz_ == kSampleRate32kHz) { in Initialize() 68 } else if (sample_rate_hz_ == kSampleRate48kHz) { in Initialize() 75 data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1); in DownSample() 76 RTC_DCHECK_EQ(sample_rate_hz_ * kChunkSizeMs / 1000, in.size()); in DownSample() 82 if (sample_rate_hz_ != kSampleRate8kHz) { in DownSample()
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/external/webrtc/modules/audio_coding/neteq/ |
D | time_stretch.h | 40 : sample_rate_hz_(sample_rate_hz), in TimeStretch() 45 assert(sample_rate_hz_ == 8000 || sample_rate_hz_ == 16000 || in TimeStretch() 46 sample_rate_hz_ == 32000 || sample_rate_hz_ == 48000); in TimeStretch() 89 const int sample_rate_hz_; variable
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D | neteq_decoder_plc_unittest.cc | 41 : input_(std::move(input)), sample_rate_hz_(sample_rate_hz) {} in AudioDecoderPlc() 44 int SampleRateHz() const override { return sample_rate_hz_; } in SampleRateHz() 51 RTC_CHECK_EQ(encoded_len / 2, 20 * sample_rate_hz_ / 1000); in DecodeInternal() 52 RTC_CHECK_EQ(sample_rate_hz, sample_rate_hz_); in DecodeInternal() 67 int dec_len = DecodeInternal(nullptr, 2 * 20 * sample_rate_hz_ / 1000, in GeneratePlc() 68 sample_rate_hz_, decoded.data(), &speech_type); in GeneratePlc() 85 const int sample_rate_hz_; member in webrtc::test::__anonf78052b90111::AudioDecoderPlc
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D | time_stretch_unittest.cc | 63 sample_rate_hz_(32000), in TimeStretchTest() 64 block_size_(30 * sample_rate_hz_ / 1000), // 30 ms in TimeStretchTest() 76 Accelerate accelerate(sample_rate_hz_, kNumChannels, background_noise_); in TestAccelerate() 101 const int sample_rate_hz_; member in webrtc::TimeStretchTest
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D | neteq_stereo_unittest.cc | 56 sample_rate_hz_(GetParam().sample_rate), in NetEqStereoTest() 57 samples_per_ms_(sample_rate_hz_ / 1000), in NetEqStereoTest() 70 config.sample_rate_hz = sample_rate_hz_; in NetEqStereoTest() 95 kPayloadTypeMono, SdpAudioFormat("l16", sample_rate_hz_, 1))); in SetUp() 98 SdpAudioFormat("l16", sample_rate_hz_, num_channels_))); in SetUp() 206 const int sample_rate_hz_; member in webrtc::NetEqStereoTest
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/external/webrtc/modules/audio_mixer/ |
D | audio_mixer_test.cc | 51 sample_rate_hz_(wav_reader_->sample_rate()), in FilePlayingSource() 52 samples_per_channel_(sample_rate_hz_ / 100), in FilePlayingSource() 59 frame->sample_rate_hz_ = target_rate_hz; in GetAudioFrameWithInfo() 61 RTC_CHECK_EQ(target_rate_hz, sample_rate_hz_); in GetAudioFrameWithInfo() 76 int PreferredSampleRate() const override { return sample_rate_hz_; } in PreferredSampleRate() 82 ss << "{rate: " << sample_rate_hz_ << ", channels: " << number_of_channels_ in ToString() 89 int sample_rate_hz_; member in webrtc::test::FilePlayingSource 169 RTC_CHECK_EQ(sample_rate, frame.sample_rate_hz_); in main()
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D | audio_mixer_impl_unittest.cc | 43 frame->sample_rate_hz_ = kDefaultSampleRateHz; in ResetFrame() 95 audio_frame->sample_rate_hz_ = sample_rate_hz; in FakeAudioFrameWithInfo() 155 audio_source->fake_frame()->sample_rate_hz_ = native_sample_rate; in MixMonoAtGivenNativeRate() 261 EXPECT_EQ(frequency, frame_for_mixing.sample_rate_hz_); in TEST() 280 participant.fake_frame()->sample_rate_hz_ = expected_mix_frequency; in TEST() 285 EXPECT_EQ(48000, frame_for_mixing.sample_rate_hz_); in TEST() 503 EXPECT_EQ(kOutputRate, frame_for_mixing.sample_rate_hz_); in TEST() 534 EXPECT_EQ(rate, frame_for_mixing.sample_rate_hz_); in TEST() 623 frame->sample_rate_hz_ = HighOutputRateCalculator::kDefaultFrequency; in TEST() 632 other_frame->sample_rate_hz_ = HighOutputRateCalculator::kDefaultFrequency; in TEST() [all …]
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/external/webrtc/modules/audio_coding/codecs/opus/ |
D | audio_encoder_opus_unittest.cc | 137 int sample_rate_hz_{GetParam()}; member in webrtc::AudioEncoderOpusTest 144 auto states = CreateCodec(sample_rate_hz_, 1); in TEST_P() 150 auto states = CreateCodec(sample_rate_hz_, 2); in TEST_P() 156 auto states = CreateCodec(sample_rate_hz_, 2); in TEST_P() 164 auto states = CreateCodec(sample_rate_hz_, 2); in TEST_P() 186 auto states = CreateCodec(sample_rate_hz_, 2); in TEST_P() 197 auto states = CreateCodec(sample_rate_hz_, 1); in TEST_P() 221 auto states = CreateCodec(sample_rate_hz_, 2); in TEST_P() 239 auto states = CreateCodec(sample_rate_hz_, 2); in TEST_P() 259 auto states = CreateCodec(sample_rate_hz_, 2); in TEST_P() [all …]
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D | audio_decoder_opus.cc | 25 : channels_{num_channels}, sample_rate_hz_{sample_rate_hz} { in AudioDecoderOpusImpl() 29 WebRtcOpus_DecoderCreate(&dec_state_, channels_, sample_rate_hz_); 63 RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_); in DecodeInternal() 84 RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_); in DecodeRedundantInternal() 110 return WebRtcOpus_FecDurationEst(encoded, encoded_len, sample_rate_hz_); in PacketDurationRedundant() 121 return sample_rate_hz_; in SampleRateHz()
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/external/webrtc/modules/audio_coding/codecs/pcm16b/ |
D | audio_decoder_pcm16b.cc | 22 : sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) { in AudioDecoderPcm16B() 32 return sample_rate_hz_; in SampleRateHz() 44 RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz); in DecodeInternal() 53 const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000); in ParsePayload()
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/external/webrtc/audio/ |
D | remix_resample_unittest.cc | 28 src_frame_.sample_rate_hz_ = 16000; in UtilityTest() 29 src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100; in UtilityTest() 52 frame->sample_rate_hz_ = sample_rate_hz; in SetMonoFrame() 62 SetMonoFrame(data, frame->sample_rate_hz_, frame); in SetMonoFrame() 73 frame->sample_rate_hz_ = sample_rate_hz; in SetStereoFrame() 84 SetStereoFrame(left, right, frame->sample_rate_hz_, frame); in SetStereoFrame() 97 frame->sample_rate_hz_ = sample_rate_hz; in SetQuadFrame() 111 EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_); in VerifyParams()
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D | remix_resample.cc | 25 src_frame.num_channels_, src_frame.sample_rate_hz_, in RemixAndResample() 57 if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_, in RemixAndResample() 60 << ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_ in RemixAndResample()
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/external/webrtc/api/audio/ |
D | audio_frame.cc | 33 swap(a.sample_rate_hz_, b.sample_rate_hz_); in swap() 61 sample_rate_hz_ = 0; in ResetWithoutMuting() 80 sample_rate_hz_ = sample_rate_hz; in UpdateFrame() 109 sample_rate_hz_ = src.sample_rate_hz_; in CopyFrom()
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/external/webrtc/modules/audio_coding/codecs/isac/ |
D | audio_decoder_isac_t_impl.h | 25 : sample_rate_hz_(config.sample_rate_hz) { in AudioDecoderIsacT() 30 RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz_)); in AudioDecoderIsacT() 44 RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz); in DecodeInternal() 74 return sample_rate_hz_; in SampleRateHz()
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/external/webrtc/modules/audio_coding/neteq/tools/ |
D | fake_decode_from_file.h | 35 sample_rate_hz_(sample_rate_hz), in FakeDecodeFromFile() 45 int SampleRateHz() const override { return sample_rate_hz_; } in SampleRateHz() 69 const int sample_rate_hz_; variable
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/external/webrtc/modules/audio_processing/aec3/ |
D | echo_canceller3_unittest.cc | 170 : sample_rate_hz_(sample_rate_hz), in EchoCanceller3Tester() 171 num_bands_(NumBandsForRate(sample_rate_hz_)), in EchoCanceller3Tester() 173 fullband_frame_length_(rtc::CheckedDivExact(sample_rate_hz_, 100)), in EchoCanceller3Tester() 192 EchoCanceller3Config(), sample_rate_hz_, 1, 1, in RunCaptureTransportVerificationTest() 217 EchoCanceller3Config(), sample_rate_hz_, 1, 1, in RunRenderTransportVerificationTest() 288 EchoCanceller3 aec3(EchoCanceller3Config(), sample_rate_hz_, 1, 1, in RunEchoPathChangeVerificationTest() 368 EchoCanceller3 aec3(EchoCanceller3Config(), sample_rate_hz_, 1, 1, in RunEchoLeakageVerificationTest() 454 EchoCanceller3 aec3(EchoCanceller3Config(), sample_rate_hz_, 1, 1, in RunCaptureSaturationVerificationTest() 494 config, sample_rate_hz_, 1, 1, in RunRenderSwapQueueVerificationTest() 503 if (sample_rate_hz_ > 16000) { in RunRenderSwapQueueVerificationTest() [all …]
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D | suppression_filter.cc | 67 sample_rate_hz_(sample_rate_hz), in SuppressionFilter() 70 e_output_old_(NumBandsForRate(sample_rate_hz_), in SuppressionFilter() 73 RTC_DCHECK(ValidFullBandRate(sample_rate_hz_)); in SuppressionFilter() 91 RTC_DCHECK_EQ(e->size(), NumBandsForRate(sample_rate_hz_)); in ApplyGain()
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D | block_processor.cc | 73 const size_t sample_rate_hz_; member in webrtc::__anon472254dc0111::BlockProcessorImpl 96 sample_rate_hz_(sample_rate_hz), in BlockProcessorImpl() 101 RTC_DCHECK(ValidFullBandRate(sample_rate_hz_)); in BlockProcessorImpl() 112 RTC_DCHECK_EQ(NumBandsForRate(sample_rate_hz_), capture_block->size()); in ProcessCapture() 205 RTC_DCHECK_EQ(NumBandsForRate(sample_rate_hz_), block.size()); in BufferRender()
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/external/webrtc/modules/audio_processing/ |
D | voice_detection.cc | 38 : sample_rate_hz_(sample_rate_hz), in VoiceDetection() 39 frame_size_samples_(static_cast<size_t>(sample_rate_hz_ / 100)), in VoiceDetection() 87 int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_, in ProcessCaptureAudio()
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D | high_pass_filter.cc | 56 : sample_rate_hz_(sample_rate_hz) { in HighPassFilter() 58 const auto& coefficients = ChooseCoefficients(sample_rate_hz_); in HighPassFilter() 107 const auto& coefficients = ChooseCoefficients(sample_rate_hz_); in Reset()
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D | high_pass_filter.h | 36 int sample_rate_hz() const { return sample_rate_hz_; } in sample_rate_hz() 40 const int sample_rate_hz_;
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/external/webrtc/modules/audio_processing/test/conversational_speech/ |
D | multiend_call.cc | 47 sample_rate_hz_ = 0; // Sample rate will be set when reading the first track. in CreateAudioTrackReaders() 61 if (sample_rate_hz_ == 0) { in CreateAudioTrackReaders() 62 sample_rate_hz_ = wavreader->SampleRate(); in CreateAudioTrackReaders() 63 } else if (sample_rate_hz_ != wavreader->SampleRate()) { in CreateAudioTrackReaders()
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/external/webrtc/modules/audio_coding/acm2/ |
D | audio_coding_module.cc | 351 if (audio_frame.sample_rate_hz_ > kMaxInputSampleRateHz) { in Add10MsDataInternal() 358 if (static_cast<size_t>(audio_frame.sample_rate_hz_ / 100) != in Add10MsDataInternal() 423 in_frame.sample_rate_hz_ != encoder_stack_->SampleRateHz(); in PreprocessToAddData() 443 static_cast<double>(in_frame.sample_rate_hz_)); in PreprocessToAddData() 490 preprocess_frame_.sample_rate_hz_ = in_frame.sample_rate_hz_; in PreprocessToAddData() 497 src_ptr_audio, in_frame.sample_rate_hz_, encoder_stack_->SampleRateHz(), in PreprocessToAddData() 507 preprocess_frame_.sample_rate_hz_ = encoder_stack_->SampleRateHz(); in PreprocessToAddData()
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/external/webrtc/modules/audio_coding/codecs/cng/ |
D | audio_encoder_cng_unittest.cc | 48 sample_rate_hz_(8000) { in AudioEncoderCngTest() 72 num_audio_samples_10ms_ = static_cast<size_t>(10 * sample_rate_hz_ / 1000); in CreateCng() 76 .WillRepeatedly(Return(sample_rate_hz_)); in CreateCng() 161 VoiceActivity(_, expected_first_block_size_ms * sample_rate_hz_ / 1000, in CheckVadInputSize() 162 sample_rate_hz_)) in CheckVadInputSize() 167 _, expected_second_block_size_ms * sample_rate_hz_ / 1000, in CheckVadInputSize() 168 sample_rate_hz_)) in CheckVadInputSize() 210 int sample_rate_hz_; member in webrtc::AudioEncoderCngTest
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