/external/webrtc/modules/audio_processing/ |
D | echo_control_mobile_bit_exact_unittest.cc | 26 void SetupComponent(int sample_rate_hz, in SetupComponent() argument 31 sample_rate_hz > 16000 ? 16000 : sample_rate_hz, 1, 1); in SetupComponent() 36 void ProcessOneFrame(int sample_rate_hz, in ProcessOneFrame() argument 41 if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) { in ProcessOneFrame() 55 if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) { in ProcessOneFrame() 60 void RunBitexactnessTest(int sample_rate_hz, in RunBitexactnessTest() argument 67 SetupComponent(sample_rate_hz, routing_mode, comfort_noise_enabled, in RunBitexactnessTest() 70 const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); in RunBitexactnessTest() 71 const StreamConfig render_config(sample_rate_hz, num_channels, false); in RunBitexactnessTest() 73 render_config.sample_rate_hz(), render_config.num_channels(), in RunBitexactnessTest() [all …]
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D | voice_detection_unittest.cc | 25 bool ProcessOneFrame(int sample_rate_hz, in ProcessOneFrame() argument 28 if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) { in ProcessOneFrame() 37 void RunBitexactnessTest(int sample_rate_hz, in RunBitexactnessTest() argument 40 int sample_rate_to_use = std::min(sample_rate_hz, 16000); in RunBitexactnessTest() 44 int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); in RunBitexactnessTest() 45 const StreamConfig capture_config(sample_rate_hz, num_channels, false); in RunBitexactnessTest() 47 capture_config.sample_rate_hz(), capture_config.num_channels(), in RunBitexactnessTest() 48 capture_config.sample_rate_hz(), capture_config.num_channels(), in RunBitexactnessTest() 49 capture_config.sample_rate_hz(), capture_config.num_channels()); in RunBitexactnessTest() 51 test::GetApmCaptureTestVectorFileName(sample_rate_hz)); in RunBitexactnessTest() [all …]
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D | gain_control_unittest.cc | 24 void ProcessOneFrame(int sample_rate_hz, in ProcessOneFrame() argument 28 if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) { in ProcessOneFrame() 39 if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) { in ProcessOneFrame() 44 void SetupComponent(int sample_rate_hz, in SetupComponent() argument 53 gain_controller->Initialize(1, sample_rate_hz); in SetupComponent() 63 void RunBitExactnessTest(int sample_rate_hz, in RunBitExactnessTest() argument 75 SetupComponent(sample_rate_hz, mode, target_level_dbfs, stream_analog_level, in RunBitExactnessTest() 79 const int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); in RunBitExactnessTest() 80 const StreamConfig render_config(sample_rate_hz, num_channels, false); in RunBitExactnessTest() 82 render_config.sample_rate_hz(), render_config.num_channels(), in RunBitExactnessTest() [all …]
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D | level_estimator_unittest.cc | 26 void RunBitexactnessTest(int sample_rate_hz, in RunBitexactnessTest() argument 30 int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); in RunBitexactnessTest() 31 StreamConfig capture_config(sample_rate_hz, num_channels, false); in RunBitexactnessTest() 33 capture_config.sample_rate_hz(), capture_config.num_channels(), in RunBitexactnessTest() 34 capture_config.sample_rate_hz(), capture_config.num_channels(), in RunBitexactnessTest() 35 capture_config.sample_rate_hz(), capture_config.num_channels()); in RunBitexactnessTest() 38 test::GetApmCaptureTestVectorFileName(sample_rate_hz)); in RunBitexactnessTest()
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D | audio_processing_unittest.cc | 247 std::string ResourceFilePath(const std::string& name, int sample_rate_hz) { in ResourceFilePath() argument 250 ss << name << sample_rate_hz / 1000 << "_stereo"; in ResourceFilePath() 364 void Init(int sample_rate_hz, 463 ap->Initialize({{{frame_.sample_rate_hz, frame_.num_channels}, in Init() 465 {revframe_.sample_rate_hz, revframe_.num_channels}, in Init() 466 {revframe_.sample_rate_hz, revframe_.num_channels}}})); in Init() 469 void ApmTest::Init(int sample_rate_hz, in Init() argument 476 SetContainerFormat(sample_rate_hz, num_input_channels, &frame_, &float_cb_); in Init() 487 std::string filename = ResourceFilePath("far", sample_rate_hz); in Init() 494 filename = ResourceFilePath("near", sample_rate_hz); in Init() [all …]
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D | gain_controller2.cc | 38 void GainController2::Initialize(int sample_rate_hz) { in Initialize() argument 39 RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz || in Initialize() 40 sample_rate_hz == AudioProcessing::kSampleRate16kHz || in Initialize() 41 sample_rate_hz == AudioProcessing::kSampleRate32kHz || in Initialize() 42 sample_rate_hz == AudioProcessing::kSampleRate48kHz); in Initialize() 43 limiter_.SetSampleRate(sample_rate_hz); in Initialize() 45 data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz); in Initialize()
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/external/webrtc/test/fuzzers/ |
D | aec3_fuzzer.cc | 20 void PrepareAudioBuffer(int sample_rate_hz, in PrepareAudioBuffer() argument 30 if (sample_rate_hz == 32000 || sample_rate_hz == 48000) { in PrepareAudioBuffer() 45 const int sample_rate_hz = in FuzzOneInput() local 54 EchoCanceller3 aec3(EchoCanceller3Config(), sample_rate_hz, in FuzzOneInput() 57 AudioBuffer capture_audio(sample_rate_hz, num_capture_channels, in FuzzOneInput() 58 sample_rate_hz, num_capture_channels, in FuzzOneInput() 59 sample_rate_hz, num_capture_channels); in FuzzOneInput() 60 AudioBuffer render_audio(sample_rate_hz, num_render_channels, sample_rate_hz, in FuzzOneInput() 61 num_render_channels, sample_rate_hz, in FuzzOneInput() 69 PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &capture_audio); in FuzzOneInput() [all …]
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/external/webrtc/modules/audio_coding/codecs/isac/ |
D | isac_webrtc_api_test.cc | 44 std::unique_ptr<PCMFile> GetPcmTestFileReader(int sample_rate_hz) { in GetPcmTestFileReader() argument 46 switch (sample_rate_hz) { in GetPcmTestFileReader() 54 RTC_NOTREACHED() << "No test file available for " << sample_rate_hz in GetPcmTestFileReader() 59 pcm_file->Open(filename, sample_rate_hz, "rb", /*auto_rewind=*/true); in GetPcmTestFileReader() 72 int sample_rate_hz, in CreateEncoder() argument 75 RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000); in CreateEncoder() 83 RTC_CHECK_EQ(16000, sample_rate_hz); in CreateEncoder() 90 config.sample_rate_hz = sample_rate_hz; in CreateEncoder() 96 std::unique_ptr<AudioDecoder> CreateDecoder(IsacImpl impl, int sample_rate_hz) { in CreateDecoder() argument 97 RTC_CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000); in CreateDecoder() [all …]
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D | audio_decoder_isac_t_impl.h | 20 return (sample_rate_hz == 16000 || sample_rate_hz == 32000); in IsOk() 25 : sample_rate_hz_(config.sample_rate_hz) { in AudioDecoderIsacT() 27 << config.sample_rate_hz; in AudioDecoderIsacT() 41 int sample_rate_hz, in DecodeInternal() argument 44 RTC_CHECK_EQ(sample_rate_hz_, sample_rate_hz); in DecodeInternal()
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/external/webrtc/modules/audio_coding/codecs/pcm16b/ |
D | audio_decoder_pcm16b.cc | 21 AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels) in AudioDecoderPcm16B() argument 22 : sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) { in AudioDecoderPcm16B() 23 RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 || in AudioDecoderPcm16B() 24 sample_rate_hz == 32000 || sample_rate_hz == 48000) in AudioDecoderPcm16B() 25 << "Unsupported sample rate " << sample_rate_hz; in AudioDecoderPcm16B() 41 int sample_rate_hz, in DecodeInternal() argument 44 RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz); in DecodeInternal()
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D | pcm16b_common.cc | 21 for (int sample_rate_hz : {8000, 16000, 32000}) { in Pcm16BAppendSupportedCodecSpecs() 23 {{"L16", sample_rate_hz, num_channels}, in Pcm16BAppendSupportedCodecSpecs() 24 {sample_rate_hz, num_channels, sample_rate_hz * num_channels * 16}}); in Pcm16BAppendSupportedCodecSpecs()
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/external/webrtc/modules/audio_coding/codecs/opus/ |
D | opus_interface.cc | 39 static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) { in FrameSizePerChannel() argument 42 RTC_DCHECK_GT(sample_rate_hz, 0); in FrameSizePerChannel() 43 RTC_DCHECK_EQ(sample_rate_hz % 1000, 0); in FrameSizePerChannel() 44 return frame_size_ms * (sample_rate_hz / 1000); in FrameSizePerChannel() 48 static int MaxFrameSizePerChannel(int sample_rate_hz) { in MaxFrameSizePerChannel() argument 49 return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz); in MaxFrameSizePerChannel() 53 static int DefaultFrameSizePerChannel(int sample_rate_hz) { in DefaultFrameSizePerChannel() argument 54 return FrameSizePerChannel(20, sample_rate_hz); in DefaultFrameSizePerChannel() 60 int sample_rate_hz) { in WebRtcOpus_EncoderCreate() argument 82 sample_rate_hz, static_cast<int>(channels), opus_app, &error); in WebRtcOpus_EncoderCreate() [all …]
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D | audio_decoder_opus.cc | 24 int sample_rate_hz) in AudioDecoderOpusImpl() argument 25 : channels_{num_channels}, sample_rate_hz_{sample_rate_hz} { in AudioDecoderOpusImpl() 27 RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000); 60 int sample_rate_hz, in DecodeInternal() argument 63 RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_); in DecodeInternal() 75 int sample_rate_hz, in DecodeRedundantInternal() argument 80 return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded, in DecodeRedundantInternal() 84 RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_); in DecodeRedundantInternal()
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/external/webrtc/modules/audio_processing/agc2/ |
D | limiter_unittest.cc | 23 const int sample_rate_hz = 48000; in TEST() local 26 Limiter limiter(sample_rate_hz, &apm_data_dumper, ""); in TEST() 28 VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100, in TEST() 34 const int sample_rate_hz = 48000; in TEST() local 40 Limiter limiter(sample_rate_hz, &apm_data_dumper, ""); in TEST() 44 VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100, in TEST() 49 VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100, in TEST()
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D | limiter.cc | 87 void CheckLimiterSampleRate(size_t sample_rate_hz) { in CheckLimiterSampleRate() argument 89 RTC_DCHECK_LE(sample_rate_hz, in CheckLimiterSampleRate() 95 Limiter::Limiter(size_t sample_rate_hz, in Limiter() argument 99 level_estimator_(sample_rate_hz, apm_data_dumper), in Limiter() 101 CheckLimiterSampleRate(sample_rate_hz); in Limiter() 137 void Limiter::SetSampleRate(size_t sample_rate_hz) { in SetSampleRate() argument 138 CheckLimiterSampleRate(sample_rate_hz); in SetSampleRate() 139 level_estimator_.SetSampleRate(sample_rate_hz); in SetSampleRate()
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D | fixed_digital_level_estimator_unittest.cc | 29 void TestLevelEstimator(int sample_rate_hz, in TestLevelEstimator() argument 35 FixedDigitalLevelEstimator level_estimator(sample_rate_hz, &apm_data_dumper); in TestLevelEstimator() 38 num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), in TestLevelEstimator() 59 float TimeMsToDecreaseLevel(int sample_rate_hz, in TimeMsToDecreaseLevel() argument 67 num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), input_level); in TimeMsToDecreaseLevel() 70 FixedDigitalLevelEstimator level_estimator(sample_rate_hz, &apm_data_dumper); in TimeMsToDecreaseLevel() 82 num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), 0); in TimeMsToDecreaseLevel()
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D | down_sampler.cc | 54 void DownSampler::Initialize(int sample_rate_hz) { in Initialize() argument 56 sample_rate_hz == kSampleRate8kHz || sample_rate_hz == kSampleRate16kHz || in Initialize() 57 sample_rate_hz == kSampleRate32kHz || sample_rate_hz == kSampleRate48kHz); in Initialize() 59 sample_rate_hz_ = sample_rate_hz; in Initialize()
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/external/webrtc/modules/audio_processing/aec3/ |
D | block_processor_unittest.cc | 38 void RunBasicSetupAndApiCallTest(int sample_rate_hz, int num_iterations) { in RunBasicSetupAndApiCallTest() argument 43 BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz, in RunBasicSetupAndApiCallTest() 46 NumBandsForRate(sample_rate_hz), in RunBasicSetupAndApiCallTest() 57 void RunRenderBlockSizeVerificationTest(int sample_rate_hz) { in RunRenderBlockSizeVerificationTest() argument 62 BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz, in RunRenderBlockSizeVerificationTest() 65 NumBandsForRate(sample_rate_hz), in RunRenderBlockSizeVerificationTest() 72 void RunCaptureBlockSizeVerificationTest(int sample_rate_hz) { in RunCaptureBlockSizeVerificationTest() argument 77 BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz, in RunCaptureBlockSizeVerificationTest() 80 NumBandsForRate(sample_rate_hz), in RunCaptureBlockSizeVerificationTest() 88 void RunRenderNumBandsVerificationTest(int sample_rate_hz) { in RunRenderNumBandsVerificationTest() argument [all …]
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D | block_processor.cc | 41 int sample_rate_hz, 87 int sample_rate_hz, in BlockProcessorImpl() argument 96 sample_rate_hz_(sample_rate_hz), in BlockProcessorImpl() 242 int sample_rate_hz, in Create() argument 246 RenderDelayBuffer::Create(config, sample_rate_hz, num_render_channels)); in Create() 249 delay_controller.reset(RenderDelayController::Create(config, sample_rate_hz, in Create() 253 config, sample_rate_hz, num_render_channels, num_capture_channels)); in Create() 254 return Create(config, sample_rate_hz, num_render_channels, in Create() 261 int sample_rate_hz, in Create() argument 267 delay_controller.reset(RenderDelayController::Create(config, sample_rate_hz, in Create() [all …]
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D | aec3_common.h | 56 constexpr size_t NumBandsForRate(int sample_rate_hz) { in NumBandsForRate() argument 57 return static_cast<size_t>(sample_rate_hz / 16000); in NumBandsForRate() 60 constexpr bool ValidFullBandRate(int sample_rate_hz) { in ValidFullBandRate() argument 61 return sample_rate_hz == 16000 || sample_rate_hz == 32000 || in ValidFullBandRate() 62 sample_rate_hz == 48000; in ValidFullBandRate()
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/external/webrtc/api/audio_codecs/isac/ |
D | audio_encoder_isac_float.cc | 27 config.sample_rate_hz = format.clockrate_hz; in SdpToConfig() 29 if (config.sample_rate_hz == 16000) { in SdpToConfig() 48 for (int sample_rate_hz : {16000, 32000}) { in AppendSupportedEncoders() 49 const SdpAudioFormat fmt = {"ISAC", sample_rate_hz, 1}; in AppendSupportedEncoders() 59 const int max_bitrate = config.sample_rate_hz == 16000 ? 32000 : 56000; in QueryAudioEncoder() 61 return {config.sample_rate_hz, 1, default_bitrate, min_bitrate, max_bitrate}; in QueryAudioEncoder() 71 c.sample_rate_hz = config.sample_rate_hz; in MakeAudioEncoder()
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/external/webrtc/api/audio_codecs/L16/ |
D | audio_decoder_L16.h | 30 return (sample_rate_hz == 8000 || sample_rate_hz == 16000 || in IsOk() 31 sample_rate_hz == 32000 || sample_rate_hz == 48000) && in IsOk() 34 int sample_rate_hz = 8000; member
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D | audio_encoder_L16.h | 30 return (sample_rate_hz == 8000 || sample_rate_hz == 16000 || in IsOk() 31 sample_rate_hz == 32000 || sample_rate_hz == 48000) && in IsOk() 35 int sample_rate_hz = 8000; member
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/external/webrtc/modules/audio_processing/test/ |
D | simulator_buffers.cc | 53 int sample_rate_hz, in CreateConfigAndBuffer() argument 60 int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100); in CreateConfigAndBuffer() 61 *config = StreamConfig(sample_rate_hz, num_channels, false); in CreateConfigAndBuffer() 63 new AudioBuffer(config->sample_rate_hz(), config->num_channels(), in CreateConfigAndBuffer() 64 config->sample_rate_hz(), config->num_channels(), in CreateConfigAndBuffer() 65 config->sample_rate_hz(), config->num_channels())); in CreateConfigAndBuffer()
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/external/webrtc/api/audio_codecs/ |
D | audio_format.cc | 60 AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, in AudioCodecInfo() argument 63 : AudioCodecInfo(sample_rate_hz, in AudioCodecInfo() 69 AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, in AudioCodecInfo() argument 74 : sample_rate_hz(sample_rate_hz), in AudioCodecInfo() 79 RTC_DCHECK_GT(sample_rate_hz, 0); in AudioCodecInfo()
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