| /external/webrtc/modules/audio_processing/agc2/ |
| D | vad_wrapper.cc | 58 int sample_rate_hz) in VoiceActivityDetectorWrapper()
66 int sample_rate_hz) in VoiceActivityDetectorWrapper()
80 void VoiceActivityDetectorWrapper::Initialize(int sample_rate_hz) { in Initialize()
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| D | limiter_unittest.cc | 23 const int sample_rate_hz = 48000; in TEST() local
34 const int sample_rate_hz = 48000; in TEST() local
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| D | limiter.cc | 89 void CheckLimiterSampleRate(int sample_rate_hz) { in CheckLimiterSampleRate()
97 Limiter::Limiter(int sample_rate_hz, in Limiter()
142 void Limiter::SetSampleRate(int sample_rate_hz) { in SetSampleRate()
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| D | fixed_digital_level_estimator.cc | 37 int sample_rate_hz, in FixedDigitalLevelEstimator()
109 void FixedDigitalLevelEstimator::SetSampleRate(int sample_rate_hz) { in SetSampleRate()
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| D | agc2_testing_common.cc | 49 int sample_rate_hz) in SineGenerator()
70 int sample_rate_hz) in PulseGenerator()
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| D | adaptive_digital_gain_controller.cc | 46 int sample_rate_hz, in AdaptiveDigitalGainController()
63 void AdaptiveDigitalGainController::Initialize(int sample_rate_hz, in Initialize()
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| D | noise_level_estimator_unittest.cc | 34 int sample_rate_hz) { in RunEstimator()
50 int sample_rate_hz() const { return GetParam(); } in sample_rate_hz() function in webrtc::__anon5217117c0111::NoiseEstimatorParametrization
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| /external/webrtc/modules/audio_processing/transient/ |
| D | voice_probability_delay_unit.cc | 20 int sample_rate_hz) { in VoiceProbabilityDelayUnit()
25 int sample_rate_hz) { in Initialize()
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| /external/webrtc/modules/audio_coding/codecs/opus/ |
| D | audio_decoder_opus.cc | 24 int sample_rate_hz) in AudioDecoderOpusImpl()
60 int sample_rate_hz, in DecodeInternal()
75 int sample_rate_hz, in DecodeRedundantInternal()
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| D | opus_interface.cc | 45 static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) { in FrameSizePerChannel()
54 static int MaxFrameSizePerChannel(int sample_rate_hz) { in MaxFrameSizePerChannel()
59 static int DefaultFrameSizePerChannel(int sample_rate_hz) { in DefaultFrameSizePerChannel()
106 int sample_rate_hz) { in WebRtcOpus_EncoderCreate()
173 const int sample_rate_hz = 48000; in WebRtcOpus_MultistreamEncoderCreate() local
469 int sample_rate_hz) { in WebRtcOpus_DecoderCreate()
746 int sample_rate_hz) { in WebRtcOpus_FecDurationEst()
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| D | opus_inst.h | 29 int sample_rate_hz; member
40 int sample_rate_hz; member
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| /external/webrtc/modules/audio_processing/aec3/ |
| D | block_processor.cc | 86 int sample_rate_hz, in BlockProcessorImpl()
241 int sample_rate_hz, in Create()
260 int sample_rate_hz, in Create()
278 int sample_rate_hz, in Create()
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| D | frame_blocker_unittest.cc | 106 void RunBlockerTest(int sample_rate_hz, size_t num_channels) { in RunBlockerTest()
141 void RunBlockerAndFramerTest(int sample_rate_hz, size_t num_channels) { in RunBlockerAndFramerTest()
188 int sample_rate_hz, in RunWronglySizedInsertAndExtractParametersTest()
213 void RunWronglySizedExtractParameterTest(int sample_rate_hz, in RunWronglySizedExtractParameterTest()
241 void RunWrongExtractOrderTest(int sample_rate_hz, in RunWrongExtractOrderTest()
262 std::string ProduceDebugText(int sample_rate_hz, size_t num_channels) { in ProduceDebugText()
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| D | block_processor_unittest.cc | 39 void RunBasicSetupAndApiCallTest(int sample_rate_hz, int num_iterations) { in RunBasicSetupAndApiCallTest()
55 void RunRenderBlockSizeVerificationTest(int sample_rate_hz) { in RunRenderBlockSizeVerificationTest()
67 void RunRenderNumBandsVerificationTest(int sample_rate_hz) { in RunRenderNumBandsVerificationTest()
82 void RunCaptureNumBandsVerificationTest(int sample_rate_hz) { in RunCaptureNumBandsVerificationTest()
99 std::string ProduceDebugText(int sample_rate_hz) { in ProduceDebugText()
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| D | block_framer_unittest.cc | 80 void RunFramerTest(int sample_rate_hz, size_t num_channels) { in RunFramerTest()
113 int sample_rate_hz, in RunWronglySizedInsertAndExtractParametersTest()
138 void RunWronglySizedInsertParameterTest(int sample_rate_hz, in RunWronglySizedInsertParameterTest()
167 void RunWronglyInsertOrderTest(int sample_rate_hz, in RunWronglyInsertOrderTest()
190 std::string ProduceDebugText(int sample_rate_hz, size_t num_channels) { in ProduceDebugText()
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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()
41 int sample_rate_hz, in DecodeInternal()
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| /external/webrtc/test/fuzzers/ |
| D | aec3_fuzzer.cc | 21 void PrepareAudioBuffer(int sample_rate_hz, in PrepareAudioBuffer()
46 const int sample_rate_hz = in FuzzOneInput() local
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| D | agc_fuzzer.cc | 23 void FillAudioBuffer(size_t sample_rate_hz, in FillAudioBuffer()
86 const auto sample_rate_hz = in FuzzGainController() local
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| /external/webrtc/api/audio_codecs/ |
| D | audio_format.cc | 60 AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, in AudioCodecInfo()
69 AudioCodecInfo::AudioCodecInfo(int sample_rate_hz, in AudioCodecInfo()
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| D | audio_decoder.cc | 84 int sample_rate_hz, in Decode()
101 int sample_rate_hz, in DecodeRedundant()
118 int sample_rate_hz, in DecodeRedundantInternal()
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| /external/webrtc/audio/ |
| D | remix_resample_unittest.cc | 24 int GetFrameSize(int sample_rate_hz) { in GetFrameSize()
52 void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) { in SetMonoFrame()
72 int sample_rate_hz, in SetStereoFrame()
96 int sample_rate_hz, in SetQuadFrame()
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| /external/webrtc/modules/audio_processing/ |
| D | high_pass_filter.cc | 38 int sample_rate_hz) { in ChooseCoefficients()
55 HighPassFilter::HighPassFilter(int sample_rate_hz, size_t num_channels) in HighPassFilter()
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| D | echo_control_mobile_impl.cc | 70 int sample_rate_hz; member
95 void Initialize(int sample_rate_hz) { in Initialize()
245 void EchoControlMobileImpl::Initialize(int sample_rate_hz, in Initialize()
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| /external/webrtc/modules/audio_processing/test/ |
| D | bitexactness_tools.cc | 25 std::string GetApmRenderTestVectorFileName(int sample_rate_hz) { in GetApmRenderTestVectorFileName()
41 std::string GetApmCaptureTestVectorFileName(int sample_rate_hz) { in GetApmCaptureTestVectorFileName()
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| /external/webrtc/modules/audio_coding/codecs/g711/ |
| D | audio_decoder_pcm.cc | 39 int sample_rate_hz, in DecodeInternal()
80 int sample_rate_hz, in DecodeInternal()
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