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Functions772 in github.com/Ryuk17/SpeechAlgorithms

↓ 2 callersFunctionMulIm
WebRTC_AEC/src/aec_core.c:145
↓ 2 callersFunctionPaddingOrTruncate
(sample, length=16000)
CommandRecognition/utils.py:21
↓ 2 callersFunctionSatW32ToW16
WebRTC_AGC/src/agc.c:215
↓ 2 callersFunctionSetBit
WebRTC_AEC/src/delay_estimator_wrapper.c:26
↓ 2 callersFunctionSineInterpolation
(x, w, L, M)
Resample/Algorithm.py:49
↓ 2 callersFunctionStateProb
(h, min, max)
VoiceChange/td_psola.py:193
↓ 2 callersFunctionWebRtcAec_ProcessFrame
WebRTC_AEC/src/aec_core.c:601
↓ 2 callersFunctionWebRtcAec_set_config
WebRTC_AEC/src/echo_cancellation.c:451
↓ 2 callersFunctionWebRtcAgc_CalculateGainTable
WebRTC_AGC/src/agc.c:374
↓ 2 callersFunctionWebRtcAgc_ExpCurve
WebRTC_AGC/src/agc.c:1443
↓ 2 callersFunctionWebRtcAgc_UpdateAgcThresholds
WebRTC_AGC/src/agc.c:1285
↓ 2 callersFunctionWebRtcAgc_set_config
WebRTC_AGC/src/agc.c:2003
↓ 2 callersFunctionWebRtcNs_set_policy_core
Changes the aggressiveness of the noise suppression method. |mode| = 0 is mild (6dB), |mode| = 1 is medium (10dB) and |mode| = 2 is aggressive (15dB).
WebRTC_ANR/src/noise_suppression.c:1700
↓ 2 callersFunctionWebRtcSpl_AddSatW16
WebRTC_AGC/src/agc.c:230
↓ 2 callersFunctionWebRtcSpl_IncreaseSeed
WebRTC_AEC/src/randomization_functions.c:92
↓ 2 callersFunctionWebRtcSpl_SatW32ToW16
WebRTC_AEC/src/spl_inl.h:27
↓ 2 callersFunctionWebRtcVad_GaussianProbability
For a normal distribution, the probability of |input| is calculated and returned (in Q20). The formula for normal distributed probability is 1 / s *
WebRTC_VAD/src/vad_gmm.c:29
↓ 2 callersFunctionWebRtcVad_set_mode_core
Set aggressiveness mode
WebRTC_VAD/src/vad_core.c:546
↓ 2 callersFunctionWebRtc_AddBinaryFarSpectrum
WebRTC_AEC/src/delay_estimator.c:311
↓ 2 callersFunctionWebRtc_FreeBinaryDelayEstimator
WebRTC_AEC/src/delay_estimator.c:326
↓ 2 callersFunctionWebRtc_FreeBinaryDelayEstimatorFarend
WebRTC_AEC/src/delay_estimator.c:261
↓ 2 callersFunctionWebRtc_FreeDelayEstimator
WebRTC_AEC/src/delay_estimator_wrapper.c:248
↓ 2 callersFunctionWebRtc_FreeDelayEstimatorFarend
WebRTC_AEC/src/delay_estimator_wrapper.c:126
↓ 2 callersFunctionWebRtc_MeanEstimatorFix
WebRTC_AEC/src/delay_estimator.c:567
↓ 2 callersFunctionWebRtc_ProcessBinarySpectrum
WebRTC_AEC/src/delay_estimator.c:416
↓ 2 callersFunctionWindowingEnergy
WebRTC_ANR/src/noise_suppression.c:1638
↓ 2 callersFunction__clz_uint32
WebRTC_AGC/src/agc.c:162
↓ 2 callersMethod__init__
(self, block, num_blocks, num_classes=6)
CommandRecognition/model.py:72
↓ 2 callersMethod_get_feats
(self, data, sr)
VowelConsonantDivision/feats.py:66
↓ 2 callersMethod_load_gmm
(self, gmmname)
VowelConsonantDivision/gmm.py:97
↓ 2 callersMethod_multi_process_gmm
(self, train_data, gmmname)
VowelConsonantDivision/gmm.py:81
↓ 2 callersFunctionaddTogether
(y, x, t, maxLen)
VoiceChange/td_psola.py:334
↓ 2 callersFunctionbitrv2_128
WebRTC_AEC/src/aec_rdft.c:119
↓ 2 callersFunctionbuild_linear_to_xlaw_table
SpeechCodec/G711/src/g711_table.c:14
↓ 2 callersFunctionbuild_xlaw_to_linear_table
SpeechCodec/G711/src/g711_table.c:24
↓ 2 callersFunctioncft1st
WebRTC_ANR/src/noise_suppression.c:426
↓ 2 callersFunctioncftmdl
WebRTC_ANR/src/noise_suppression.c:530
↓ 2 callersFunctiondrwav__bswap_samples
WebRTC_VAD/src/dr_wav.h:1446
↓ 2 callersFunctiondrwav__read_chunk_header
WebRTC_AGC/src/dr_wav.h:843
↓ 2 callersFunctiondrwav__riff_chunk_size_riff
WebRTC_VAD/src/dr_wav.h:2251
↓ 2 callersFunctiondrwav__riff_chunk_size_w64
WebRTC_VAD/src/dr_wav.h:2269
↓ 2 callersFunctiondrwav__seek_forward
WebRTC_AGC/src/dr_wav.h:876
↓ 2 callersFunctiondrwav__seek_forward
WebRTC_VAD/src/dr_wav.h:1625
↓ 2 callersFunctiondrwav__seek_from_start
WebRTC_VAD/src/dr_wav.h:1644
↓ 2 callersFunctiondrwav_copy_allocation_callbacks_or_defaults
WebRTC_VAD/src/dr_wav.h:1550
↓ 2 callersFunctiondrwav_fopen
WebRTC_VAD/src/dr_wav.h:3015
↓ 2 callersFunctiondrwav_init_file__internal_FILE
WebRTC_VAD/src/dr_wav.h:3174
↓ 2 callersFunctiondrwav_init_file_write__internal
WebRTC_VAD/src/dr_wav.h:3245
↓ 2 callersFunctiondrwav_init_file_write__internal_FILE
WebRTC_VAD/src/dr_wav.h:3224
↓ 2 callersFunctiondrwav_init_file_write_w__internal
WebRTC_VAD/src/dr_wav.h:3258
↓ 2 callersFunctiondrwav_init_memory_write__internal
WebRTC_VAD/src/dr_wav.h:3467
↓ 2 callersFunctiondrwav_open
WebRTC_AGC/src/dr_wav.h:1712
↓ 2 callersFunctiondrwav_open_write
WebRTC_AGC/src/dr_wav.h:1727
↓ 2 callersFunctiondrwav_read_pcm_frames_s16__ima
WebRTC_VAD/src/dr_wav.h:4083
↓ 2 callersFunctiondrwav_read_pcm_frames_s16__msadpcm
WebRTC_VAD/src/dr_wav.h:3896
↓ 2 callersFunctiondrwav_read_s16__ima
WebRTC_AGC/src/dr_wav.h:2088
↓ 2 callersFunctiondrwav_read_s16__msadpcm
WebRTC_AGC/src/dr_wav.h:1920
↓ 2 callersFunctiondrwav_seek_to_first_sample
WebRTC_AGC/src/dr_wav.h:1785
↓ 2 callersFunctiondrwav_wfopen
WebRTC_VAD/src/dr_wav.h:3073
↓ 2 callersFunctiondrwav_write_raw
WebRTC_VAD/src/dr_wav.h:3781
↓ 2 callersFunctionfast_sqrt
WebRTC_AGC/src/agc.c:256
↓ 2 callersMethodgetNoiseSpectrum
estimate noise spectrum by using the front self.noise_frames frames :return: noise spectrum
SpectralSubtraction/SpectralSubtraction.py:26
↓ 2 callersFunctioninfer
(model, loader)
VoiceActivityDetection/train.py:22
↓ 2 callersFunctioninfer
(model, loader)
GenderClassify/train.py:22
↓ 2 callersFunctioninfer
(model, loader, criterion)
CommandRecognition/train.py:22
↓ 2 callersFunctionnanotimer
WebRTC_VAD/src/timing.h:20
↓ 2 callersFunctionnorm_l
This is a GSM function, for concrete processors she may be replaced for same processor's optimized function (norm_l)
Goertzel/DtmfDetector.cpp:54
↓ 2 callersFunctionpcm16_to_xlaw
SpeechCodec/G711/src/g711_table.c:34
↓ 2 callersFunctionsearch
SpeechCodec/G711/src/g711.c:103
↓ 2 callersFunctionsgn
(data)
VoiceActivityDetection/utils.py:48
↓ 2 callersFunctionxlaw_to_pcm16
SpeechCodec/G711/src/g711_table.c:47
↓ 1 callersFunctionAutoCorrelation
(x, lags)
VoiceChange/td_psola.py:64
↓ 1 callersMethodBeroutiSpectralSubtraction
Berouti spectral subtraction :return: enhanced speech
SpectralSubtraction/SpectralSubtraction.py:82
↓ 1 callersFunctionBitCountComparison
Compares the |binary_vector| with all rows of the |binary_matrix| and counts per row the number of times they have the same value. Inputs: - binary_v
WebRTC_AEC/src/delay_estimator.c:66
↓ 1 callersFunctionCenterClipping
(x, clip_rate)
VoiceChange/td_psola.py:54
↓ 1 callersFunctionComfortNoise
WebRTC_AEC/src/aec_core.c:1365
↓ 1 callersFunctionComputeDdBasedWienerFilter
Estimate prior SNR decision-directed and compute DD based Wiener Filter. Input: |magn| is the signal magnitude spectrum estimate. Output: |theFilter|
WebRTC_ANR/src/noise_suppression.c:1673
↓ 1 callersFunctionComputeSnr
Compute prior and post SNR based on quantile noise estimation. Compute DD estimate of prior SNR. Inputs: |magn| is the signal magnitude spectrum estim
WebRTC_ANR/src/noise_suppression.c:1242
↓ 1 callersFunctionComputeSpectralDifference
Compute the difference measure between input spectrum and a template/learned noise spectrum. |magnIn| is the input spectrum. The reference/template sp
WebRTC_ANR/src/noise_suppression.c:1271
↓ 1 callersFunctionComputeSpectralFlatness
Compute spectral flatness on input spectrum. |magnIn| is the magnitude spectrum. Spectral flatness is returned in self->featureData[0].
WebRTC_ANR/src/noise_suppression.c:1199
↓ 1 callersFunctionDotProductWithScale
WebRTC_AGC/src/agc.c:234
↓ 1 callersFunctionEnergy
Calculates the energy of a buffer. Inputs: |buffer| is the buffer over which the energy is calculated. |length| is the length of the buffer. Returns t
WebRTC_ANR/src/noise_suppression.c:1656
↓ 1 callersFunctionEstBufDelayExtended
WebRTC_AEC/src/echo_cancellation.c:920
↓ 1 callersFunctionEstBufDelayNormal
WebRTC_AEC/src/echo_cancellation.c:867
↓ 1 callersFunctionEstimateSkew
WebRTC_AEC/src/aec_resampler.c:152
↓ 1 callersFunctionFeatureUpdate
Update the noise features. Inputs: |magn| is the signal magnitude spectrum estimate. |updateParsFlag| is an update flag for parameters.
WebRTC_ANR/src/noise_suppression.c:1423
↓ 1 callersFunctionFindPeakCandidates
(x, MaxCandidateNumber, Offset)
VoiceChange/td_psola.py:127
↓ 1 callersFunctionGetBufferReadRegions
Get address of region(s) from which we can read data. If the region is contiguous, |data_ptr_bytes_2| will be zero. If non-contiguous, |data_ptr_bytes
WebRTC_AEC/src/ring_buffer.c:39
↓ 1 callersFunctionGetHighbandGain
WebRTC_AEC/src/aec_core.c:1355
↓ 1 callersFunctionGmmProbability
Calculates the probabilities for both speech and background noise using Gaussian Mixture Models (GMM). A hypothesis-test is performed to decide which
WebRTC_VAD/src/vad_core.c:131
↓ 1 callersFunctionHighPassFilter
High pass filtering, with a cut-off frequency at 80 Hz, if the |data_in| is sampled at 500 Hz. - data_in [i] : Input audio data sampled at 500
WebRTC_VAD/src/vad_filterbank.c:36
↓ 1 callersFunctionHistogramBasedValidation
Validates the |candidate_delay|, estimated in WebRtc_ProcessBinarySpectrum(), based on a mix of counting concurring hits with a modified histogram of
WebRTC_AEC/src/delay_estimator.c:173
↓ 1 callersFunctionIFFT
Transforms the signal from frequency to time domain. Inputs: |real| is the real part of the frequency domain. |imag| is the imaginary part of the freq
WebRTC_ANR/src/noise_suppression.c:1613
↓ 1 callersFunctionIsPeak
(index, low, high, x)
VoiceChange/td_psola.py:72
↓ 1 callersFunctionMeanEstimatorFloat
Calculates the mean recursively. Same version as WebRtc_MeanEstimatorFix(), but for float. Inputs: - new_value : New additional value. -
WebRTC_AEC/src/delay_estimator_wrapper.c:43
↓ 1 callersFunctionNoiseEstimation
Estimate noise.
WebRTC_ANR/src/noise_suppression.c:902
↓ 1 callersFunctionNonLinearProcessing
WebRTC_AEC/src/aec_core.c:1002
↓ 1 callersFunctionOverflowingMulS16ByS32ToS32
An s16 x s32 -> s32 multiplication that's allowed to overflow. (It's still undefined behavior, so not a good idea; this just makes UBSan ignore the vi
WebRTC_VAD/src/vad_core.c:115
↓ 1 callersFunctionOverlap1
()
EnframeWindowFFT/fft-window.py:15
↓ 1 callersFunctionOverlap2
()
EnframeWindowFFT/fft-window.py:37
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