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Functions772 in github.com/Ryuk17/SpeechAlgorithms

⨍ Functions 772◇ Types & classes 51
↓ 27 callersFunctiondrwav__bytes_to_u32
WebRTC_AGC/src/dr_wav.h:785
↓ 25 callersFunctiondrwav__bytes_to_s16
WebRTC_AGC/src/dr_wav.h:780
↓ 25 callersFunctiondrwav_uninit
WebRTC_AGC/src/dr_wav.h:1636
↓ 18 callersFunctiondrwav__bytes_to_u16
WebRTC_AGC/src/dr_wav.h:771
↓ 18 callersFunctiondrwav_get_bytes_per_pcm_frame
WebRTC_VAD/src/dr_wav.h:1816
↓ 15 callersFunctiondrwav_read
WebRTC_AGC/src/dr_wav.h:1765
↓ 13 callersFunctionWebRtc_MoveReadPtr
WebRTC_AEC/src/ring_buffer.c:197
↓ 10 callersFunctionWebRtcAec_FreeAec
WebRTC_AEC/src/aec_core.c:244
↓ 10 callersFunctiondrwav__on_read
WebRTC_VAD/src/dr_wav.h:1783
↓ 9 callersFunctionWebRtc_CreateBuffer
WebRTC_AEC/src/ring_buffer.c:68
↓ 9 callersFunctionWebRtc_FreeBuffer
WebRTC_AEC/src/ring_buffer.c:107
↓ 9 callersFunctionWebRtc_InitBuffer
WebRTC_AEC/src/ring_buffer.c:92
↓ 9 callersFunctionWebRtc_ReadBuffer
WebRTC_AEC/src/ring_buffer.c:117
↓ 9 callersFunctionWebRtc_WriteBuffer
WebRTC_AEC/src/ring_buffer.c:163
↓ 9 callersFunctiondrwav_init
WebRTC_AGC/src/dr_wav.h:1311
↓ 9 callersFunctiongoertzel_filter
Goertzel/DtmfDetector.cpp:21
↓ 8 callersFunctionWebRtc_available_read
WebRTC_AEC/src/ring_buffer.c:234
↓ 8 callersFunctiondrwav__is_little_endian
WebRTC_VAD/src/dr_wav.h:1200
↓ 7 callersFunctionWebRtcSpl_DivW32W16
WebRTC_VAD/src/signal_processing_library.c:63
↓ 6 callersFunctionDivW32W16ResW16
WebRTC_AGC/src/agc.c:152
↓ 6 callersFunctionLogOfEnergy
Calculates the energy of |data_in| in dB, and also updates an overall |total_energy| if necessary. - data_in [i] : Input audio data for energy
WebRTC_VAD/src/vad_filterbank.c:149
↓ 6 callersFunctionWebRtcAec_MoveFarReadPtr
WebRTC_AEC/src/aec_core.c:591
↓ 6 callersFunctionWebRtcAec_system_delay
WebRTC_AEC/src/aec_core.c:780
↓ 6 callersMethod_LSBReplace
embed watermarking a single wave :param secret_message: secret message :return:
Watermarking/LSB.py:47
↓ 6 callersFunctiondrwav_init_file
WebRTC_AGC/src/dr_wav.h:1017
↓ 6 callersFunctiondrwav_init_memory
WebRTC_AGC/src/dr_wav.h:1214
↓ 6 callersFunctionlogsumexp
Stable log sum exp.
CtcSearcher/main.py:23
↓ 5 callersFunctionSplitFilter
Splits |data_in| into |hp_data_out| and |lp_data_out| corresponding to an upper (high pass) part and a lower (low pass) part respectively. - data_in
WebRTC_VAD/src/vad_filterbank.c:115
↓ 5 callersFunctionUpdateBuffer
Updates |buffer| with a new |frame|. Inputs: |frame| is a new speech frame or NULL for setting to zero. |frame_length| is the length of the new frame.
WebRTC_ANR/src/noise_suppression.c:1521
↓ 5 callersFunctionWebRtcAec_Free
WebRTC_AEC/src/echo_cancellation.c:183
↓ 5 callersFunctionWeightedAverage
Calculates the weighted average w.r.t. number of Gaussians. The |data| are updated with an |offset| before averaging. - data [i/o] : Data to aver
WebRTC_VAD/src/vad_core.c:99
↓ 5 callersFunctionaec_rdft_forward_128
WebRTC_AEC/src/aec_rdft.c:620
↓ 5 callersFunctionallocate_buffer
SpeechCodec/G711/src/g711_codec.c:41
↓ 5 callersFunctiondrwav__free_from_callbacks
WebRTC_VAD/src/dr_wav.h:1539
↓ 5 callersFunctiondrwav_read_s16
WebRTC_AGC/src/dr_wav.h:2422
↓ 5 callersFunctiondrwav_result_from_errno
WebRTC_VAD/src/dr_wav.h:2482
↓ 4 callersFunctionDivW32W16
WebRTC_AGC/src/agc.c:157
↓ 4 callersFunctionInitLevel
WebRTC_AEC/src/aec_core.c:1452
↓ 4 callersFunctionInitStats
WebRTC_AEC/src/aec_core.c:1464
↓ 4 callersFunctionMPY48SR
Goertzel/DtmfDetector.cpp:13
↓ 4 callersFunctionNormU32
Return the number of steps a can be left-shifted without overflow, or 0 if a == 0.
WebRTC_AGC/src/agc.c:208
↓ 4 callersFunctionUpdateLevel
WebRTC_AEC/src/aec_core.c:1489
↓ 4 callersFunctionWebRtcVad_CalcVad8khz
WebRTC_VAD/src/vad_core.c:668
↓ 4 callersMethod_make_layer
(self, block, out_channels, num_blocks, stride)
CommandRecognition/model.py:84
↓ 4 callersMethod_saveWave
save stego wave :param stego: stego wavsignal :param cover_path: cover path :param stego_path: stego path
Watermarking/LSB.py:85
↓ 4 callersFunctionaec_rdft_inverse_128
WebRTC_AEC/src/aec_rdft.c:630
↓ 4 callersFunctiondrwav__bswap_samples_s16
WebRTC_VAD/src/dr_wav.h:1318
↓ 4 callersFunctiondrwav__malloc_from_callbacks
WebRTC_VAD/src/dr_wav.h:1492
↓ 4 callersFunctiondrwav__read_pcm_frames_and_close_f32
WebRTC_VAD/src/dr_wav.h:5472
↓ 4 callersFunctiondrwav__read_pcm_frames_and_close_s16
WebRTC_VAD/src/dr_wav.h:5429
↓ 4 callersFunctiondrwav__read_pcm_frames_and_close_s32
WebRTC_VAD/src/dr_wav.h:5515
↓ 4 callersFunctiondrwav_init_write__internal
WebRTC_VAD/src/dr_wav.h:2323
↓ 4 callersFunctiondrwav_preinit_write
WebRTC_VAD/src/dr_wav.h:2281
↓ 4 callersFunctionextractFeature
(wav_path, vad, frame_length=240, hop_length=240, sr=8000)
GenderClassify/utils.py:73
↓ 4 callersFunctionextractVad
(vad_path)
GenderClassify/utils.py:67
↓ 3 callersFunctionMulRe
WebRTC_AEC/src/aec_core.c:141
↓ 3 callersFunctionNormW32
Return the number of steps a can be left-shifted without overflow, or 0 if a == 0.
WebRTC_AGC/src/agc.c:222
↓ 3 callersFunctionResNet18
()
CommandRecognition/model.py:104
↓ 3 callersFunctionTimeToFrequency
WebRTC_AEC/src/aec_core.c:1676
↓ 3 callersFunctionWebRtcAgc_InitVad
WebRTC_AGC/src/agc.c:859
↓ 3 callersFunctionWebRtcAgc_ProcessVad
WebRTC_AGC/src/agc.c:886
↓ 3 callersFunctionWebRtcSpl_CountLeadingZeros32
Returns the number of leading zero bits in the argument.
WebRTC_VAD/src/spl_inl.h:39
↓ 3 callersFunctionWebRtcSpl_NormW32
Return the number of steps a can be left-shifted without overflow, or 0 if a == 0.
WebRTC_VAD/src/spl_inl.h:65
↓ 3 callersFunctionWebRtcVad_Downsampling
TODO(bjornv): Move this function to vad_filterbank.c. Downsampling filter based on splitting filter and allpass functions.
WebRTC_VAD/src/vad_sp.c:21
↓ 3 callersFunctionWebRtc_available_write
WebRTC_AEC/src/ring_buffer.c:246
↓ 3 callersFunctionWebRtc_rdft
WebRTC_ANR/src/noise_suppression.c:139
↓ 3 callersMethod_LSBExtract
extract LSB from stego wavsignal :param roulette: :param wavsignal: :return: secret message
Watermarking/LSB.py:193
↓ 3 callersFunctionbitrv2
WebRTC_ANR/src/noise_suppression.c:230
↓ 3 callersFunctioncftfsub
WebRTC_ANR/src/noise_suppression.c:328
↓ 3 callersFunctiondisplaySpeech
display waveform of a given speech sample :param sample_name: speech sample name :param fs: sample frequency :return:
SpectralSubtraction/basic_functions.py:106
↓ 3 callersFunctiondownsampleBy2
WebRTC_AGC/src/agc.c:271
↓ 3 callersFunctiondrwav__bswap_samples_f32
WebRTC_VAD/src/dr_wav.h:1365
↓ 3 callersFunctiondrwav__bswap_samples_s32
WebRTC_VAD/src/dr_wav.h:1346
↓ 3 callersFunctiondrwav__chunk_padding_size_riff
WebRTC_VAD/src/dr_wav.h:1573
↓ 3 callersFunctiondrwav__chunk_padding_size_w64
WebRTC_VAD/src/dr_wav.h:1577
↓ 3 callersFunctiondrwav__is_little_endian
WebRTC_AGC/src/dr_wav.h:765
↓ 3 callersFunctiondrwav__read_and_close_f32
WebRTC_AGC/src/dr_wav.h:3239
↓ 3 callersFunctiondrwav__read_and_close_s16
WebRTC_AGC/src/dr_wav.h:3208
↓ 3 callersFunctiondrwav__read_and_close_s32
WebRTC_AGC/src/dr_wav.h:3270
↓ 3 callersFunctiondrwav__read_chunk_header
WebRTC_VAD/src/dr_wav.h:1589
↓ 3 callersFunctiondrwav_init__internal
WebRTC_VAD/src/dr_wav.h:1861
↓ 3 callersFunctiondrwav_init_write
WebRTC_AGC/src/dr_wav.h:1538
↓ 3 callersFunctiondrwav_preinit
WebRTC_VAD/src/dr_wav.h:1841
↓ 3 callersFunctiondrwav_s16_to_f32
WebRTC_AGC/src/dr_wav.h:2803
↓ 3 callersFunctiondrwav_s16_to_f32
WebRTC_VAD/src/dr_wav.h:4963
↓ 3 callersFunctiondrwav_s16_to_s32
WebRTC_AGC/src/dr_wav.h:3135
↓ 3 callersFunctiondrwav_s16_to_s32
WebRTC_VAD/src/dr_wav.h:5351
↓ 3 callersFunctionextractFeature
(wav_path, frame_length=240, hop_length=240, sr=8000)
VoiceActivityDetection/utils.py:77
↓ 3 callersFunctionextractLabel
(label_path)
VoiceActivityDetection/utils.py:71
↓ 2 callersFunctionAllPassFilter
All pass filtering of |data_in|, used before splitting the signal into two frequency bands (low pass vs high pass). Note that |data_in| and |data_out|
WebRTC_VAD/src/vad_filterbank.c:78
↓ 2 callersFunctionBinarySpectrumFix
Computes the binary spectrum by comparing the input |spectrum| with a |threshold_spectrum|. Float and fixed point versions. Inputs: - spectrum
WebRTC_AEC/src/delay_estimator_wrapper.c:61
↓ 2 callersFunctionBinarySpectrumFloat
WebRTC_AEC/src/delay_estimator_wrapper.c:96
↓ 2 callersFunctionBitCount
Counts and returns number of bits of a 32-bit word.
WebRTC_AEC/src/delay_estimator.c:43
↓ 2 callersFunctionDirectInterpolation
(x, L, M)
Resample/Algorithm.py:15
↓ 2 callersFunctionFFT
Transforms the signal from time to frequency domain. Inputs: |time_data| is the signal in the time domain. |time_data_length| is the length of the ana
WebRTC_ANR/src/noise_suppression.c:1553
↓ 2 callersFunctionFeatureParameterExtraction
Extract thresholds for feature parameters. Histograms are computed over some window size (given by self->modelUpdatePars[1]). Thresholds and weights a
WebRTC_ANR/src/noise_suppression.c:970
↓ 2 callersFunctionIncreaseMarking
(x, p, fs, m)
VoiceChange/td_psola.py:162
↓ 2 callersFunctionInitMetrics
WebRTC_AEC/src/aec_core.c:1476
↓ 2 callersFunctionLagrangeInterpolation
(x, w, L, M)
Resample/Algorithm.py:29
↓ 2 callersFunctionMPY48SR
Goertzel/DtmfGenerator.cpp:13
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