Searched defs:FilterState (Results 1 - 9 of 9) sorted by relevance
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | filters.c | 69 int32_t *FilterState) //Q16 78 b = WEBRTC_SPL_ADD_SAT_W32(a, FilterState[j]); //Q16+Q16=Q16 82 FilterState[j] = WEBRTC_SPL_ADD_SAT_W32( 66 AllpassFilterForDec32(int16_t *InOut16, const int32_t *APSectionFactors, int16_t lengthInOut, int32_t *FilterState) argument
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | filters.c | 75 WebRtc_Word32 *FilterState) //Q16 84 b = WEBRTC_SPL_ADD_SAT_W32(a, FilterState[j]); //Q16+Q16=Q16 88 FilterState[j] = WEBRTC_SPL_ADD_SAT_W32( 72 AllpassFilterForDec32(WebRtc_Word16 *InOut16, const WebRtc_Word32 *APSectionFactors, WebRtc_Word16 lengthInOut, WebRtc_Word32 *FilterState) argument
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H A D | filterbanks.c | 31 WebRtc_Word32 *FilterState) //Q16 42 b = WEBRTC_SPL_ADD_SAT_W32(a, FilterState[j]); //Q16+Q16=Q16 44 FilterState[j] = WEBRTC_SPL_ADD_SAT_W32(WEBRTC_SPL_LSHIFT_W32(a,1), WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)InOut16[n],16)); // Q15<<1 + Q0<<16 = Q16 + Q16 = Q16 27 AllpassFilter2FixDec16(WebRtc_Word16 *InOut16, const WebRtc_Word16 *APSectionFactors, WebRtc_Word16 lengthInOut, WebRtc_Word16 NumberOfSections, WebRtc_Word32 *FilterState) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | filterbanks.c | 31 float *FilterState) 37 temp = FilterState[j] + APSectionFactors[j] * InOut[n]; 38 FilterState[j] = -APSectionFactors[j] * temp + InOut[n]; 29 WebRtcIsac_AllPassFilter2Float(float *InOut, const float *APSectionFactors, int lengthInOut, int NumberOfSections, float *FilterState) argument
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H A D | filter_functions.c | 201 double *FilterState) 210 InOut[n] = FilterState[j] + APSectionFactors[j]*temp; 211 FilterState[j] = -APSectionFactors[j]*InOut[n] + temp; 198 WebRtcIsac_AllpassFilterForDec(double *InOut, const double *APSectionFactors, int lengthInOut, double *FilterState) argument
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/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
H A D | high_pass_filter_impl.cc | 29 struct FilterState { struct in namespace:webrtc::__anon15206 35 int InitializeFilter(FilterState* hpf, int sample_rate_hz) { 50 int Filter(FilterState* hpf, int16_t* data, int length) { 105 typedef FilterState Handle; 148 return new FilterState;
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | filterbanks.c | 31 float *FilterState) 37 temp = FilterState[j] + APSectionFactors[j] * InOut[n]; 38 FilterState[j] = -APSectionFactors[j] * temp + InOut[n]; 29 WebRtcIsac_AllPassFilter2Float(float *InOut, const float *APSectionFactors, int lengthInOut, int NumberOfSections, float *FilterState) argument
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H A D | filter_functions.c | 201 double *FilterState) 210 InOut[n] = FilterState[j] + APSectionFactors[j]*temp; 211 FilterState[j] = -APSectionFactors[j]*InOut[n] + temp; 198 WebRtcIsac_AllpassFilterForDec(double *InOut, const double *APSectionFactors, int lengthInOut, double *FilterState) argument
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/external/webrtc/src/modules/audio_processing/ |
H A D | high_pass_filter_impl.cc | 30 struct FilterState { struct in namespace:webrtc::__anon32617 36 int InitializeFilter(FilterState* hpf, int sample_rate_hz) { 51 int Filter(FilterState* hpf, WebRtc_Word16* data, int length) { 106 typedef FilterState Handle; 154 return new FilterState;
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