| /external/openfst/src/include/fst/ |
| compose-filter.h | 36 // class FilterState {
39 // FilterState();
40 // FilterState(const FilterState &f);
42 // static const FilterState NoState();
46 // bool operator==(const FilterState &f) const;
48 // bool operator!=(const FilterState &f) const;
50 // FilterState& operator=(const FilterState& f);
156 // ComposeFilter::FilterState. The basic filters handle correc
[all...] |
| lookahead-filter.h | 204 typedef typename F::FilterState FilterState;
240 FilterState Start() const {
244 void SetState(StateId s1, StateId s2, const FilterState &f) {
248 FilterState FilterArc(Arc *arc1, Arc *arc2) const {
251 const FilterState &f = filter_.FilterArc(arc1, arc2);
252 if (f == FilterState::NoState())
253 return FilterState::NoState();
294 FilterState LookAheadFilterArc(Arc *arca, Arc *arcb,
295 const FilterState &f) const
[all...] |
| state-table.h | 196 // typedef F FilterState;
217 typedef F FilterState;
221 filter_state(FilterState::NoState()) {}
223 ComposeStateTuple(StateId s1, StateId s2, const FilterState &f)
228 FilterState filter_state; // State of composition filter
273 typedef F FilterState;
294 typedef F FilterState;
322 typedef F FilterState;
334 typedef F FilterState;
352 typedef F FilterState;
[all...] |
| compose.h | 51 class T = GenericComposeStateTable<A, typename F::FilterState> >
82 typename F::FilterState> >
211 typedef typename F::FilterState FilterState;
219 typedef ComposeStateTuple<StateId, FilterState> StateTuple;
318 const FilterState &f = filter_->FilterArc(&arcb, &arca);
319 if (f != FilterState::NoState())
322 const FilterState &f = filter_->FilterArc(&arca, &arcb);
323 if (f != FilterState::NoState())
332 const FilterState &f)
[all...] |
| intersect.h | 37 class T = GenericComposeStateTable<A, typename F::FilterState> >
|
| difference.h | 38 class T = GenericComposeStateTable<A, typename F::FilterState> >
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/ |
| high_pass_filter_impl.cc | 29 struct FilterState {
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;
|
| /external/webrtc/src/modules/audio_processing/ |
| high_pass_filter_impl.cc | 30 struct FilterState {
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;
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| filters.c | 69 int32_t *FilterState) //Q16
78 b = WebRtcSpl_AddSatW32(a, FilterState[j]); //Q16+Q16=Q16
82 FilterState[j] = WebRtcSpl_AddSatW32(
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| 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(
|
| 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
|
| /external/openfst/src/include/fst/extensions/pdt/ |
| compose.h | 287 typedef typename F::FilterState FilterState1;
291 typedef PairFilterState<FilterState1, FilterState2> FilterState;
302 f_(FilterState::NoState()),
324 f_(FilterState::NoState()),
328 FilterState Start() const {
329 return FilterState(filter_.Start(), FilterState2(0));
332 void SetState(StateId s1, StateId s2, const FilterState &f) {
352 FilterState FilterArc(Arc *arc1, Arc *arc2) const {
356 return FilterState::NoState();
373 return FilterState(f1, f2)
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| background_noise.h | 57 const int16_t* FilterState(size_t channel) const;
|
| background_noise.cc | 145 const int16_t* BackgroundNoise::FilterState(size_t channel) const {
|
| expand.cc | [all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| filter_functions.c | 202 double *FilterState)
211 InOut[n] = FilterState[j] + APSectionFactors[j]*temp;
212 FilterState[j] = -APSectionFactors[j]*InOut[n] + temp;
|
| filterbanks.c | 31 float *FilterState)
37 temp = FilterState[j] + APSectionFactors[j] * InOut[n];
38 FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
| filter_functions.c | 201 double *FilterState)
210 InOut[n] = FilterState[j] + APSectionFactors[j]*temp;
211 FilterState[j] = -APSectionFactors[j]*InOut[n] + temp;
|
| filterbanks.c | 31 float *FilterState)
37 temp = FilterState[j] + APSectionFactors[j] * InOut[n];
38 FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
|
