/external/chromium_org/third_party/webrtc/common_audio/vad/ |
H A D | vad_sp.h | 27 // - filter_state : Current filter states of the two all-pass filters. The 28 // |filter_state| is updated after all samples have been 35 int32_t* filter_state,
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H A D | vad_filterbank.c | 38 // - filter_state [i/o] : State of the filter. 42 int16_t* filter_state, int16_t* data_out) { 60 tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]); 61 tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2], filter_state[1]); 62 filter_state[1] = filter_state[0]; 63 filter_state[0] = *in_ptr++; 66 tmp32 -= WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[1], filter_state[2]); 67 tmp32 -= WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]); 68 filter_state[ 41 HighPassFilter(const int16_t* data_in, int data_length, int16_t* filter_state, int16_t* data_out) argument 83 AllPassFilter(const int16_t* data_in, int data_length, int16_t filter_coefficient, int16_t* filter_state, int16_t* data_out) argument [all...] |
H A D | vad_sp.c | 29 int32_t* filter_state, 32 int32_t tmp32_1 = filter_state[0]; 33 int32_t tmp32_2 = filter_state[1]; 54 filter_state[0] = tmp32_1; 55 filter_state[1] = tmp32_2; 27 WebRtcVad_Downsampling(const int16_t* signal_in, int16_t* signal_out, int32_t* filter_state, int in_length) argument
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/external/webrtc/src/common_audio/vad/ |
H A D | vad_sp.h | 27 // - filter_state : Current filter states of the two all-pass filters. The 28 // |filter_state| is updated after all samples have been 35 int32_t* filter_state,
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H A D | vad_filterbank.h | 31 * - filter_state : Current state of the filter 35 * - filter_state : Updated state of the filter 40 int16_t* filter_state, 55 * - filter_state : Current state of the filter (Q(-1)) 59 * - filter_state : Updated state of the filter (Q(-1)) 65 int16_t* filter_state,
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H A D | vad_filterbank.c | 38 int16_t* filter_state, 57 tmp32 += (int32_t) WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]); 59 filter_state[1]); // Q14 60 filter_state[1] = filter_state[0]; 61 filter_state[0] = *in_ptr++; 65 filter_state[2]); // Q14 66 tmp32 -= (int32_t) WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]); 67 filter_state[3] = filter_state[ 36 WebRtcVad_HpOutput(int16_t* in_vector, int in_vector_length, int16_t* filter_state, int16_t* out_vector) argument 73 WebRtcVad_Allpass(int16_t* in_vector, int16_t filter_coefficients, int vector_length, int16_t* filter_state, int16_t* out_vector) argument [all...] |
H A D | vad_sp.c | 27 int32_t* filter_state, 30 int32_t tmp32_1 = filter_state[0]; 31 int32_t tmp32_2 = filter_state[1]; 52 filter_state[0] = tmp32_1; 53 filter_state[1] = tmp32_2; 25 WebRtcVad_Downsampling(int16_t* signal_in, int16_t* signal_out, int32_t* filter_state, int in_length) argument
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/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | splitting_filter.c | 41 // - filter_state : Filter state (length 6, Q10). 50 int32_t* filter_state) 78 diff = WebRtcSpl_SubSatW32(in_data[0], filter_state[1]); 80 out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]); 92 filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time 93 filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time 97 diff = WebRtcSpl_SubSatW32(out_data[0], filter_state[3]); 99 in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]); 108 filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time 109 filter_state[ 48 WebRtcSpl_AllPassQMF(int32_t* in_data, int data_length, int32_t* out_data, const uint16_t* filter_coefficients, int32_t* filter_state) argument [all...] |
/external/webrtc/src/common_audio/signal_processing/ |
H A D | splitting_filter.c | 39 // - filter_state : Filter state (length 6, Q10). 48 WebRtc_Word32* filter_state) 75 diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[1]); // = (x[0] - y_1[-1]) 77 out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]); 88 filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time 89 filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time 92 diff = WEBRTC_SPL_SUB_SAT_W32(out_data[0], filter_state[3]); // = (y_1[0] - y_2[-1]) 94 in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]); 102 filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time 103 filter_state[ 46 WebRtcSpl_AllPassQMF(WebRtc_Word32* in_data, const WebRtc_Word16 data_length, WebRtc_Word32* out_data, const WebRtc_UWord16* filter_coefficients, WebRtc_Word32* filter_state) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | pitch_filter.c | 260 static void FilterFrame(const double* in_data, PitchFiltstr* filter_state, argument 276 memcpy(filter_parameters.buffer, filter_state->ubuf, 277 sizeof(filter_state->ubuf)); 278 memcpy(filter_parameters.damper_state, filter_state->ystate, 279 sizeof(filter_state->ystate)); 298 old_lag = *filter_state->oldlagp; 299 old_gain = *filter_state->oldgainp; 339 memcpy(filter_state->ubuf, &filter_parameters.buffer[PITCH_FRAME_LEN], 340 sizeof(filter_state->ubuf)); 341 memcpy(filter_state [all...] |
/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | pitch_filter.c | 260 static void FilterFrame(const double* in_data, PitchFiltstr* filter_state, argument 276 memcpy(filter_parameters.buffer, filter_state->ubuf, 277 sizeof(filter_state->ubuf)); 278 memcpy(filter_parameters.damper_state, filter_state->ystate, 279 sizeof(filter_state->ystate)); 298 old_lag = *filter_state->oldlagp; 299 old_gain = *filter_state->oldgainp; 339 memcpy(filter_state->ubuf, &filter_parameters.buffer[PITCH_FRAME_LEN], 340 sizeof(filter_state->ubuf)); 341 memcpy(filter_state [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | background_noise.h | 95 memset(filter_state, 0, sizeof(filter_state)); 107 int16_t filter_state[kMaxLpcOrder]; member in struct:webrtc::BackgroundNoise::ChannelParameters 124 const int16_t* filter_state,
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H A D | background_noise.cc | 147 return channel_parameters_[channel].filter_state; 154 memcpy(channel_parameters_[channel].filter_state, input, 227 const int16_t* filter_state, 234 memcpy(parameters.filter_state, filter_state, 225 SaveParameters(size_t channel, const int16_t* lpc_coefficients, const int16_t* filter_state, int32_t sample_energy, int32_t residual_energy) argument
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/external/chromium_org/content/browser/renderer_host/input/ |
H A D | mock_input_router_client.h | 42 void set_filter_state(InputEventAckState filter_state) { argument 43 filter_state_ = filter_state;
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/external/openfst/src/include/fst/ |
H A D | state-table.h | 221 filter_state(FilterState::NoState()) {} 224 : state_id1(s1), state_id2(s2), filter_state(f) {} 228 FilterState filter_state; // State of composition filter member in struct:fst::ComposeStateTuple 239 x.filter_state == y.filter_state; 249 t.filter_state.Hash() * kPrime1; 308 tuple.filter_state.Hash() * mult2_;
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H A D | compose.h | 269 filter_->SetState(s1, s2, tuple.filter_state); 365 filter_->SetState(s1, s2, tuple.filter_state);
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/external/chromium_org/third_party/webrtc/common_audio/signal_processing/include/ |
H A D | signal_processing_library.h | 663 int16_t* filter_state, 1502 // - filter_state : Current state (higher part) of the filter. 1503 // - filter_state_length : Length (in samples) of |filter_state|. 1510 // - filter_state : Updated state (upper part) vector.
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/external/webrtc/src/common_audio/signal_processing/include/ |
H A D | signal_processing_library.h | 378 WebRtc_Word16* filter_state, int filter_state_length, 1406 // - filter_state : Current state (higher part) of the filter. 1407 // - filter_state_length : Length (in samples) of |filter_state|. 1414 // - filter_state : Updated state (upper part) vector.
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/external/srec/srec/Recognizer/src/ |
H A D | RecognizerImpl.c | 1851 } filter_state = WORD; local 1870 switch (filter_state) 1874 filter_state = WORD; 1877 filter_state = FRAME; 1892 filter_state = FRAME; 1909 filter_state = WORD;
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