/external/webrtc/webrtc/common_audio/vad/ |
H A D | vad_core.h | 33 int16_t noise_means[kTableSize]; 34 int16_t speech_means[kTableSize]; 35 int16_t noise_stds[kTableSize]; 36 int16_t speech_stds[kTableSize]; 39 int16_t over_hang; // Over Hang 40 int16_t num_of_speech; 42 int16_t index_vector[16 * kNumChannels]; 43 int16_t low_value_vector[16 * kNumChannels]; 45 int16_t mean_value[kNumChannels]; 46 int16_t upper_stat [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | bw_expand.c | 27 int16_t *out, /* (o) the bandwidth expanded lpc coefficients */ 28 int16_t *in, /* (i) the lpc coefficients before bandwidth 30 int16_t *coef, /* (i) the bandwidth expansion factor Q15 */ 31 int16_t length /* (i) the length of lpc coefficient vectors */ 40 out[i] = (int16_t)((coef[i] * in[i] + 16384) >> 15);
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H A D | cb_search.h | 25 int16_t *index, /* (o) Codebook indices */ 26 int16_t *gain_index, /* (o) Gain quantization indices */ 27 int16_t *intarget, /* (i) Target vector for encoding */ 28 int16_t *decResidual,/* (i) Decoded residual for codebook construction */ 31 int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
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H A D | cb_update_best_index.h | 26 int16_t CritNewSh, /* (i) Shift value of above Criteria */ 29 int16_t invEnergyNew, /* (i) Inversed energy new index */ 30 int16_t energyShiftNew, /* (i) Energy shifts of new index */ 32 int16_t *shTotMax, /* (i/o) Shifts of maximum criteria */ 35 int16_t *bestGain); /* (i/o) Gain in Q14 that corresponds
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H A D | defines.h | 121 int16_t lsf[LSF_NSPLIT*LPC_N_MAX]; 122 int16_t cb_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB index */ 123 int16_t gain_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB gain */ 125 int16_t state_first; 126 int16_t idxVec[STATE_SHORT_LEN_30MS]; 127 int16_t firstbits; 135 int16_t mode; 140 int16_t nasub; 142 int16_t lpc_n; 146 int16_t anaMe [all...] |
H A D | augmented_cb_corr.h | 29 int16_t *target, /* (i) Target vector */ 30 int16_t *buffer, /* (i) Memory buffer */ 31 int16_t *interpSamples, /* (i) buffer with
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H A D | refiner.h | 34 int16_t *idata, /* (i) original data buffer */ 38 int16_t *surround, /* (i/o) The contribution from this sequence 40 int16_t gain /* (i) Gain to use for this sequence */
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H A D | xcorr_coef.h | 30 int16_t *target, /* (i) first array */ 31 int16_t *regressor, /* (i) second array */ 35 int16_t step /* (i) +1 or -1 */
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H A D | smooth_out_data.c | 23 int16_t *odata, 24 int16_t *psseq, 25 int16_t *surround, 26 int16_t C) 30 int16_t err; 34 odata[i]= (int16_t)((C * surround[i] + 1024) >> 11);
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H A D | abs_quant.h | 35 int16_t *in, /* (i) vector to encode */ 36 int16_t *weightDenum /* (i) denominator of synthesis filter */
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H A D | decode.h | 29 int16_t *decblock, /* (o) decoded signal block */ 33 int16_t mode /* (i) 0: bad packet, PLC,
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H A D | decode_residual.h | 33 int16_t *decresidual, /* (o) decoded residual frame */ 34 int16_t *syntdenum /* (i) the decoded synthesis filter
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H A D | filtered_cb_vecs.h | 31 int16_t *cbvectors, /* (o) Codebook vector for the higher section */ 32 int16_t *CBmem, /* (i) Codebook memory that is filtered to create a
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H A D | get_cd_vec.h | 23 int16_t *cbvec, /* (o) Constructed codebook vector */ 24 int16_t *mem, /* (i) Codebook buffer */
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | bandwidth_estimator.h | 62 const int16_t frameSize, 69 int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bwest_str, 70 const int16_t Index); 90 int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bwest_str); 93 int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str); 96 int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str); 111 int16_t StreamSize, /* bytes in bitstream */ 112 const int16_t FrameLen, /* ms per frame */ 113 const int16_t BottleNeck, /* bottle neck rate; excl headers (bps) */ 114 const int16_t DelayBuildU [all...] |
H A D | lpc_masking_model.h | 27 void WebRtcIsacfix_GetVars(const int16_t *input, 28 const int16_t *pitchGains_Q12, 30 int16_t *varscale); 32 void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0, 33 int16_t *inHiQ0, 35 int16_t snrQ10, 36 const int16_t *pitchGains_Q12, 38 int16_t *lo_coeffQ15, 39 int16_t *hi_coeffQ15); 44 int16_t* a_polynomia [all...] |
H A D | pitch_lag_tables.c | 69 const int16_t WebRtcIsacfix_kLowerLimitLo[4] = { 73 const int16_t WebRtcIsacfix_kUpperLimitLo[4] = { 84 const int16_t WebRtcIsacfix_kMeanLag2Lo[19] = { 89 const int16_t WebRtcIsacfix_kMeanLag4Lo[9] = { 156 const int16_t WebRtcIsacfix_kLowerLimitMid[4] = { 160 const int16_t WebRtcIsacfix_kUpperLimitMid[4] = { 171 const int16_t WebRtcIsacfix_kMeanLag2Mid[35] = { 180 const int16_t WebRtcIsacfix_kMeanLag4Mid[19] = { 278 const int16_t WebRtcIsacfix_kLowerLimitHi[4] = { 282 const int16_t WebRtcIsacfix_kUpperLimitH [all...] |
H A D | pitch_lag_tables.h | 39 extern const int16_t WebRtcIsacfix_kLowerLimitLo[4]; 40 extern const int16_t WebRtcIsacfix_kUpperLimitLo[4]; 46 extern const int16_t WebRtcIsacfix_kMeanLag2Lo[19]; 47 extern const int16_t WebRtcIsacfix_kMeanLag4Lo[9]; 65 extern const int16_t WebRtcIsacfix_kLowerLimitMid[4]; 66 extern const int16_t WebRtcIsacfix_kUpperLimitMid[4]; 72 extern const int16_t WebRtcIsacfix_kMeanLag2Mid[35]; 73 extern const int16_t WebRtcIsacfix_kMeanLag4Mid[19]; 90 extern const int16_t WebRtcIsacfix_kLowerLimitHi[4]; 91 extern const int16_t WebRtcIsacfix_kUpperLimitH [all...] |
H A D | codec.h | 35 int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16, 39 void WebRtcIsacfix_DecodePlcImpl(int16_t* decoded, 43 int WebRtcIsacfix_EncodeImpl(int16_t* in, 46 int16_t CodingMode); 72 typedef void (*Time2Spec)(int16_t* inre1Q9, 73 int16_t* inre2Q9, 74 int16_t* outre, 75 int16_t* outim); 76 typedef void (*Spec2Time)(int16_t* inreQ7, 77 int16_t* inimQ [all...] |
H A D | structs.h | 34 int16_t full; /* 0 - first byte in memory filled, second empty*/ 47 int16_t full; /* 0 - first byte in memory filled, second empty*/ 55 int16_t DataBufferLoQ0[WINLEN]; 56 int16_t DataBufferHiQ0[WINLEN]; 61 int16_t CorrBufLoQdom[ORDERLO+1]; 62 int16_t CorrBufHiQdom[ORDERHI+1]; 75 int16_t PostStateLoGQ0[ORDERLO+1]; 76 int16_t PostStateHiGQ0[ORDERHI+1]; 95 int16_t INLABUF1_fix[QLOOKAHEAD]; 96 int16_t INLABUF2_fi [all...] |
/external/webrtc/webrtc/modules/audio_processing/agc/legacy/ |
H A D | digital_agc.h | 28 int16_t HPstate; 29 int16_t counter; 30 int16_t logRatio; // log( P(active) / P(inactive) ) (Q10) 31 int16_t meanLongTerm; // Q10 33 int16_t stdLongTerm; // Q10 34 int16_t meanShortTerm; // Q10 36 int16_t stdShortTerm; // Q10 45 int16_t gatePrevious; 46 int16_t agcMode; 55 int32_t WebRtcAgc_InitDigital(DigitalAgc* digitalAgcInst, int16_t agcMod [all...] |
/external/libvpx/libvpx/vpx_dsp/ |
H A D | quantize.h | 23 const int16_t *round_ptr, const int16_t quant_ptr, 25 const int16_t dequant_ptr, uint16_t *eob_ptr); 27 const int16_t *round_ptr, const int16_t quant_ptr, 29 const int16_t dequant_ptr, uint16_t *eob_ptr); 34 const int16_t *round_ptr, const int16_t quant_ptr, 36 const int16_t dequant_ptr, uint16_t *eob_ptr); 39 const int16_t *round_pt [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | dsp_helper.h | 28 static const int16_t kDownsample8kHzTbl[3]; 29 static const int16_t kDownsample16kHzTbl[5]; 30 static const int16_t kDownsample32kHzTbl[7]; 31 static const int16_t kDownsample48kHzTbl[7]; 56 static int RampSignal(const int16_t* input, 60 int16_t* output); 63 static int RampSignal(int16_t* signal, 81 static void PeakDetection(int16_t* data, size_t data_length, 83 size_t* peak_index, int16_t* peak_value); 91 static void ParabolicFit(int16_t* signal_point [all...] |
H A D | random_vector.h | 25 static const int16_t kRandomTable[kRandomTableSize]; 34 void Generate(size_t length, int16_t* output); 36 void IncreaseSeedIncrement(int16_t increase_by); 39 int16_t seed_increment() { return seed_increment_; } 40 void set_seed_increment(int16_t value) { seed_increment_ = value; } 44 int16_t seed_increment_;
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/external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
H A D | g711_interface.h | 41 size_t WebRtcG711_EncodeA(const int16_t* speechIn, 62 size_t WebRtcG711_EncodeU(const int16_t* speechIn, 87 int16_t* decoded, 88 int16_t* speechType); 111 int16_t* decoded, 112 int16_t* speechType); 129 int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
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