/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
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 | structs.h | 33 int16_t full; /* 0 - first byte in memory filled, second empty*/ 46 int16_t full; /* 0 - first byte in memory filled, second empty*/ 54 int16_t DataBufferLoQ0[WINLEN]; 55 int16_t DataBufferHiQ0[WINLEN]; 60 int16_t CorrBufLoQdom[ORDERLO+1]; 61 int16_t CorrBufHiQdom[ORDERHI+1]; 74 int16_t PostStateLoGQ0[ORDERLO+1]; 75 int16_t PostStateHiGQ0[ORDERHI+1]; 94 int16_t INLABUF1_fix[QLOOKAHEAD]; 95 int16_t INLABUF2_fi [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_amr.h | 28 explicit ACMAMR(int16_t codec_id); 34 int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); 36 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 38 int16_t SetAMREncoderPackingFormat(const ACMAMRPackingFormat packing_format); 42 int16_t SetAMRDecoderPackingFormat(const ACMAMRPackingFormat packing_format); 49 int16_t InternalCreateEncoder(); 51 int16_t SetBitRateSafe(const int32_t rate); 53 int16_t EnableDTX(); 55 int16_t DisableDT [all...] |
H A D | acm_amrwb.h | 26 explicit ACMAMRwb(int16_t codec_id); 32 int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); 34 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 36 int16_t SetAMRwbEncoderPackingFormat( 41 int16_t SetAMRwbDecoderPackingFormat( 49 int16_t InternalCreateEncoder(); 51 int16_t SetBitRateSafe(const int32_t rate); 53 int16_t EnableDTX(); 55 int16_t DisableDT [all...] |
H A D | acm_g7291.h | 26 explicit ACMG729_1(int16_t codec_id); 32 int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); 34 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 39 int16_t InternalCreateEncoder(); 41 int16_t SetBitRateSafe(const int32_t rate); 46 int16_t flag_8khz_; 47 int16_t flag_g729_mode_;
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | filterbank_internal.h | 26 void WebRtcIsacfix_HighpassFilterFixDec32(int16_t *io, 27 int16_t len, 28 const int16_t *coefficient,
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/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | cross_correlation.c | 15 const int16_t* seq1, 16 const int16_t* seq2, 17 int16_t dim_seq, 18 int16_t dim_cross_correlation, 19 int16_t right_shifts, 20 int16_t step_seq2) {
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H A D | filter_ma_fast_q12.c | 20 void WebRtcSpl_FilterMAFastQ12(int16_t* in_ptr, 21 int16_t* out_ptr, 22 int16_t* B, 23 int16_t B_length, 24 int16_t length) 30 const int16_t* b_ptr = &B[0]; 31 const int16_t* x_ptr = &in_ptr[i]; 46 *out_ptr++ = (int16_t)((o + (int32_t)2048) >> 12);
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H A D | vector_scaling_operations_mips.c | 19 int WebRtcSpl_ScaleAndAddVectorsWithRound_mips(const int16_t* in_vector1, 20 int16_t in_vector1_scale, 21 const int16_t* in_vector2, 22 int16_t in_vector2_scale, 24 int16_t* out_vector, 26 int16_t r0 = 0, r1 = 0; 27 int16_t *in1 = (int16_t*)in_vector1; 28 int16_t *in2 = (int16_t*)in_vector [all...] |
H A D | sqrt_of_one_minus_x_squared.c | 20 void WebRtcSpl_SqrtOfOneMinusXSquared(int16_t *xQ15, int vector_length, 21 int16_t *yQ15) 25 int16_t tmp; 33 yQ15[m] = (int16_t)sq;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | comp_corr.c | 29 int16_t *buffer, /* (i) signal buffer */ 30 int16_t lag, /* (i) pitch lag */ 31 int16_t bLen, /* (i) length of buffer */ 32 int16_t sRange, /* (i) correlation search length */ 33 int16_t scale /* (i) number of rightshifts to use */ 35 int16_t *w16ptr;
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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 | lsf_to_lsp.c | 27 int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */ 28 int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */ 29 int16_t m /* (i) number of coefficients */ 31 int16_t i, k; 32 int16_t diff; /* difference, which is used for the 34 int16_t freq; /* normalized frequency in Q15 (0..1) */ 39 freq = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(lsf[i], 20861, 15); 57 lsp[i] = WebRtcIlbcfix_kCos[k]+(int16_t)(WEBRTC_SPL_RSHIFT_W32(tmpW32, 12));
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H A D | get_sync_seq.c | 29 int16_t *idata, /* (i) original data */ 30 int16_t idatal, /* (i) dimension of data */ 31 int16_t centerStartPos, /* (i) where current block starts */ 32 int16_t *period, /* (i) rough-pitch-period array (Q-2) */ 33 int16_t *plocs, /* (i) where periods of period array are taken (Q-2) */ 34 int16_t periodl, /* (i) dimension period array */ 35 int16_t hl, /* (i) 2*hl+1 is the number of sequences */ 36 int16_t *surround /* (i/o) The contribution from this sequence 39 int16_t i,centerEndPos,q; 41 int16_t lagBloc [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
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/chromium_org/third_party/webrtc/voice_engine/ |
H A D | utility.h | 32 PushResampler<int16_t>* resampler, 44 void DownConvertToCodecFormat(const int16_t* src_data, 50 int16_t* mono_buffer, 51 PushResampler<int16_t>* resampler, 54 void MixWithSat(int16_t target[], 56 const int16_t source[],
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H A D | dtmf_inband.h | 44 int Get10msTone(int16_t output[320], uint16_t& outputSizeInSamples); 52 int16_t DtmfFix_generate(int16_t* decoded, 53 int16_t value, 54 int16_t volume, 55 int16_t frameLen, 56 int16_t fs); 63 int16_t DtmfFix_generateSignal(int16_t a1_times2, 64 int16_t a2_times [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/pcm16b/ |
H A D | pcm16b.c | 26 /* Encoder with int16_t Output */ 27 int16_t WebRtcPcm16b_EncodeW16(const int16_t* speechIn16b, 28 int16_t length_samples, 29 int16_t* speechOut16b) 32 memcpy(speechOut16b, speechIn16b, length_samples * sizeof(int16_t)); 44 int16_t WebRtcPcm16b_Encode(int16_t *speech16b, 45 int16_t len, 48 int16_t sample [all...] |
/external/chromium_org/third_party/ots/src/ |
H A D | metrics.h | 17 int16_t ascent; 18 int16_t descent; 19 int16_t linegap; 21 int16_t min_sb1; 22 int16_t min_sb2; 23 int16_t max_extent; 24 int16_t caret_slope_rise; 25 int16_t caret_slope_run; 26 int16_t caret_offset; 31 std::vector<std::pair<uint16_t, int16_t> > entrie [all...] |
/external/chromium_org/third_party/webrtc/common_audio/vad/ |
H A D | vad_gmm.c | 17 static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12. 30 int32_t WebRtcVad_GaussianProbability(int16_t input, 31 int16_t mean, 32 int16_t std, 33 int16_t* delta) { 34 int16_t tmp16, inv_std, inv_std2, exp_value = 0; 41 inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std); 46 inv_std2 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2); 48 // |inv_std2| = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(|inv_std|, |inv_std|, 6); 57 *delta = (int16_t) WEBRTC_SPL_MUL_16_16_RSF [all...] |
/external/webrtc/src/common_audio/vad/ |
H A D | vad_gmm.c | 17 static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12. 30 int32_t WebRtcVad_GaussianProbability(int16_t input, 31 int16_t mean, 32 int16_t std, 33 int16_t* delta) { 34 int16_t tmp16, inv_std, inv_std2, exp_value = 0; 41 inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std); 46 inv_std2 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2); 48 // |inv_std2| = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(|inv_std|, |inv_std|, 6); 57 *delta = (int16_t) WEBRTC_SPL_MUL_16_16_RSF [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/agc/include/ |
H A D | gain_control.h | 42 int16_t targetLevelDbfs; // default 3 (-3 dBOv) 43 int16_t compressionGaindB; // default 9 dB 68 const int16_t* inFar, 69 int16_t samples); 95 int16_t* inMic, 96 int16_t* inMic_H, 97 int16_t samples); 126 int16_t* inMic, 127 int16_t* inMic_H, 128 int16_t sample [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_processing/ns/ |
H A D | nsx_core.h | 25 const int16_t* window; 26 int16_t analysisBuffer[ANAL_BLOCKL_MAX]; 27 int16_t synthesisBuffer[ANAL_BLOCKL_MAX]; 31 const int16_t* factor2Table; 32 int16_t noiseEstLogQuantile[SIMULT* HALF_ANAL_BLOCKL]; 33 int16_t noiseEstDensity[SIMULT* HALF_ANAL_BLOCKL]; 34 int16_t noiseEstCounter[SIMULT]; 35 int16_t noiseEstQuantile[HALF_ANAL_BLOCKL]; 51 int16_t weightLogLrt; 55 int16_t weightSpecDif [all...] |