/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | refl_coef_to_lpc.c | 20 void WebRtcSpl_ReflCoefToLpc(const int16_t *k, int use_order, int16_t *a) 22 int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1]; 23 int16_t *aptr, *aptr2, *anyptr; 24 const int16_t *kptr; 45 + (int16_t)WEBRTC_SPL_MUL_16_16_RSFT((*aptr2), (*kptr), 15);
|
H A D | real_fft.c | 48 const int16_t* real_data_in, 49 int16_t* complex_data_out) { 56 int16_t complex_buffer[2 << kMaxFFTOrder]; 69 memcpy(complex_data_out, complex_buffer, sizeof(int16_t) * (n + 2)); 75 const int16_t* complex_data_in, 76 int16_t* real_data_out) { 82 int16_t complex_buffer[2 << kMaxFFTOrder]; 87 memcpy(complex_buffer, complex_data_in, sizeof(int16_t) * (n + 2)); 116 const int16_t* real_data_in, 117 int16_t* complex_data_ou [all...] |
H A D | filter_ar_fast_q12.c | 16 void WebRtcSpl_FilterARFastQ12(const int16_t* data_in, 17 int16_t* data_out, 18 const int16_t* __restrict coefficients, 40 data_out[i] = (int16_t)((output + 2048) >> 12);
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/cng/ |
H A D | cng_helpfuns.c | 18 void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a) { 19 int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1]; 20 int16_t *aptr, *aptr2, *anyptr; 21 const int16_t *kptr; 39 (int16_t)((((int32_t)(*aptr2--) * (int32_t) * kptr) + 16384) >> 15);
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | lsf_to_poly.c | 25 int16_t *a, /* (o) predictor coefficients (order = 10) in Q12 */ 26 int16_t *lsf /* (i) line spectral frequencies in Q13 */ 31 int16_t *a1ptr, *a2ptr; 33 int16_t lsp[10]; 74 (*a1ptr) = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13); 77 (*a2ptr) = (int16_t)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
|
H A D | energy_inverse.c | 24 int16_t *energy, /* (i/o) Energy and inverse 30 int16_t *energyPtr; 43 (*energyPtr) = (int16_t)WebRtcSpl_DivW32W16(Nom, (*energyPtr));
|
H A D | chebyshev.c | 29 int16_t WebRtcIlbcfix_Chebyshev( 31 int16_t x, /* (i) Value to the Chevyshev polynomial */ 32 int16_t *f /* (i) The coefficients in the polynomial */ 34 int16_t b1_high, b1_low; /* Use the high, low format to increase the accuracy */ 49 b1_high = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16); 50 b1_low = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((int32_t)b1_high),16), 1); 64 b1_high = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16); 65 b1_low = (int16_t)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((int32_t)b1_high),16), 1); 74 /* Handle overflows and set to maximum or minimum int16_t instead */ 80 return((int16_t)WEBRTC_SPL_RSHIFT_W3 [all...] |
H A D | window32_w32.h | 32 int16_t N /* length to process */
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
H A D | pitch_lag_tables.c | 51 const int16_t WebRtcIsac_kQIndexLowerLimitLagLo[4] = { 54 const int16_t WebRtcIsac_kQIndexUpperLimitLagLo[4] = { 125 const int16_t WebRtcIsac_kQIndexLowerLimitLagMid[4] = { 128 const int16_t WebRtcIsac_kQIndexUpperLimitLagMid[4] = { 233 const int16_t WebRtcIsac_kQindexLowerLimitLagHi[4] = { 236 const int16_t WebRtcIsac_kQindexUpperLimitLagHi[4] = {
|
H A D | pitch_lag_tables.h | 39 extern const int16_t WebRtcIsac_kQIndexLowerLimitLagLo[4]; 40 extern const int16_t WebRtcIsac_kQIndexUpperLimitLagLo[4]; 67 extern const int16_t WebRtcIsac_kQIndexLowerLimitLagMid[4]; 68 extern const int16_t WebRtcIsac_kQIndexUpperLimitLagMid[4]; 95 extern const int16_t WebRtcIsac_kQindexLowerLimitLagHi[4]; 96 extern const int16_t WebRtcIsac_kQindexUpperLimitLagHi[4];
|
H A D | lpc_analysis.h | 26 void WebRtcIsac_GetVars(const double *input, const int16_t *pitchGains_Q12, 30 double signal_noise_ratio, const int16_t *pitchGains_Q12, 48 int16_t bandwidth);
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_amr.cc | 24 // int16_t WebRtcAmr_CreateEnc(AMR_encinst_t_** enc_inst); 25 // int16_t WebRtcAmr_CreateDec(AMR_decinst_t_** dec_inst); 26 // int16_t WebRtcAmr_FreeEnc(AMR_encinst_t_* enc_inst); 27 // int16_t WebRtcAmr_FreeDec(AMR_decinst_t_* dec_inst); 28 // int16_t WebRtcAmr_Encode(AMR_encinst_t_* enc_inst, 29 // int16_t* input, 30 // int16_t len, 31 // int16_t*output, 32 // int16_t mode); 33 // int16_t WebRtcAmr_EncoderIni [all...] |
H A D | acm_amrwb.cc | 24 // int16_t WebRtcAmrWb_CreateEnc(AMRWB_encinst_t_** enc_inst); 25 // int16_t WebRtcAmrWb_CreateDec(AMRWB_decinst_t_** dec_inst); 26 // int16_t WebRtcAmrWb_FreeEnc(AMRWB_encinst_t_* enc_inst); 27 // int16_t WebRtcAmrWb_FreeDec(AMRWB_decinst_t_* dec_inst); 28 // int16_t WebRtcAmrWb_Encode(AMRWB_encinst_t_* enc_inst, int16_t* input, 29 // int16_t len, int16_t* output, int16_t mode); 30 // int16_t WebRtcAmrWb_EncoderIni [all...] |
H A D | acm_g722.h | 30 explicit ACMG722(int16_t codec_id); 36 int16_t InternalEncode(uint8_t* bitstream, 37 int16_t* bitstream_len_byte) OVERRIDE 40 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 44 const int16_t* data, 52 int16_t InternalCreateEncoder();
|
H A D | acm_opus.h | 26 explicit ACMOpus(int16_t codec_id); 31 int16_t InternalEncode(uint8_t* bitstream, 32 int16_t* bitstream_len_byte) OVERRIDE 35 int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); 46 int16_t InternalCreateEncoder(); 48 int16_t SetBitRateSafe(const int32_t rate) OVERRIDE
|
H A D | acm_g7291.cc | 28 ACMG729_1::ACMG729_1(int16_t /* codec_id */) 38 int16_t ACMG729_1::InternalEncode(uint8_t* /* bitstream */, 39 int16_t* /* bitstream_len_byte */) { 43 int16_t ACMG729_1::InternalInitEncoder( 50 int16_t ACMG729_1::InternalCreateEncoder() { return -1; } 54 int16_t ACMG729_1::SetBitRateSafe(const int32_t /*rate*/) { return -1; } 60 ACMG729_1::ACMG729_1(int16_t codec_id) 79 int16_t ACMG729_1::InternalEncode(uint8_t* bitstream, 80 int16_t* bitstream_len_byte) { 82 int16_t num_encoded_sample [all...] |
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | lattice_c.c | 24 void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, // Input samples 25 int16_t* ar_f_Q0, // Input samples 26 int16_t* cth_Q15, // Filter coefficients 27 int16_t* sth_Q15, // Filter coefficients 28 int16_t order_coef) { // order of the filter 33 int16_t tmpAR = 0;
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | pitch_filter.c | 22 static const int16_t kDivFactor = 6553; 26 static const int16_t kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = { 40 int16_t gain, 42 int16_t sign, 43 int16_t* inputState, 44 int16_t* outputBuf2, 45 const int16_t* coefficient, 46 int16_t* inputBuf, 47 int16_t* outputBuf, 51 int16_t qDomai [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | random_vector.cc | 15 const int16_t RandomVector::kRandomTable[RandomVector::kRandomTableSize] = { 45 void RandomVector::Generate(size_t length, int16_t* output) { 53 void RandomVector::IncreaseSeedIncrement(int16_t increase_by) {
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | input_audio_file.h | 35 bool Read(size_t samples, int16_t* destination); 42 static void DuplicateInterleaved(const int16_t* source, size_t samples, 43 size_t channels, int16_t* destination);
|
/external/libvpx/libvpx/vp8/common/arm/neon/ |
H A D | dequantizeb_neon.c | 14 int16_t *Q, 15 int16_t *DQC, 16 int16_t *DQ) {
|
/external/chromium_org/sdch/open-vcdiff/vsprojects/ |
H A D | stdint.h | 22 typedef __int16 int16_t; typedef
|
/external/chromium_org/third_party/brotli/src/woff2/ |
H A D | glyph.h | 36 int16_t x_min; 37 int16_t x_max; 38 int16_t y_min; 39 int16_t y_max;
|
/external/chromium_org/third_party/libvpx/source/libvpx/vp9/encoder/ |
H A D | vp9_block.h | 30 DECLARE_ALIGNED(16, int16_t, src_diff[64 * 64]); 37 int16_t *quant_fp; 38 int16_t *round_fp; 39 int16_t *quant; 40 int16_t *quant_shift; 41 int16_t *zbin; 42 int16_t *round; 46 int16_t zbin_extra; 122 void (*fwd_txm4x4)(const int16_t *input, tran_low_t *output, int stride);
|
H A D | vp9_tokenize.h | 28 int16_t token; 29 int16_t extra; 34 int16_t extra; 50 extern const int16_t *vp9_dct_value_cost_ptr;
|