/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g722/ |
H A D | g722_interface.c | 19 int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst) 29 int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst) 42 int16_t WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst) 48 int16_t WebRtcG722_Encode(G722EncInst *G722enc_inst, 49 int16_t *speechIn, 50 int16_t len, 51 int16_t *encoded) 59 int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst) 69 int16_t WebRtcG722_DecoderInit(G722DecInst *G722dec_inst) 82 int16_t WebRtcG722_FreeDecode [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | energy_inverse.h | 27 int16_t *energy, /* (i/o) Energy and inverse
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H A D | enh_upsample.h | 30 int16_t *seq1 /* (i) unupsampled sequence */
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H A D | filtered_cb_vecs.c | 29 int16_t *cbvectors, /* (o) Codebook vector for the higher section */ 30 int16_t *CBmem, /* (i) Codebook memory that is filtered to create a 33 int16_t samples /* (i) Number of samples to filter */ 45 (int16_t*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples);
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H A D | get_lsp_poly.h | 44 int16_t *lsp, /* (i) LSP in Q15 */
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H A D | lsf_check.h | 29 int16_t *lsf, /* LSF parameters */
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H A D | interpolate_samples.c | 23 int16_t *interpSamples, /* (o) The interpolated samples */ 24 int16_t *CBmem, /* (i) The CB memory */ 25 int16_t lMem /* (i) Length of the CB memory */ 27 int16_t *ppi, *ppo, i, j, temp1, temp2; 28 int16_t *tmpPtr; 40 *tmpPtr++ = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp2],*ppo, 15) + 41 (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp1], *ppi, 15);
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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)WEBRTC_SPL_RSHIFT_W32( 40 err=(int16_t)WEBRTC_SPL_RSHIFT_W16((psseq[i]-odata[i]), 3);
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H A D | split_vq.c | 29 int16_t *qX, /* (o) the quantized vector in Q13 */ 30 int16_t *index, /* (o) a vector of indexes for all vector 32 int16_t *X, /* (i) the vector to quantize */ 33 int16_t *CB, /* (i) the quantizer codebook in Q13 */ 34 int16_t *dim, /* (i) the dimension of X and qX */ 35 int16_t *cbsize /* (i) the number of vectors in the codebook */ 38 int16_t *qXPtr, *indexPtr, *CBPtr, *XPtr;
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H A D | augmented_cb_corr.c | 24 int16_t *target, /* (i) Target vector */ 25 int16_t *buffer, /* (i) Memory buffer */ 26 int16_t *interpSamples, /* (i) buffer with 31 int16_t low, /* (i) Lag to start from (typically 33 int16_t high, /* (i) Lag to end at (typically 39) */ 34 int16_t scale) /* (i) Scale factor to use for 38 int16_t ilow; 39 int16_t *targetPtr; 41 int16_t *iSPtr=interpSamples; 49 ilow = (int16_t) (lagcoun [all...] |
H A D | sort_sq.c | 26 int16_t *xq, /* (o) the quantized value */ 27 int16_t *index, /* (o) the quantization index */ 28 int16_t x, /* (i) the value to quantize */ 29 const int16_t *cb, /* (i) the quantization codebook */ 30 int16_t cb_size /* (i) the size of the quantization codebook */
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H A D | abs_quant_loop.c | 23 void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN, 24 int16_t *weightDenumIN, int16_t *quantLenIN, 25 int16_t *idxVecIN ) { 27 int16_t index; 30 int16_t tmp16a; 31 int16_t xq; 33 int16_t *syntOut = syntOutIN; 34 int16_t *in_weighte [all...] |
/external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
H A D | get_scaling_square.c | 20 int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector, 24 int16_t nbits = WebRtcSpl_GetSizeInBits(times); 26 int16_t smax = -1; 27 int16_t sabs; 28 int16_t *sptr = in_vector; 29 int16_t t;
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H A D | randomization_functions.c | 26 static const int16_t kRandNTable[] = { 98 int16_t WebRtcSpl_RandU(uint32_t* seed) { 99 return (int16_t)(IncreaseSeed(seed) >> 16); 102 int16_t WebRtcSpl_RandN(uint32_t* seed) { 107 int16_t WebRtcSpl_RandUArray(int16_t* vector, 108 int16_t vector_length,
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H A D | copy_set_operations.c | 29 void WebRtcSpl_MemSetW16(int16_t *ptr, int16_t set_value, int length) 32 int16_t *arrptr = ptr; 51 void WebRtcSpl_MemCpyReversedOrder(int16_t* dest, int16_t* source, int length) 54 int16_t* destPtr = dest; 55 int16_t* sourcePtr = source; 63 void WebRtcSpl_CopyFromEndW16(const int16_t *vector_in, 66 int16_t *vector_out) 72 void WebRtcSpl_ZerosArrayW16(int16_t *vecto [all...] |
H A D | vector_scaling_operations.c | 25 void WebRtcSpl_VectorBitShiftW16(int16_t *res, int16_t length, 26 const int16_t *in, int16_t right_shifts) 46 int16_t vector_length, 48 int16_t right_shifts) 67 void WebRtcSpl_VectorBitShiftW32ToW16(int16_t* out, int length, 86 void WebRtcSpl_ScaleVector(const int16_t *in_vector, int16_t *out_vector, 87 int16_t gai [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_cng.h | 26 explicit ACMCNG(int16_t codec_id); 32 int16_t InternalEncode(uint8_t* bitstream, 33 int16_t* bitstream_len_byte); 35 int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); 41 int16_t InternalCreateEncoder(); 43 int16_t EnableDTX() { 47 int16_t DisableDTX() {
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H A D | acm_g729.h | 26 explicit ACMG729(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 EnableDTX(); 43 int16_t DisableDTX();
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H A D | acm_pcm16b.cc | 26 ACMPCM16B::ACMPCM16B(int16_t /* codec_id */) { return; } 30 int16_t ACMPCM16B::InternalEncode(uint8_t* /* bitstream */, 31 int16_t* /* bitstream_len_byte */) { 35 int16_t ACMPCM16B::InternalInitEncoder( 42 int16_t ACMPCM16B::InternalCreateEncoder() { return -1; } 47 ACMPCM16B::ACMPCM16B(int16_t codec_id) { 54 int16_t ACMPCM16B::InternalEncode(uint8_t* bitstream, 55 int16_t* bitstream_len_byte) { 65 int16_t ACMPCM16B::InternalInitEncoder( 73 int16_t ACMPCM16 [all...] |
H A D | acm_pcm16b.h | 22 explicit ACMPCM16B(int16_t codec_id); 28 int16_t InternalEncode(uint8_t* bitstream, 29 int16_t* bitstream_len_byte) OVERRIDE 32 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 38 int16_t InternalCreateEncoder();
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H A D | acm_pcma.h | 22 explicit ACMPCMA(int16_t codec_id); 28 int16_t InternalEncode(uint8_t* bitstream, 29 int16_t* bitstream_len_byte) OVERRIDE 32 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 37 int16_t InternalCreateEncoder();
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H A D | acm_pcmu.h | 22 explicit ACMPCMU(int16_t codec_id); 28 int16_t InternalEncode(uint8_t* bitstream, 29 int16_t* bitstream_len_byte) OVERRIDE 32 int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params); 38 int16_t InternalCreateEncoder();
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
H A D | lpc_tables.h | 38 extern const int16_t WebRtcIsacfix_kQuantMinGain[12]; 40 extern const int16_t WebRtcIsacfix_kQuantMinShape[108]; 65 extern const int16_t WebRtcIsacfix_kLevelsShapeQ10[1735]; 78 extern const int16_t WebRtcIsacfix_kCodeLenGainQ11[392]; 80 extern const int16_t WebRtcIsacfix_kCodeLenShapeQ11[577]; 83 extern const int16_t WebRtcIsacfix_kT1GainQ15[KLT_NUM_MODELS][4]; 85 extern const int16_t WebRtcIsacfix_kT1ShapeQ15[KLT_NUM_MODELS][324]; 88 extern const int16_t WebRtcIsacfix_kT2GainQ15[KLT_NUM_MODELS][36]; 90 extern const int16_t WebRtcIsacfix_kT2ShapeQ15[KLT_NUM_MODELS][36]; 93 extern const int16_t WebRtcIsacfix_kMeansGainQ [all...] |
H A D | spectrum_ar_model_tables.c | 61 const int16_t WebRtcIsacfix_kRc1Levels[11] = { 66 const int16_t WebRtcIsacfix_kRc2Levels[11] = { 71 const int16_t WebRtcIsacfix_kRc3Levels[11] = { 76 const int16_t WebRtcIsacfix_kRc4Levels[11] = { 81 const int16_t WebRtcIsacfix_kRc5Levels[11] = { 86 const int16_t WebRtcIsacfix_kRc6Levels[11] = { 91 const int16_t WebRtcIsacfix_kRcBound[12] = { 112 const int16_t *WebRtcIsacfix_kRcLevPtr[AR_ORDER] = { 156 const int16_t WebRtcIsacfix_kCos[6][60] = {
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/external/chromium_org/third_party/webrtc/modules/audio_processing/aecm/ |
H A D | aecm_core.h | 30 int16_t real; 31 int16_t imag; 46 int16_t farBuf[FAR_BUF_LEN]; 48 int16_t mult; 61 int16_t nlpFlag; 62 int16_t fixedDelay; 66 int16_t dfaCleanQDomain; 67 int16_t dfaCleanQDomainOld; 68 int16_t dfaNoisyQDomain; 69 int16_t dfaNoisyQDomainOl [all...] |