/external/libopus/silk/ |
H A D | typedef.h | 45 #define silk_int16_MIN ((opus_int16)0x8000) /* -2^15 = -32768 */
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H A D | decode_core.c | 41 opus_int16 xq[], /* O Decoded speech */ 46 opus_int16 *A_Q12, *B_Q14, *pxq, A_Q12_tmp[ MAX_LPC_ORDER ]; 47 VARDECL( opus_int16, sLTP ); 57 ALLOC( sLTP, psDec->ltp_mem_length, opus_int16 ); 101 silk_memcpy( A_Q12_tmp, A_Q12, psDec->LPC_order * sizeof( opus_int16 ) ); 128 silk_memset( B_Q14, 0, LTP_ORDER * sizeof( opus_int16 ) ); 146 silk_memcpy( &psDec->outBuf[ psDec->ltp_mem_length ], xq, 2 * psDec->subfr_length * sizeof( opus_int16 ) ); 224 pxq[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( sLPC_Q14[ MAX_LPC_ORDER + i ], Gain_Q10 ), 8 ) ); 227 /* DEBUG_STORE_DATA( dec.pcm, pxq, psDec->subfr_length * sizeof( opus_int16 ) ) */
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H A D | quant_LTP_gains.c | 36 opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* I/O (un)quantized LTP gains */ 51 const opus_int16 *b_Q14_ptr;
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H A D | NLSF_del_dec_quant.c | 37 const opus_int16 x_Q10[], /* I Input [ order ] */ 38 const opus_int16 w_Q5[], /* I Weights [ order ] */ 40 const opus_int16 ec_ix[], /* I Indices to entropy coding tables [ order ] */ 43 const opus_int16 inv_quant_step_size_Q6, /* I Inverse quantization step size */ 45 const opus_int16 order /* I Number of input values */ 53 opus_int16 prev_out_Q10[ 2 * NLSF_QUANT_DEL_DEC_STATES ];
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H A D | dec_API.c | 86 opus_int16 *samplesOut, /* O Decoded output speech vector */ 92 opus_int16 *samplesOut1_tmp[ 2 ]; 93 VARDECL( opus_int16, samplesOut1_tmp_storage ); 94 VARDECL( opus_int16, samplesOut2_tmp ); 96 opus_int16 *resample_out_ptr; 257 opus_int16 ); 289 silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) ); 299 silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) ); 300 silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) ); 308 decControl->nChannelsAPI == 2 ? *nSamplesOut : ALLOC_NONE, opus_int16 ); [all...] |
H A D | NSQ_del_dec.c | 66 const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ 86 opus_int16 xq[], /* O */ 89 const opus_int16 a_Q12[], /* I Short term prediction coefs */ 90 const opus_int16 b_Q14[], /* I Long term prediction coefs */ 91 const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ 115 const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ 116 const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ 117 const opus_int16 AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ 129 const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; 130 opus_int16 *px [all...] |
/external/libopus/tests/ |
H A D | test_opus_padding.c | 49 opus_int16 *out = malloc(FRAMESIZE*CHANNELS*sizeof(*out));
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/external/chromium_org/third_party/opus/src/include/ |
H A D | opus_custom.h | 213 * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian). 226 const opus_int16 *pcm, 315 * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length 316 * is frame_size*channels*sizeof(opus_int16) 324 opus_int16 *pcm,
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H A D | opus.h | 136 * <li>audio_frame is the audio data in opus_int16 (or float for opus_encode_float())</li> 237 * @param [in] pcm <tt>opus_int16*</tt>: Input signal (interleaved if 2 channels). length is frame_size*channels*sizeof(opus_int16) 265 const opus_int16 *pcm, 373 * @li decoded is the decoded audio data in opus_int16 (or float for opus_decode_float()) 450 * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length 451 * is frame_size*channels*sizeof(opus_int16) 466 opus_int16 *pcm, 523 * @param [out] size <tt>opus_int16[48]</tt> sizes of the encapsulated frames 532 opus_int16 siz [all...] |
/external/chromium_org/third_party/opus/src/silk/ |
H A D | decode_core.c | 41 opus_int16 xq[], /* O Decoded speech */ 46 opus_int16 *A_Q12, *B_Q14, *pxq, A_Q12_tmp[ MAX_LPC_ORDER ]; 47 VARDECL( opus_int16, sLTP ); 57 ALLOC( sLTP, psDec->ltp_mem_length, opus_int16 ); 101 silk_memcpy( A_Q12_tmp, A_Q12, psDec->LPC_order * sizeof( opus_int16 ) ); 128 silk_memset( B_Q14, 0, LTP_ORDER * sizeof( opus_int16 ) ); 146 silk_memcpy( &psDec->outBuf[ psDec->ltp_mem_length ], xq, 2 * psDec->subfr_length * sizeof( opus_int16 ) ); 224 pxq[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( silk_SMULWW( sLPC_Q14[ MAX_LPC_ORDER + i ], Gain_Q10 ), 8 ) ); 227 /* DEBUG_STORE_DATA( dec.pcm, pxq, psDec->subfr_length * sizeof( opus_int16 ) ) */
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H A D | quant_LTP_gains.c | 36 opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* I/O (un)quantized LTP gains */ 51 const opus_int16 *b_Q14_ptr;
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H A D | NLSF_del_dec_quant.c | 37 const opus_int16 x_Q10[], /* I Input [ order ] */ 38 const opus_int16 w_Q5[], /* I Weights [ order ] */ 40 const opus_int16 ec_ix[], /* I Indices to entropy coding tables [ order ] */ 43 const opus_int16 inv_quant_step_size_Q6, /* I Inverse quantization step size */ 45 const opus_int16 order /* I Number of input values */ 53 opus_int16 prev_out_Q10[ 2 * NLSF_QUANT_DEL_DEC_STATES ];
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H A D | dec_API.c | 86 opus_int16 *samplesOut, /* O Decoded output speech vector */ 92 opus_int16 *samplesOut1_tmp[ 2 ]; 93 VARDECL( opus_int16, samplesOut1_tmp_storage ); 94 VARDECL( opus_int16, samplesOut2_tmp ); 96 opus_int16 *resample_out_ptr; 257 opus_int16 ); 289 silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) ); 299 silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) ); 300 silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) ); 308 decControl->nChannelsAPI == 2 ? *nSamplesOut : ALLOC_NONE, opus_int16 ); [all...] |
H A D | NSQ_del_dec.c | 66 const opus_int16 sLTP[], /* I Re-whitened LTP state in Q0 */ 86 opus_int16 xq[], /* O */ 89 const opus_int16 a_Q12[], /* I Short term prediction coefs */ 90 const opus_int16 b_Q14[], /* I Long term prediction coefs */ 91 const opus_int16 AR_shp_Q13[], /* I Noise shaping coefs */ 115 const opus_int16 PredCoef_Q12[ 2 * MAX_LPC_ORDER ], /* I Short term prediction coefs */ 116 const opus_int16 LTPCoef_Q14[ LTP_ORDER * MAX_NB_SUBFR ], /* I Long term prediction coefs */ 117 const opus_int16 AR2_Q13[ MAX_NB_SUBFR * MAX_SHAPE_LPC_ORDER ], /* I Noise shaping coefs */ 129 const opus_int16 *A_Q12, *B_Q14, *AR_shp_Q13; 130 opus_int16 *px [all...] |
/external/chromium_org/third_party/opus/src/silk/fixed/ |
H A D | schur_FIX.c | 37 opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */ 89 rc_Q15[ k ] = (opus_int16)rc_tmp_Q15;
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H A D | find_LTP_FIX.c | 40 opus_int16 LTP_coefs_Q14[ LTP_ORDER ] 44 opus_int16 b_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* O LTP coefs */ 47 const opus_int16 r_lpc[], /* I residual signal after LPC signal + state for first 10 ms */ 57 const opus_int16 *r_ptr, *lag_ptr; 58 opus_int16 *b_Q14_ptr; 236 opus_int16 LTP_coefs_Q14[ LTP_ORDER ] 242 LTP_coefs_Q14[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( LTP_coefs_Q16[ i ], 2 ) );
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/external/chromium_org/third_party/opus/src/silk/float/ |
H A D | find_LPC_FLP.c | 39 opus_int16 NLSF_Q15[], /* O NLSFs */ 49 opus_int16 NLSF0_Q15[ MAX_LPC_ORDER ];
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/external/libopus/include/ |
H A D | opus_custom.h | 213 * @param [in] pcm <tt>opus_int16*</tt>: PCM audio in signed 16-bit format (native endian). 226 const opus_int16 *pcm, 315 * @param [out] pcm <tt>opus_int16*</tt>: Output signal (interleaved if 2 channels). length 316 * is frame_size*channels*sizeof(opus_int16) 324 opus_int16 *pcm,
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/external/libopus/silk/fixed/ |
H A D | schur_FIX.c | 37 opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */ 89 rc_Q15[ k ] = (opus_int16)rc_tmp_Q15;
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H A D | find_LTP_FIX.c | 40 opus_int16 LTP_coefs_Q14[ LTP_ORDER ] 44 opus_int16 b_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* O LTP coefs */ 47 const opus_int16 r_lpc[], /* I residual signal after LPC signal + state for first 10 ms */ 57 const opus_int16 *r_ptr, *lag_ptr; 58 opus_int16 *b_Q14_ptr; 236 opus_int16 LTP_coefs_Q14[ LTP_ORDER ] 242 LTP_coefs_Q14[ i ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( LTP_coefs_Q16[ i ], 2 ) );
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/external/libopus/silk/float/ |
H A D | find_LPC_FLP.c | 39 opus_int16 NLSF_Q15[], /* O NLSFs */ 49 opus_int16 NLSF0_Q15[ MAX_LPC_ORDER ];
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/external/chromium_org/third_party/opus/src/src/ |
H A D | opus.c | 149 static int parse_size(const unsigned char *data, opus_int32 len, opus_int16 *size) 171 const unsigned char *frames[48], opus_int16 size[48], 208 size[0] = (opus_int16)last_size; 272 size[i] = (opus_int16)last_size; 300 size[count-1] = (opus_int16)last_size; 324 opus_int16 size[48], int *payload_offset)
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/external/libopus/src/ |
H A D | opus.c | 149 static int parse_size(const unsigned char *data, opus_int32 len, opus_int16 *size) 171 const unsigned char *frames[48], opus_int16 size[48], 208 size[0] = (opus_int16)last_size; 272 size[i] = (opus_int16)last_size; 300 size[count-1] = (opus_int16)last_size; 324 opus_int16 size[48], int *payload_offset)
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/ |
H A D | opus_interface.c | 68 opus_int16* audio = (opus_int16*) audio_in; 229 opus_int16* audio = (opus_int16*) decoded; 246 opus_int16* audio = (opus_int16*) decoded; 497 opus_int16 frame_sizes[48];
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/external/chromium_org/third_party/opus/src/silk/arm/ |
H A D | macros_armv5e.h | 32 /* (a32 * (opus_int32)((opus_int16)(b32))) >> 16 output have to be 32bit int */ 34 static OPUS_INLINE opus_int32 silk_SMULWB_armv5e(opus_int32 a, opus_int16 b) 47 /* a32 + (b32 * (opus_int32)((opus_int16)(c32))) >> 16 output have to be 32bit int */ 50 opus_int16 c) 94 /* (opus_int32)((opus_int16)(a3))) * (opus_int32)((opus_int16)(b32)) output have to be 32bit int */ 109 /* a32 + (opus_int32)((opus_int16)(b32)) * (opus_int32)((opus_int16)(c32)) output have to be 32bit int */ 125 /* (opus_int32)((opus_int16)(a32)) * (b32 >> 16) */ 140 /* a32 + (opus_int32)((opus_int16)(b3 [all...] |