Searched refs:opus_int16 (Results 151 - 175 of 257) sorted by relevance

1234567891011

/external/libopus/silk/
H A Dtypedef.h45 #define silk_int16_MIN ((opus_int16)0x8000) /* -2^15 = -32768 */
H A Ddecode_core.c41 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 ) ) */
H A Dquant_LTP_gains.c36 opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* I/O (un)quantized LTP gains */
51 const opus_int16 *b_Q14_ptr;
H A DNLSF_del_dec_quant.c37 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 ];
H A Ddec_API.c86 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 DNSQ_del_dec.c66 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 Dtest_opus_padding.c49 opus_int16 *out = malloc(FRAMESIZE*CHANNELS*sizeof(*out));
/external/chromium_org/third_party/opus/src/include/
H A Dopus_custom.h213 * @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,
H A Dopus.h136 * <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 Ddecode_core.c41 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 ) ) */
H A Dquant_LTP_gains.c36 opus_int16 B_Q14[ MAX_NB_SUBFR * LTP_ORDER ], /* I/O (un)quantized LTP gains */
51 const opus_int16 *b_Q14_ptr;
H A DNLSF_del_dec_quant.c37 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 ];
H A Ddec_API.c86 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 DNSQ_del_dec.c66 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 Dschur_FIX.c37 opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */
89 rc_Q15[ k ] = (opus_int16)rc_tmp_Q15;
H A Dfind_LTP_FIX.c40 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 ) );
/external/chromium_org/third_party/opus/src/silk/float/
H A Dfind_LPC_FLP.c39 opus_int16 NLSF_Q15[], /* O NLSFs */
49 opus_int16 NLSF0_Q15[ MAX_LPC_ORDER ];
/external/libopus/include/
H A Dopus_custom.h213 * @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,
/external/libopus/silk/fixed/
H A Dschur_FIX.c37 opus_int16 *rc_Q15, /* O reflection coefficients [order] Q15 */
89 rc_Q15[ k ] = (opus_int16)rc_tmp_Q15;
H A Dfind_LTP_FIX.c40 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 ) );
/external/libopus/silk/float/
H A Dfind_LPC_FLP.c39 opus_int16 NLSF_Q15[], /* O NLSFs */
49 opus_int16 NLSF0_Q15[ MAX_LPC_ORDER ];
/external/chromium_org/third_party/opus/src/src/
H A Dopus.c149 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)
/external/libopus/src/
H A Dopus.c149 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)
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/opus/
H A Dopus_interface.c68 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];
/external/chromium_org/third_party/opus/src/silk/arm/
H A Dmacros_armv5e.h32 /* (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...]

Completed in 354 milliseconds

1234567891011