diff options
Diffstat (limited to 'src/libffmpeg/libavcodec/g726.c')
-rw-r--r-- | src/libffmpeg/libavcodec/g726.c | 429 |
1 files changed, 0 insertions, 429 deletions
diff --git a/src/libffmpeg/libavcodec/g726.c b/src/libffmpeg/libavcodec/g726.c deleted file mode 100644 index c509292b6..000000000 --- a/src/libffmpeg/libavcodec/g726.c +++ /dev/null @@ -1,429 +0,0 @@ -/* - * G.726 ADPCM audio codec - * Copyright (c) 2004 Roman Shaposhnik. - * - * This is a very straightforward rendition of the G.726 - * Section 4 "Computational Details". - * - * This file is part of FFmpeg. - * - * FFmpeg is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * FFmpeg is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ -#include <limits.h> -#include "avcodec.h" -#include "common.h" -#include "bitstream.h" - -/** - * G.726 11bit float. - * G.726 Standard uses rather odd 11bit floating point arithmentic for - * numerous occasions. It's a mistery to me why they did it this way - * instead of simply using 32bit integer arithmetic. - */ -typedef struct Float11 { - int sign; /**< 1bit sign */ - int exp; /**< 4bit exponent */ - int mant; /**< 6bit mantissa */ -} Float11; - -static inline Float11* i2f(int16_t i, Float11* f) -{ - f->sign = (i < 0); - if (f->sign) - i = -i; - f->exp = av_log2_16bit(i) + !!i; - f->mant = i? (i<<6) >> f->exp : 1<<5; - return f; -} - -static inline int16_t mult(Float11* f1, Float11* f2) -{ - int res, exp; - - exp = f1->exp + f2->exp; - res = (((f1->mant * f2->mant) + 0x30) >> 4) << 7; - res = exp > 26 ? res << (exp - 26) : res >> (26 - exp); - return (f1->sign ^ f2->sign) ? -res : res; -} - -static inline int sgn(int value) -{ - return (value < 0) ? -1 : 1; -} - -typedef struct G726Tables { - int bits; /**< bits per sample */ - int* quant; /**< quantization table */ - int* iquant; /**< inverse quantization table */ - int* W; /**< special table #1 ;-) */ - int* F; /**< special table #2 */ -} G726Tables; - -typedef struct G726Context { - G726Tables* tbls; /**< static tables needed for computation */ - - Float11 sr[2]; /**< prev. reconstructed samples */ - Float11 dq[6]; /**< prev. difference */ - int a[2]; /**< second order predictor coeffs */ - int b[6]; /**< sixth order predictor coeffs */ - int pk[2]; /**< signs of prev. 2 sez + dq */ - - int ap; /**< scale factor control */ - int yu; /**< fast scale factor */ - int yl; /**< slow scale factor */ - int dms; /**< short average magnitude of F[i] */ - int dml; /**< long average magnitude of F[i] */ - int td; /**< tone detect */ - - int se; /**< estimated signal for the next iteration */ - int sez; /**< estimated second order prediction */ - int y; /**< quantizer scaling factor for the next iteration */ -} G726Context; - -static int quant_tbl16[] = /**< 16kbit/s 2bits per sample */ - { 260, INT_MAX }; -static int iquant_tbl16[] = - { 116, 365, 365, 116 }; -static int W_tbl16[] = - { -22, 439, 439, -22 }; -static int F_tbl16[] = - { 0, 7, 7, 0 }; - -static int quant_tbl24[] = /**< 24kbit/s 3bits per sample */ - { 7, 217, 330, INT_MAX }; -static int iquant_tbl24[] = - { INT_MIN, 135, 273, 373, 373, 273, 135, INT_MIN }; -static int W_tbl24[] = - { -4, 30, 137, 582, 582, 137, 30, -4 }; -static int F_tbl24[] = - { 0, 1, 2, 7, 7, 2, 1, 0 }; - -static int quant_tbl32[] = /**< 32kbit/s 4bits per sample */ - { -125, 79, 177, 245, 299, 348, 399, INT_MAX }; -static int iquant_tbl32[] = - { INT_MIN, 4, 135, 213, 273, 323, 373, 425, - 425, 373, 323, 273, 213, 135, 4, INT_MIN }; -static int W_tbl32[] = - { -12, 18, 41, 64, 112, 198, 355, 1122, - 1122, 355, 198, 112, 64, 41, 18, -12}; -static int F_tbl32[] = - { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 }; - -static int quant_tbl40[] = /**< 40kbit/s 5bits per sample */ - { -122, -16, 67, 138, 197, 249, 297, 338, - 377, 412, 444, 474, 501, 527, 552, INT_MAX }; -static int iquant_tbl40[] = - { INT_MIN, -66, 28, 104, 169, 224, 274, 318, - 358, 395, 429, 459, 488, 514, 539, 566, - 566, 539, 514, 488, 459, 429, 395, 358, - 318, 274, 224, 169, 104, 28, -66, INT_MIN }; -static int W_tbl40[] = - { 14, 14, 24, 39, 40, 41, 58, 100, - 141, 179, 219, 280, 358, 440, 529, 696, - 696, 529, 440, 358, 280, 219, 179, 141, - 100, 58, 41, 40, 39, 24, 14, 14 }; -static int F_tbl40[] = - { 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6, - 6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 }; - -static G726Tables G726Tables_pool[] = - {{ 2, quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 }, - { 3, quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 }, - { 4, quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 }, - { 5, quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }}; - - -/** - * Para 4.2.2 page 18: Adaptive quantizer. - */ -static inline uint8_t quant(G726Context* c, int d) -{ - int sign, exp, i, dln; - - sign = i = 0; - if (d < 0) { - sign = 1; - d = -d; - } - exp = av_log2_16bit(d); - dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2); - - while (c->tbls->quant[i] < INT_MAX && c->tbls->quant[i] < dln) - ++i; - - if (sign) - i = ~i; - if (c->tbls->bits != 2 && i == 0) /* I'm not sure this is a good idea */ - i = 0xff; - - return i; -} - -/** - * Para 4.2.3 page 22: Inverse adaptive quantizer. - */ -static inline int16_t inverse_quant(G726Context* c, int i) -{ - int dql, dex, dqt; - - dql = c->tbls->iquant[i] + (c->y >> 2); - dex = (dql>>7) & 0xf; /* 4bit exponent */ - dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */ - return (dql < 0) ? 0 : ((dqt<<7) >> (14-dex)); -} - -static inline int16_t g726_iterate(G726Context* c, int16_t I) -{ - int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0; - Float11 f; - - dq = inverse_quant(c, I); - if (I >> (c->tbls->bits - 1)) /* get the sign */ - dq = -dq; - re_signal = c->se + dq; - - /* Transition detect */ - ylint = (c->yl >> 15); - ylfrac = (c->yl >> 10) & 0x1f; - thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint; - if (c->td == 1 && abs(dq) > ((thr2+(thr2>>1))>>1)) - tr = 1; - else - tr = 0; - - /* Update second order predictor coefficient A2 and A1 */ - pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0; - dq0 = dq ? sgn(dq) : 0; - if (tr) { - c->a[0] = 0; - c->a[1] = 0; - for (i=0; i<6; i++) - c->b[i] = 0; - } else { - /* This is a bit crazy, but it really is +255 not +256 */ - fa1 = clip((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255); - - c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7); - c->a[1] = clip(c->a[1], -12288, 12288); - c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8); - c->a[0] = clip(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]); - - for (i=0; i<6; i++) - c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8); - } - - /* Update Dq and Sr and Pk */ - c->pk[1] = c->pk[0]; - c->pk[0] = pk0 ? pk0 : 1; - c->sr[1] = c->sr[0]; - i2f(re_signal, &c->sr[0]); - for (i=5; i>0; i--) - c->dq[i] = c->dq[i-1]; - i2f(dq, &c->dq[0]); - c->dq[0].sign = I >> (c->tbls->bits - 1); /* Isn't it crazy ?!?! */ - - /* Update tone detect [I'm not sure 'tr == 0' is really needed] */ - c->td = (tr == 0 && c->a[1] < -11776); - - /* Update Ap */ - c->dms += ((c->tbls->F[I]<<9) - c->dms) >> 5; - c->dml += ((c->tbls->F[I]<<11) - c->dml) >> 7; - if (tr) - c->ap = 256; - else if (c->y > 1535 && !c->td && (abs((c->dms << 2) - c->dml) < (c->dml >> 3))) - c->ap += (-c->ap) >> 4; - else - c->ap += (0x200 - c->ap) >> 4; - - /* Update Yu and Yl */ - c->yu = clip(c->y + (((c->tbls->W[I] << 5) - c->y) >> 5), 544, 5120); - c->yl += c->yu + ((-c->yl)>>6); - - /* Next iteration for Y */ - al = (c->ap >= 256) ? 1<<6 : c->ap >> 2; - c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6; - - /* Next iteration for SE and SEZ */ - c->se = 0; - for (i=0; i<6; i++) - c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]); - c->sez = c->se >> 1; - for (i=0; i<2; i++) - c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]); - c->se >>= 1; - - return clip(re_signal << 2, -0xffff, 0xffff); -} - -static int g726_reset(G726Context* c, int bit_rate) -{ - int i; - - c->tbls = &G726Tables_pool[bit_rate/8000 - 2]; - for (i=0; i<2; i++) { - i2f(0, &c->sr[i]); - c->a[i] = 0; - c->pk[i] = 1; - } - for (i=0; i<6; i++) { - i2f(0, &c->dq[i]); - c->b[i] = 0; - } - c->ap = 0; - c->dms = 0; - c->dml = 0; - c->yu = 544; - c->yl = 34816; - c->td = 0; - - c->se = 0; - c->sez = 0; - c->y = 544; - - return 0; -} - -static int16_t g726_decode(G726Context* c, int16_t i) -{ - return g726_iterate(c, i); -} - -#ifdef CONFIG_ENCODERS -static int16_t g726_encode(G726Context* c, int16_t sig) -{ - uint8_t i; - - i = quant(c, sig/4 - c->se) & ((1<<c->tbls->bits) - 1); - g726_iterate(c, i); - return i; -} -#endif - -/* Interfacing to the libavcodec */ - -typedef struct AVG726Context { - G726Context c; - int bits_left; - int bit_buffer; - int code_size; -} AVG726Context; - -static int g726_init(AVCodecContext * avctx) -{ - AVG726Context* c = (AVG726Context*)avctx->priv_data; - - if (avctx->channels != 1 || - (avctx->bit_rate != 16000 && avctx->bit_rate != 24000 && - avctx->bit_rate != 32000 && avctx->bit_rate != 40000)) { - av_log(avctx, AV_LOG_ERROR, "G726: unsupported audio format\n"); - return -1; - } - if (avctx->sample_rate != 8000 && avctx->strict_std_compliance>FF_COMPLIANCE_INOFFICIAL) { - av_log(avctx, AV_LOG_ERROR, "G726: unsupported audio format\n"); - return -1; - } - g726_reset(&c->c, avctx->bit_rate); - c->code_size = c->c.tbls->bits; - c->bit_buffer = 0; - c->bits_left = 0; - - avctx->coded_frame = avcodec_alloc_frame(); - if (!avctx->coded_frame) - return -ENOMEM; - avctx->coded_frame->key_frame = 1; - - return 0; -} - -static int g726_close(AVCodecContext *avctx) -{ - av_freep(&avctx->coded_frame); - return 0; -} - -#ifdef CONFIG_ENCODERS -static int g726_encode_frame(AVCodecContext *avctx, - uint8_t *dst, int buf_size, void *data) -{ - AVG726Context *c = avctx->priv_data; - short *samples = data; - PutBitContext pb; - - init_put_bits(&pb, dst, 1024*1024); - - for (; buf_size; buf_size--) - put_bits(&pb, c->code_size, g726_encode(&c->c, *samples++)); - - flush_put_bits(&pb); - - return put_bits_count(&pb)>>3; -} -#endif - -static int g726_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, - uint8_t *buf, int buf_size) -{ - AVG726Context *c = avctx->priv_data; - short *samples = data; - uint8_t code; - uint8_t mask; - GetBitContext gb; - - if (!buf_size) - goto out; - - mask = (1<<c->code_size) - 1; - init_get_bits(&gb, buf, buf_size * 8); - if (c->bits_left) { - int s = c->code_size - c->bits_left;; - code = (c->bit_buffer << s) | get_bits(&gb, s); - *samples++ = g726_decode(&c->c, code & mask); - } - - while (get_bits_count(&gb) + c->code_size <= buf_size*8) - *samples++ = g726_decode(&c->c, get_bits(&gb, c->code_size) & mask); - - c->bits_left = buf_size*8 - get_bits_count(&gb); - c->bit_buffer = get_bits(&gb, c->bits_left); - -out: - *data_size = (uint8_t*)samples - (uint8_t*)data; - return buf_size; -} - -#ifdef CONFIG_ENCODERS -AVCodec adpcm_g726_encoder = { - "g726", - CODEC_TYPE_AUDIO, - CODEC_ID_ADPCM_G726, - sizeof(AVG726Context), - g726_init, - g726_encode_frame, - g726_close, - NULL, -}; -#endif //CONFIG_ENCODERS - -AVCodec adpcm_g726_decoder = { - "g726", - CODEC_TYPE_AUDIO, - CODEC_ID_ADPCM_G726, - sizeof(AVG726Context), - g726_init, - NULL, - g726_close, - g726_decode_frame, -}; |