Mercurial > libavcodec.hg
view g726.c @ 6920:d02af7474bff libavcodec
Prevent 128*1<<trellis from becoming 0 and creating 0 sized arrays.
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author | michael |
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date | Wed, 28 May 2008 11:59:41 +0000 |
parents | a4104482ceef |
children | e943e1409077 |
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/* * 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 "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 */ const int* quant; /**< quantization table */ const int* iquant; /**< inverse quantization table */ const int* W; /**< special table #1 ;-) */ const int* F; /**< special table #2 */ } G726Tables; typedef struct G726Context { const 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 const int quant_tbl16[] = /**< 16kbit/s 2bits per sample */ { 260, INT_MAX }; static const int iquant_tbl16[] = { 116, 365, 365, 116 }; static const int W_tbl16[] = { -22, 439, 439, -22 }; static const int F_tbl16[] = { 0, 7, 7, 0 }; static const int quant_tbl24[] = /**< 24kbit/s 3bits per sample */ { 7, 217, 330, INT_MAX }; static const int iquant_tbl24[] = { INT_MIN, 135, 273, 373, 373, 273, 135, INT_MIN }; static const int W_tbl24[] = { -4, 30, 137, 582, 582, 137, 30, -4 }; static const int F_tbl24[] = { 0, 1, 2, 7, 7, 2, 1, 0 }; static const int quant_tbl32[] = /**< 32kbit/s 4bits per sample */ { -125, 79, 177, 245, 299, 348, 399, INT_MAX }; static const int iquant_tbl32[] = { INT_MIN, 4, 135, 213, 273, 323, 373, 425, 425, 373, 323, 273, 213, 135, 4, INT_MIN }; static const int W_tbl32[] = { -12, 18, 41, 64, 112, 198, 355, 1122, 1122, 355, 198, 112, 64, 41, 18, -12}; static const int F_tbl32[] = { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 }; static const 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 const 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 const 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 const 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 const 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 = av_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] = av_clip(c->a[1], -12288, 12288); c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8); c->a[0] = av_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 = av_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 av_clip(re_signal << 2, -0xffff, 0xffff); } static av_cold 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 av_cold 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 AVERROR(ENOMEM); avctx->coded_frame->key_frame = 1; return 0; } static av_cold 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, const 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, .long_name = "G.726 ADPCM", }; #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, .long_name = "G.726 ADPCM", };