From e5966052ee7e07c30a078f31141e3e068847fc7a Mon Sep 17 00:00:00 2001 From: Roman Shaposhnik Date: Sat, 7 Feb 2004 08:20:00 +0000 Subject: [PATCH] * Initial implementation of the G.726 ADPCM audio codec. Originally committed as revision 2759 to svn://svn.ffmpeg.org/ffmpeg/trunk --- libavcodec/Makefile | 2 +- libavcodec/allcodecs.c | 1 + libavcodec/avcodec.h | 2 + libavcodec/g726.c | 417 +++++++++++++++++++++++++++++++++++++++++ libavformat/wav.c | 1 + 5 files changed, 422 insertions(+), 1 deletion(-) create mode 100644 libavcodec/g726.c diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 3d4c321d44..8bfaf7efe5 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -19,7 +19,7 @@ OBJS= common.o utils.o mem.o allcodecs.o \ vp3.o asv1.o 4xm.o cabac.o ffv1.o ra144.o ra288.o vcr1.o cljr.o \ roqvideo.o dpcm.o interplayvideo.o xan.o rpza.o cinepak.o msrle.o \ msvideo1.o vqavideo.o idcinvideo.o adx.o rational.o faandct.o 8bps.o \ - smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o + smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o g726.o ifeq ($(AMR_NB),yes) ifeq ($(AMR_NB_FIXED),yes) diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index 4abc462e90..5fe6dae62b 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -203,6 +203,7 @@ PCM_CODEC(CODEC_ID_ADPCM_4XM, adpcm_4xm); PCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa); PCM_CODEC(CODEC_ID_ADPCM_ADX, adpcm_adx); PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea); +PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_g726); #undef PCM_CODEC diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index 342065a221..8f3b9dc640 100644 --- a/libavcodec/avcodec.h +++ b/libavcodec/avcodec.h @@ -117,6 +117,7 @@ enum CodecID { CODEC_ID_ADPCM_XA, CODEC_ID_ADPCM_ADX, CODEC_ID_ADPCM_EA, + CODEC_ID_ADPCM_G726, /* AMR */ CODEC_ID_AMR_NB, @@ -1730,6 +1731,7 @@ PCM_CODEC(CODEC_ID_ADPCM_4XM, adpcm_4xm); PCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa); PCM_CODEC(CODEC_ID_ADPCM_ADX, adpcm_adx); PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea); +PCM_CODEC(CODEC_ID_ADPCM_G726, adpcm_g726); #undef PCM_CODEC diff --git a/libavcodec/g726.c b/libavcodec/g726.c new file mode 100644 index 0000000000..2166e3b5ef --- /dev/null +++ b/libavcodec/g726.c @@ -0,0 +1,417 @@ +/* + * 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 library 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 of the License, or (at your option) any later version. + * + * This library 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 this library; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ +#include +#include "avcodec.h" +#include "common.h" + +/* + * 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 clamp(int value, int min, int max) +{ + if (value < min) + return min; + else if (value > max) + return max; + else + return value; +} + +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 = clamp((-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] = clamp(c->a[1], -12288, 12288); + c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8); + c->a[0] = clamp(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 = clamp(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 clamp(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); +} + +static int16_t g726_encode(G726Context* c, int16_t sig) +{ + uint8_t i; + + i = quant(c, sig/4 - c->se) & ((1<tbls->bits) - 1); + g726_iterate(c, i); + return i; +} + +/* 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->sample_rate != 8000 || 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; + } + g726_reset(&c->c, avctx->bit_rate); + c->code_size = c->c.tbls->bits; + c->bit_buffer = 0; + c->bits_left = 0; + + return 0; +} + +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; +} + +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<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, + NULL, + NULL, +}; +#endif //CONFIG_ENCODERS + +AVCodec adpcm_g726_decoder = { + "g726", + CODEC_TYPE_AUDIO, + CODEC_ID_ADPCM_G726, + sizeof(AVG726Context), + g726_init, + NULL, + NULL, + g726_decode_frame, +}; diff --git a/libavformat/wav.c b/libavformat/wav.c index 22e605fb48..787c7d542f 100644 --- a/libavformat/wav.c +++ b/libavformat/wav.c @@ -29,6 +29,7 @@ const CodecTag codec_wav_tags[] = { { CODEC_ID_PCM_MULAW, 0x07 }, { CODEC_ID_ADPCM_MS, 0x02 }, { CODEC_ID_ADPCM_IMA_WAV, 0x11 }, + { CODEC_ID_ADPCM_G726, 0x45 }, { CODEC_ID_ADPCM_IMA_DK4, 0x61 }, /* rogue format number */ { CODEC_ID_ADPCM_IMA_DK3, 0x62 }, /* rogue format number */ { CODEC_ID_WMAV1, 0x160 },