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* Initial implementation of the G.726 ADPCM audio codec.
Originally committed as revision 2759 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Roman Shaposhnik
parent
2d9aa1e3a2
commit
e5966052ee
@ -19,7 +19,7 @@ OBJS= common.o utils.o mem.o allcodecs.o \
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vp3.o asv1.o 4xm.o cabac.o ffv1.o ra144.o ra288.o vcr1.o cljr.o \
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roqvideo.o dpcm.o interplayvideo.o xan.o rpza.o cinepak.o msrle.o \
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msvideo1.o vqavideo.o idcinvideo.o adx.o rational.o faandct.o 8bps.o \
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smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o
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smc.o parser.o flicvideo.o truemotion1.o vmdav.o lcl.o qtrle.o g726.o
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ifeq ($(AMR_NB),yes)
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ifeq ($(AMR_NB_FIXED),yes)
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@ -203,6 +203,7 @@ PCM_CODEC(CODEC_ID_ADPCM_4XM, adpcm_4xm);
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PCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa);
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PCM_CODEC(CODEC_ID_ADPCM_ADX, adpcm_adx);
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PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea);
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PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_g726);
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#undef PCM_CODEC
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@ -117,6 +117,7 @@ enum CodecID {
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CODEC_ID_ADPCM_XA,
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CODEC_ID_ADPCM_ADX,
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CODEC_ID_ADPCM_EA,
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CODEC_ID_ADPCM_G726,
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/* AMR */
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CODEC_ID_AMR_NB,
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@ -1730,6 +1731,7 @@ PCM_CODEC(CODEC_ID_ADPCM_4XM, adpcm_4xm);
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PCM_CODEC(CODEC_ID_ADPCM_XA, adpcm_xa);
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PCM_CODEC(CODEC_ID_ADPCM_ADX, adpcm_adx);
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PCM_CODEC(CODEC_ID_ADPCM_EA, adpcm_ea);
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PCM_CODEC(CODEC_ID_ADPCM_G726, adpcm_g726);
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#undef PCM_CODEC
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417
libavcodec/g726.c
Normal file
417
libavcodec/g726.c
Normal file
@ -0,0 +1,417 @@
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/*
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* G.726 ADPCM audio codec
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* Copyright (c) 2004 Roman Shaposhnik.
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*
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* This is a very straightforward rendition of the G.726
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* Section 4 "Computational Details".
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include <limits.h>
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#include "avcodec.h"
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#include "common.h"
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/*
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* G.726 Standard uses rather odd 11bit floating point arithmentic for
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* numerous occasions. It's a mistery to me why they did it this way
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* instead of simply using 32bit integer arithmetic.
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*/
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typedef struct Float11 {
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int sign; /* 1bit sign */
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int exp; /* 4bit exponent */
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int mant; /* 6bit mantissa */
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} Float11;
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static inline Float11* i2f(int16_t i, Float11* f)
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{
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f->sign = (i < 0);
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if (f->sign)
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i = -i;
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f->exp = av_log2_16bit(i) + !!i;
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f->mant = i? (i<<6) >> f->exp :
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1<<5;
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return f;
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}
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static inline int16_t mult(Float11* f1, Float11* f2)
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{
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int res, exp;
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exp = f1->exp + f2->exp;
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res = (((f1->mant * f2->mant) + 0x30) >> 4) << 7;
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res = exp > 26 ? res << (exp - 26) : res >> (26 - exp);
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return (f1->sign ^ f2->sign) ? -res : res;
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}
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static inline int clamp(int value, int min, int max)
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{
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if (value < min)
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return min;
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else if (value > max)
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return max;
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else
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return value;
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}
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static inline int sgn(int value)
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{
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return (value < 0) ? -1 : 1;
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}
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typedef struct G726Tables {
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int bits; /* bits per sample */
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int* quant; /* quantization table */
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int* iquant; /* inverse quantization table */
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int* W; /* special table #1 ;-) */
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int* F; /* special table #2 */
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} G726Tables;
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typedef struct G726Context {
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G726Tables* tbls; /* static tables needed for computation */
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Float11 sr[2]; /* prev. reconstructed samples */
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Float11 dq[6]; /* prev. difference */
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int a[2]; /* second order predictor coeffs */
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int b[6]; /* sixth order predictor coeffs */
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int pk[2]; /* signs of prev. 2 sez + dq */
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int ap; /* scale factor control */
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int yu; /* fast scale factor */
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int yl; /* slow scale factor */
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int dms; /* short average magnitude of F[i] */
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int dml; /* long average magnitude of F[i] */
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int td; /* tone detect */
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int se; /* estimated signal for the next iteration */
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int sez; /* estimated second order prediction */
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int y; /* quantizer scaling factor for the next iteration */
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} G726Context;
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static int quant_tbl16[] = /* 16kbit/s 2bits per sample */
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{ 260, INT_MAX };
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static int iquant_tbl16[] =
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{ 116, 365, 365, 116 };
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static int W_tbl16[] =
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{ -22, 439, 439, -22 };
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static int F_tbl16[] =
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{ 0, 7, 7, 0 };
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static int quant_tbl24[] = /* 24kbit/s 3bits per sample */
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{ 7, 217, 330, INT_MAX };
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static int iquant_tbl24[] =
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{ INT_MIN, 135, 273, 373, 373, 273, 135, INT_MIN };
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static int W_tbl24[] =
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{ -4, 30, 137, 582, 582, 137, 30, -4 };
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static int F_tbl24[] =
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{ 0, 1, 2, 7, 7, 2, 1, 0 };
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static int quant_tbl32[] = /* 32kbit/s 4bits per sample */
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{ -125, 79, 177, 245, 299, 348, 399, INT_MAX };
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static int iquant_tbl32[] =
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{ INT_MIN, 4, 135, 213, 273, 323, 373, 425,
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425, 373, 323, 273, 213, 135, 4, INT_MIN };
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static int W_tbl32[] =
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{ -12, 18, 41, 64, 112, 198, 355, 1122,
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1122, 355, 198, 112, 64, 41, 18, -12};
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static int F_tbl32[] =
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{ 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
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static int quant_tbl40[] = /* 40kbit/s 5bits per sample */
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{ -122, -16, 67, 138, 197, 249, 297, 338,
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377, 412, 444, 474, 501, 527, 552, INT_MAX };
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static int iquant_tbl40[] =
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{ INT_MIN, -66, 28, 104, 169, 224, 274, 318,
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358, 395, 429, 459, 488, 514, 539, 566,
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566, 539, 514, 488, 459, 429, 395, 358,
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318, 274, 224, 169, 104, 28, -66, INT_MIN };
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static int W_tbl40[] =
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{ 14, 14, 24, 39, 40, 41, 58, 100,
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141, 179, 219, 280, 358, 440, 529, 696,
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696, 529, 440, 358, 280, 219, 179, 141,
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100, 58, 41, 40, 39, 24, 14, 14 };
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static int F_tbl40[] =
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{ 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6,
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6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 };
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static G726Tables G726Tables_pool[] =
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{{ 2, quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
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{ 3, quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
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{ 4, quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
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{ 5, quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }};
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/*
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* Para 4.2.2 page 18: Adaptive quantizer.
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*/
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static inline uint8_t quant(G726Context* c, int d)
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{
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int sign, exp, i, dln;
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sign = i = 0;
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if (d < 0) {
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sign = 1;
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d = -d;
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}
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exp = av_log2_16bit(d);
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dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2);
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while (c->tbls->quant[i] < INT_MAX && c->tbls->quant[i] < dln)
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++i;
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if (sign)
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i = ~i;
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if (c->tbls->bits != 2 && i == 0) /* I'm not sure this is a good idea */
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i = 0xff;
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return i;
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}
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/*
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* Para 4.2.3 page 22: Inverse adaptive quantizer.
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*/
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static inline int16_t inverse_quant(G726Context* c, int i)
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{
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int dql, dex, dqt;
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dql = c->tbls->iquant[i] + (c->y >> 2);
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dex = (dql>>7) & 0xf; /* 4bit exponent */
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dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
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return (dql < 0) ? 0 : ((dqt<<7) >> (14-dex));
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}
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static inline int16_t g726_iterate(G726Context* c, int16_t I)
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{
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int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
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Float11 f;
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dq = inverse_quant(c, I);
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if (I >> (c->tbls->bits - 1)) /* get the sign */
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dq = -dq;
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re_signal = c->se + dq;
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/* Transition detect */
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ylint = (c->yl >> 15);
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ylfrac = (c->yl >> 10) & 0x1f;
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thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
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if (c->td == 1 && abs(dq) > ((thr2+(thr2>>1))>>1))
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tr = 1;
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else
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tr = 0;
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/* Update second order predictor coefficient A2 and A1 */
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pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0;
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dq0 = dq ? sgn(dq) : 0;
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if (tr) {
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c->a[0] = 0;
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c->a[1] = 0;
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for (i=0; i<6; i++)
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c->b[i] = 0;
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} else {
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/* This is a bit crazy, but it really is +255 not +256 */
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fa1 = clamp((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255);
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c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7);
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c->a[1] = clamp(c->a[1], -12288, 12288);
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c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8);
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c->a[0] = clamp(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]);
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for (i=0; i<6; i++)
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c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8);
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}
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/* Update Dq and Sr and Pk */
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c->pk[1] = c->pk[0];
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c->pk[0] = pk0 ? pk0 : 1;
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c->sr[1] = c->sr[0];
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i2f(re_signal, &c->sr[0]);
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for (i=5; i>0; i--)
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c->dq[i] = c->dq[i-1];
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i2f(dq, &c->dq[0]);
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c->dq[0].sign = I >> (c->tbls->bits - 1); /* Isn't it crazy ?!?! */
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/* Update tone detect [I'm not sure 'tr == 0' is really needed] */
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c->td = (tr == 0 && c->a[1] < -11776);
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/* Update Ap */
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c->dms += ((c->tbls->F[I]<<9) - c->dms) >> 5;
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c->dml += ((c->tbls->F[I]<<11) - c->dml) >> 7;
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if (tr)
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c->ap = 256;
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else if (c->y > 1535 && !c->td && (abs((c->dms << 2) - c->dml) < (c->dml >> 3)))
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c->ap += (-c->ap) >> 4;
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else
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c->ap += (0x200 - c->ap) >> 4;
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/* Update Yu and Yl */
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c->yu = clamp(c->y + (((c->tbls->W[I] << 5) - c->y) >> 5), 544, 5120);
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c->yl += c->yu + ((-c->yl)>>6);
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/* Next iteration for Y */
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al = (c->ap >= 256) ? 1<<6 : c->ap >> 2;
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c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6;
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/* Next iteration for SE and SEZ */
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c->se = 0;
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for (i=0; i<6; i++)
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c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]);
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c->sez = c->se >> 1;
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for (i=0; i<2; i++)
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c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]);
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c->se >>= 1;
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return clamp(re_signal << 2, -0xffff, 0xffff);
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}
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static int g726_reset(G726Context* c, int bit_rate)
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{
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int i;
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c->tbls = &G726Tables_pool[bit_rate/8000 - 2];
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for (i=0; i<2; i++) {
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i2f(0, &c->sr[i]);
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c->a[i] = 0;
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c->pk[i] = 1;
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}
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for (i=0; i<6; i++) {
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i2f(0, &c->dq[i]);
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c->b[i] = 0;
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}
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c->ap = 0;
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c->dms = 0;
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c->dml = 0;
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c->yu = 544;
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c->yl = 34816;
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c->td = 0;
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c->se = 0;
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c->sez = 0;
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c->y = 544;
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return 0;
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}
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static int16_t g726_decode(G726Context* c, int16_t i)
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{
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return g726_iterate(c, i);
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}
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static int16_t g726_encode(G726Context* c, int16_t sig)
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{
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uint8_t i;
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i = quant(c, sig/4 - c->se) & ((1<<c->tbls->bits) - 1);
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g726_iterate(c, i);
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return i;
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}
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/* Interfacing to the libavcodec */
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typedef struct AVG726Context {
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G726Context c;
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int bits_left;
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int bit_buffer;
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int code_size;
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} AVG726Context;
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static int g726_init(AVCodecContext * avctx)
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{
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AVG726Context* c = (AVG726Context*)avctx->priv_data;
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if (avctx->sample_rate != 8000 || avctx->channels != 1 ||
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(avctx->bit_rate != 16000 && avctx->bit_rate != 24000 &&
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avctx->bit_rate != 32000 && avctx->bit_rate != 40000)) {
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av_log(avctx, AV_LOG_ERROR, "G726: unsupported audio format\n");
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return -1;
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}
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g726_reset(&c->c, avctx->bit_rate);
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c->code_size = c->c.tbls->bits;
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c->bit_buffer = 0;
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c->bits_left = 0;
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return 0;
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}
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static int g726_encode_frame(AVCodecContext *avctx,
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uint8_t *dst, int buf_size, void *data)
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{
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AVG726Context *c = avctx->priv_data;
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short *samples = data;
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PutBitContext pb;
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init_put_bits(&pb, dst, 1024*1024);
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for (; buf_size; buf_size--)
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put_bits(&pb, c->code_size, g726_encode(&c->c, *samples++));
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flush_put_bits(&pb);
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return put_bits_count(&pb)>>3;
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}
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static int g726_decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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uint8_t *buf, int buf_size)
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{
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AVG726Context *c = avctx->priv_data;
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short *samples = data;
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uint8_t code;
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uint8_t mask;
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GetBitContext gb;
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if (!buf_size)
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goto out;
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mask = (1<<c->code_size) - 1;
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init_get_bits(&gb, buf, buf_size * 8);
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if (c->bits_left) {
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int s = c->code_size - c->bits_left;;
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code = (c->bit_buffer << s) | get_bits(&gb, s);
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*samples++ = g726_decode(&c->c, code & mask);
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}
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while (get_bits_count(&gb) + c->code_size <= buf_size*8)
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*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,
|
||||
};
|
||||
@ -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 },
|
||||
|
||||
Loading…
Reference in New Issue
Block a user