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ffmpeg/libavcodec/g726.c
Michael Niedermayer bb015b519e Merge commit '0ab76ddf313eeab70d06619ae0376fd7dd40761b' into release/0.10 
* commit '0ab76ddf313eeab70d06619ae0376fd7dd40761b':
  avcodec: Introduce ff_get_buffer

Conflicts:
	libavcodec/8svx.c
	libavcodec/dpcm.c
	libavcodec/utils.c
	libavcodec/vmdav.c
	libavcodec/yop.c

See: 668494acd8
Merged-by: Michael Niedermayer <michaelni@gmx.at>
2014-08-12 19:53:37 +02:00

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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 "libavutil/avassert.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "put_bits.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 {
uint8_t sign; /**< 1bit sign */
uint8_t exp; /**< 4bit exponent */
uint8_t mant; /**< 6bit mantissa */
} Float11;
static inline Float11* i2f(int 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);
res = exp > 19 ? res << (exp - 19) : res >> (19 - exp);
return (f1->sign ^ f2->sign) ? -res : res;
}
static inline int sgn(int value)
{
return (value < 0) ? -1 : 1;
}
typedef struct G726Tables {
const int* quant; /**< quantization table */
const int16_t* iquant; /**< inverse quantization table */
const int16_t* W; /**< special table #1 ;-) */
const uint8_t* F; /**< special table #2 */
} G726Tables;
typedef struct G726Context {
AVClass *class;
AVFrame frame;
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 */
int code_size;
} G726Context;
static const int quant_tbl16[] = /**< 16kbit/s 2bits per sample */
{ 260, INT_MAX };
static const int16_t iquant_tbl16[] =
{ 116, 365, 365, 116 };
static const int16_t W_tbl16[] =
{ -22, 439, 439, -22 };
static const uint8_t F_tbl16[] =
{ 0, 7, 7, 0 };
static const int quant_tbl24[] = /**< 24kbit/s 3bits per sample */
{ 7, 217, 330, INT_MAX };
static const int16_t iquant_tbl24[] =
{ INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN };
static const int16_t W_tbl24[] =
{ -4, 30, 137, 582, 582, 137, 30, -4 };
static const uint8_t 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 int16_t iquant_tbl32[] =
{ INT16_MIN, 4, 135, 213, 273, 323, 373, 425,
425, 373, 323, 273, 213, 135, 4, INT16_MIN };
static const int16_t W_tbl32[] =
{ -12, 18, 41, 64, 112, 198, 355, 1122,
1122, 355, 198, 112, 64, 41, 18, -12};
static const uint8_t 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 int16_t iquant_tbl40[] =
{ INT16_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, INT16_MIN };
static const int16_t 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 uint8_t 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[] =
{{ quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
{ quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
{ quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
{ 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->code_size != 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<<dex) >> 7);
}
static int16_t g726_decode(G726Context* c, int I)
{
int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
Float11 f;
int I_sig= I >> (c->code_size - 1);
dq = inverse_quant(c, I);
/* Transition detect */
ylint = (c->yl >> 15);
ylfrac = (c->yl >> 10) & 0x1f;
thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
tr= (c->td == 1 && dq > ((3*thr2)>>2));
if (I_sig) /* get the sign */
dq = -dq;
re_signal = c->se + dq;
/* 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_sig; /* Isn't it crazy ?!?! */
c->td = c->a[1] < -11776;
/* Update Ap */
c->dms += (c->tbls.F[I]<<4) + ((- c->dms) >> 5);
c->dml += (c->tbls.F[I]<<4) + ((- c->dml) >> 7);
if (tr)
c->ap = 256;
else {
c->ap += (-c->ap) >> 4;
if (c->y <= 1535 || c->td || abs((c->dms << 2) - c->dml) >= (c->dml >> 3))
c->ap += 0x20;
}
/* Update Yu and Yl */
c->yu = av_clip(c->y + c->tbls.W[I] + ((-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 i;
c->tbls = G726Tables_pool[c->code_size - 2];
for (i=0; i<2; i++) {
c->sr[i].mant = 1<<5;
c->pk[i] = 1;
}
for (i=0; i<6; i++) {
c->dq[i].mant = 1<<5;
}
c->yu = 544;
c->yl = 34816;
c->y = 544;
return 0;
}
#if CONFIG_ADPCM_G726_ENCODER
static int16_t g726_encode(G726Context* c, int16_t sig)
{
uint8_t i;
i = quant(c, sig/4 - c->se) & ((1<<c->code_size) - 1);
g726_decode(c, i);
return i;
}
/* Interfacing to the libavcodec */
static av_cold int g726_encode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
if (avctx->strict_std_compliance > FF_COMPLIANCE_UNOFFICIAL &&
avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Sample rates other than 8kHz are not "
"allowed when the compliance level is higher than unofficial. "
"Resample or reduce the compliance level.\n");
return AVERROR(EINVAL);
}
av_assert0(avctx->sample_rate > 0);
if(avctx->channels != 1){
av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
return AVERROR(EINVAL);
}
if (avctx->bit_rate)
c->code_size = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate;
c->code_size = av_clip(c->code_size, 2, 5);
avctx->bit_rate = c->code_size * avctx->sample_rate;
avctx->bits_per_coded_sample = c->code_size;
g726_reset(c);
avctx->coded_frame = avcodec_alloc_frame();
if (!avctx->coded_frame)
return AVERROR(ENOMEM);
avctx->coded_frame->key_frame = 1;
/* select a frame size that will end on a byte boundary and have a size of
approximately 1024 bytes */
avctx->frame_size = ((int[]){ 4096, 2736, 2048, 1640 })[c->code_size - 2];
return 0;
}
static av_cold int g726_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
static int g726_encode_frame(AVCodecContext *avctx,
uint8_t *dst, int buf_size, void *data)
{
G726Context *c = avctx->priv_data;
const int16_t *samples = data;
PutBitContext pb;
int i;
init_put_bits(&pb, dst, 1024*1024);
for (i = 0; i < avctx->frame_size; i++)
put_bits(&pb, c->code_size, g726_encode(c, *samples++));
flush_put_bits(&pb);
return put_bits_count(&pb)>>3;
}
#define OFFSET(x) offsetof(G726Context, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
{ "code_size", "Bits per code", OFFSET(code_size), AV_OPT_TYPE_INT, { 4 }, 2, 5, AE },
{ NULL },
};
static const AVClass class = {
.class_name = "g726",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static const AVCodecDefault defaults[] = {
{ "b", "0" },
{ NULL },
};
AVCodec ff_adpcm_g726_encoder = {
.name = "g726",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ADPCM_G726,
.priv_data_size = sizeof(G726Context),
.init = g726_encode_init,
.encode = g726_encode_frame,
.close = g726_encode_close,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
.priv_class = &class,
.defaults = defaults,
};
#endif
#if CONFIG_ADPCM_G726_DECODER
static av_cold int g726_decode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
if (avctx->strict_std_compliance >= FF_COMPLIANCE_STRICT &&
avctx->sample_rate != 8000) {
av_log(avctx, AV_LOG_ERROR, "Only 8kHz sample rate is allowed when "
"the compliance level is strict. Reduce the compliance level "
"if you wish to decode the stream anyway.\n");
return AVERROR(EINVAL);
}
if(avctx->channels != 1){
av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
return AVERROR(EINVAL);
}
c->code_size = avctx->bits_per_coded_sample;
if (c->code_size < 2 || c->code_size > 5) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of bits %d\n", c->code_size);
return AVERROR(EINVAL);
}
g726_reset(c);
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
return 0;
}
static int g726_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
G726Context *c = avctx->priv_data;
int16_t *samples;
GetBitContext gb;
int out_samples, ret;
out_samples = buf_size * 8 / c->code_size;
/* get output buffer */
c->frame.nb_samples = out_samples;
if ((ret = ff_get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = (int16_t *)c->frame.data[0];
init_get_bits(&gb, buf, buf_size * 8);
while (out_samples--)
*samples++ = g726_decode(c, get_bits(&gb, c->code_size));
if (get_bits_left(&gb) > 0)
av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return buf_size;
}
static void g726_decode_flush(AVCodecContext *avctx)
{
G726Context *c = avctx->priv_data;
g726_reset(c);
}
AVCodec ff_adpcm_g726_decoder = {
.name = "g726",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ADPCM_G726,
.priv_data_size = sizeof(G726Context),
.init = g726_decode_init,
.decode = g726_decode_frame,
.flush = g726_decode_flush,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
};
#endif
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