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FFmpeg/libavcodec/g726.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks 
This is possible, because every given FFCodec has to implement
exactly one of these. Doing so decreases sizeof(FFCodec) and
therefore decreases the size of the binary.
Notice that in case of position-independent code the decrease
is in .data.rel.ro, so that this translates to decreased
memory consumption.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-04-05 20:02:37 +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 "config_components.h"
#include <limits.h>
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avcodec.h"
#include "codec_internal.h"
#include "encode.h"
#include "internal.h"
#include "get_bits.h"
#include "put_bits.h"
/**
* G.726 11-bit float.
* G.726 Standard uses rather odd 11-bit floating point arithmetic for
* numerous occasions. It's a mystery to me why they did it this way
* instead of simply using 32-bit integer arithmetic.
*/
typedef struct Float11 {
uint8_t sign; /**< 1 bit sign */
uint8_t exp; /**< 4 bits exponent */
uint8_t mant; /**< 6 bits 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;
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;
int little_endian; /**< little-endian bitstream as used in aiff and Sun AU */
} G726Context;
static const int quant_tbl16[] = /**< 16kbit/s 2 bits 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 3 bits 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 4 bits 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 5 bits 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 }};
/**
* Paragraph 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;
}
/**
* Paragraph 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; /* 4-bit 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 = (int16_t)(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_intp2((-c->a[0]*c->pk[0]*pk0)>>5, 8);
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 * 4, -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 || CONFIG_ADPCM_G726LE_ENCODER
static int16_t g726_encode(G726Context* c, int16_t sig)
{
uint8_t i;
i = av_mod_uintp2(quant(c, sig/4 - c->se), c->code_size);
g726_decode(c, i);
return i;
}
/* Interfacing to the libavcodec */
static av_cold int g726_encode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
c->little_endian = !strcmp(avctx->codec->name, "g726le");
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);
}
if (avctx->sample_rate <= 0) {
av_log(avctx, AV_LOG_ERROR, "Invalid sample rate %d\n",
avctx->sample_rate);
return AVERROR(EINVAL);
}
if (avctx->ch_layout.nb_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);
/* 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 int g726_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
G726Context *c = avctx->priv_data;
const int16_t *samples = (const int16_t *)frame->data[0];
PutBitContext pb;
int i, ret, out_size;
out_size = (frame->nb_samples * c->code_size + 7) / 8;
if ((ret = ff_get_encode_buffer(avctx, avpkt, out_size, 0)) < 0)
return ret;
init_put_bits(&pb, avpkt->data, avpkt->size);
for (i = 0; i < frame->nb_samples; i++)
if (c->little_endian) {
put_bits_le(&pb, c->code_size, g726_encode(c, *samples++));
} else {
put_bits(&pb, c->code_size, g726_encode(c, *samples++));
}
if (c->little_endian) {
flush_put_bits_le(&pb);
} else {
flush_put_bits(&pb);
}
*got_packet_ptr = 1;
return 0;
}
#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, { .i64 = 4 }, 2, 5, AE },
{ NULL },
};
static const AVClass g726_class = {
.class_name = "g726",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static const FFCodecDefault defaults[] = {
{ "b", "0" },
{ NULL },
};
#endif
#if CONFIG_ADPCM_G726_ENCODER
const FFCodec ff_adpcm_g726_encoder = {
.p.name = "g726",
.p.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_ADPCM_G726,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SMALL_LAST_FRAME,
.priv_data_size = sizeof(G726Context),
.init = g726_encode_init,
FF_CODEC_ENCODE_CB(g726_encode_frame),
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.p.priv_class = &g726_class,
.defaults = defaults,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_ADPCM_G726LE_ENCODER
const FFCodec ff_adpcm_g726le_encoder = {
.p.name = "g726le",
.p.long_name = NULL_IF_CONFIG_SMALL("G.726 little endian ADPCM (\"right-justified\")"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_ADPCM_G726LE,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_SMALL_LAST_FRAME,
.priv_data_size = sizeof(G726Context),
.init = g726_encode_init,
FF_CODEC_ENCODE_CB(g726_encode_frame),
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.p.priv_class = &g726_class,
.defaults = defaults,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_ADPCM_G726_DECODER || CONFIG_ADPCM_G726LE_DECODER
static av_cold int g726_decode_init(AVCodecContext *avctx)
{
G726Context* c = avctx->priv_data;
if (avctx->ch_layout.nb_channels > 1){
avpriv_request_sample(avctx, "Decoding more than one channel");
return AVERROR_PATCHWELCOME;
}
av_channel_layout_uninit(&avctx->ch_layout);
avctx->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
c->little_endian = !strcmp(avctx->codec->name, "g726le");
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;
return 0;
}
static int g726_decode_frame(AVCodecContext *avctx, AVFrame *frame,
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 */
frame->nb_samples = out_samples;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
samples = (int16_t *)frame->data[0];
init_get_bits(&gb, buf, buf_size * 8);
while (out_samples--)
*samples++ = g726_decode(c, c->little_endian ?
get_bits_le(&gb, c->code_size) :
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;
return buf_size;
}
static void g726_decode_flush(AVCodecContext *avctx)
{
G726Context *c = avctx->priv_data;
g726_reset(c);
}
#endif
#if CONFIG_ADPCM_G726_DECODER
const FFCodec ff_adpcm_g726_decoder = {
.p.name = "g726",
.p.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_ADPCM_G726,
.priv_data_size = sizeof(G726Context),
.init = g726_decode_init,
FF_CODEC_DECODE_CB(g726_decode_frame),
.flush = g726_decode_flush,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_ADPCM_G726LE_DECODER
const FFCodec ff_adpcm_g726le_decoder = {
.p.name = "g726le",
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_ADPCM_G726LE,
.priv_data_size = sizeof(G726Context),
.init = g726_decode_init,
FF_CODEC_DECODE_CB(g726_decode_frame),
.flush = g726_decode_flush,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
.p.long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM little-endian"),
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
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