Commit 58e37eaf authored by Martin Storsjö's avatar Martin Storsjö
Browse files

Add G.722 ADPCM audio decoder

Originally committed as revision 25086 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent a3f0d2b9
......@@ -33,6 +33,7 @@ version <next>:
- Apple HTTP Live Streaming demuxer
- a64 codec
- MMS-HTTP support
- G.722 ADPCM audio decoder
version 0.6:
......
......@@ -535,6 +535,7 @@ following image formats are supported:
@item ADPCM Electronic Arts R2 @tab @tab X
@item ADPCM Electronic Arts R3 @tab @tab X
@item ADPCM Electronic Arts XAS @tab @tab X
@item ADPCM G.722 @tab @tab X
@item ADPCM G.726 @tab X @tab X
@item ADPCM IMA AMV @tab @tab X
@tab Used in AMV files
......
......@@ -475,6 +475,7 @@ OBJS-$(CONFIG_ADPCM_EA_R1_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_EA_R2_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_EA_R3_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_EA_XAS_DECODER) += adpcm.o
OBJS-$(CONFIG_ADPCM_G722_DECODER) += g722.o
OBJS-$(CONFIG_ADPCM_G726_DECODER) += g726.o
OBJS-$(CONFIG_ADPCM_G726_ENCODER) += g726.o
OBJS-$(CONFIG_ADPCM_IMA_AMV_DECODER) += adpcm.o
......
......@@ -317,6 +317,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (ADPCM_EA_R2, adpcm_ea_r2);
REGISTER_DECODER (ADPCM_EA_R3, adpcm_ea_r3);
REGISTER_DECODER (ADPCM_EA_XAS, adpcm_ea_xas);
REGISTER_DECODER (ADPCM_G722, adpcm_g722);
REGISTER_ENCDEC (ADPCM_G726, adpcm_g726);
REGISTER_DECODER (ADPCM_IMA_AMV, adpcm_ima_amv);
REGISTER_DECODER (ADPCM_IMA_DK3, adpcm_ima_dk3);
......
......@@ -31,8 +31,8 @@
#include "libavutil/cpu.h"
#define LIBAVCODEC_VERSION_MAJOR 52
#define LIBAVCODEC_VERSION_MINOR 87
#define LIBAVCODEC_VERSION_MICRO 5
#define LIBAVCODEC_VERSION_MINOR 88
#define LIBAVCODEC_VERSION_MICRO 0
#define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
LIBAVCODEC_VERSION_MINOR, \
......@@ -284,6 +284,7 @@ enum CodecID {
CODEC_ID_ADPCM_EA_XAS,
CODEC_ID_ADPCM_EA_MAXIS_XA,
CODEC_ID_ADPCM_IMA_ISS,
CODEC_ID_ADPCM_G722,
/* AMR */
CODEC_ID_AMR_NB= 0x12000,
......
/*
* G.722 ADPCM audio decoder
*
* Copyright (c) CMU 1993 Computer Science, Speech Group
* Chengxiang Lu and Alex Hauptmann
* Copyright (c) 2005 Steve Underwood <steveu at coppice.org>
* Copyright (c) 2009 Kenan Gillet
* Copyright (c) 2010 Martin Storsjo
*
* 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
*/
/**
* @file
*
* G.722 ADPCM audio codec
*
* This G.722 decoder is a bit-exact implementation of the ITU G.722
* specification for all three specified bitrates - 64000bps, 56000bps
* and 48000bps. It passes the ITU tests.
*
* @note For the 56000bps and 48000bps bitrates, the lowest 1 or 2 bits
* respectively of each byte are ignored.
*/
#include "avcodec.h"
#include "mathops.h"
#include "get_bits.h"
#define PREV_SAMPLES_BUF_SIZE 1024
typedef struct {
int16_t prev_samples[PREV_SAMPLES_BUF_SIZE]; ///< memory of past decoded samples
int prev_samples_pos; ///< the number of values in prev_samples
/**
* The band[0] and band[1] correspond respectively to the lower band and higher band.
*/
struct G722Band {
int16_t s_predictor; ///< predictor output value
int32_t s_zero; ///< previous output signal from zero predictor
int8_t part_reconst_mem[2]; ///< signs of previous partially reconstructed signals
int16_t prev_qtzd_reconst; ///< previous quantized reconstructed signal (internal value, using low_inv_quant4)
int16_t pole_mem[2]; ///< second-order pole section coefficient buffer
int32_t diff_mem[6]; ///< quantizer difference signal memory
int16_t zero_mem[6]; ///< Seventh-order zero section coefficient buffer
int16_t log_factor; ///< delayed 2-logarithmic quantizer factor
int16_t scale_factor; ///< delayed quantizer scale factor
} band[2];
} G722Context;
static const int8_t sign_lookup[2] = { -1, 1 };
static const int16_t inv_log2_table[32] = {
2048, 2093, 2139, 2186, 2233, 2282, 2332, 2383,
2435, 2489, 2543, 2599, 2656, 2714, 2774, 2834,
2896, 2960, 3025, 3091, 3158, 3228, 3298, 3371,
3444, 3520, 3597, 3676, 3756, 3838, 3922, 4008
};
static const int16_t high_log_factor_step[2] = { 798, -214 };
static const int16_t high_inv_quant[4] = { -926, -202, 926, 202 };
/**
* low_log_factor_step[index] == wl[rl42[index]]
*/
static const int16_t low_log_factor_step[16] = {
-60, 3042, 1198, 538, 334, 172, 58, -30,
3042, 1198, 538, 334, 172, 58, -30, -60
};
static const int16_t low_inv_quant4[16] = {
0, -2557, -1612, -1121, -786, -530, -323, -150,
2557, 1612, 1121, 786, 530, 323, 150, 0
};
/**
* quadrature mirror filter (QMF) coefficients
*
* ITU-T G.722 Table 11
*/
static const int16_t qmf_coeffs[12] = {
3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
};
/**
* adaptive predictor
*
* @param cur_diff the dequantized and scaled delta calculated from the
* current codeword
*/
static void do_adaptive_prediction(struct G722Band *band, const int cur_diff)
{
int sg[2], limit, i, cur_qtzd_reconst;
const int cur_part_reconst = band->s_zero + cur_diff < 0;
sg[0] = sign_lookup[cur_part_reconst != band->part_reconst_mem[0]];
sg[1] = sign_lookup[cur_part_reconst == band->part_reconst_mem[1]];
band->part_reconst_mem[1] = band->part_reconst_mem[0];
band->part_reconst_mem[0] = cur_part_reconst;
band->pole_mem[1] = av_clip((sg[0] * av_clip(band->pole_mem[0], -8191, 8191) >> 5) +
(sg[1] << 7) + (band->pole_mem[1] * 127 >> 7), -12288, 12288);
limit = 15360 - band->pole_mem[1];
band->pole_mem[0] = av_clip(-192 * sg[0] + (band->pole_mem[0] * 255 >> 8), -limit, limit);
if (cur_diff) {
for (i = 0; i < 6; i++)
band->zero_mem[i] = ((band->zero_mem[i]*255) >> 8) +
((band->diff_mem[i]^cur_diff) < 0 ? -128 : 128);
} else
for (i = 0; i < 6; i++)
band->zero_mem[i] = (band->zero_mem[i]*255) >> 8;
for (i = 5; i > 0; i--)
band->diff_mem[i] = band->diff_mem[i-1];
band->diff_mem[0] = av_clip_int16(cur_diff << 1);
band->s_zero = 0;
for (i = 5; i >= 0; i--)
band->s_zero += (band->zero_mem[i]*band->diff_mem[i]) >> 15;
cur_qtzd_reconst = av_clip_int16((band->s_predictor + cur_diff) << 1);
band->s_predictor = av_clip_int16(band->s_zero +
(band->pole_mem[0] * cur_qtzd_reconst >> 15) +
(band->pole_mem[1] * band->prev_qtzd_reconst >> 15));
band->prev_qtzd_reconst = cur_qtzd_reconst;
}
static int inline linear_scale_factor(const int log_factor)
{
const int wd1 = inv_log2_table[(log_factor >> 6) & 31];
const int shift = log_factor >> 11;
return shift < 0 ? wd1 >> -shift : wd1 << shift;
}
static void update_low_predictor(struct G722Band *band, const int ilow)
{
do_adaptive_prediction(band,
band->scale_factor * low_inv_quant4[ilow] >> 10);
// quantizer adaptation
band->log_factor = av_clip((band->log_factor * 127 >> 7) +
low_log_factor_step[ilow], 0, 18432);
band->scale_factor = linear_scale_factor(band->log_factor - (8 << 11));
}
static void update_high_predictor(struct G722Band *band, const int dhigh,
const int ihigh)
{
do_adaptive_prediction(band, dhigh);
// quantizer adaptation
band->log_factor = av_clip((band->log_factor * 127 >> 7) +
high_log_factor_step[ihigh&1], 0, 22528);
band->scale_factor = linear_scale_factor(band->log_factor - (10 << 11));
}
static void apply_qmf(const int16_t *prev_samples, int *xout1, int *xout2)
{
int i;
*xout1 = 0;
*xout2 = 0;
for (i = 0; i < 12; i++) {
MAC16(*xout2, prev_samples[2*i ], qmf_coeffs[i ]);
MAC16(*xout1, prev_samples[2*i+1], qmf_coeffs[11-i]);
}
}
static av_cold int g722_init(AVCodecContext * avctx)
{
G722Context *c = avctx->priv_data;
if (avctx->channels != 1) {
av_log(avctx, AV_LOG_ERROR, "Only mono tracks are allowed.\n");
return AVERROR_INVALIDDATA;
}
avctx->sample_fmt = SAMPLE_FMT_S16;
switch (avctx->bits_per_coded_sample) {
case 8:
case 7:
case 6:
break;
default:
av_log(avctx, AV_LOG_WARNING, "Unsupported bits_per_coded_sample [%d], "
"assuming 8\n",
avctx->bits_per_coded_sample);
case 0:
avctx->bits_per_coded_sample = 8;
break;
}
c->band[0].scale_factor = 8;
c->band[1].scale_factor = 2;
c->prev_samples_pos = 22;
if (avctx->lowres)
avctx->sample_rate /= 2;
return 0;
}
static const int16_t low_inv_quant5[32] = {
-35, -35, -2919, -2195, -1765, -1458, -1219, -1023,
-858, -714, -587, -473, -370, -276, -190, -110,
2919, 2195, 1765, 1458, 1219, 1023, 858, 714,
587, 473, 370, 276, 190, 110, 35, -35
};
static const int16_t low_inv_quant6[64] = {
-17, -17, -17, -17, -3101, -2738, -2376, -2088,
-1873, -1689, -1535, -1399, -1279, -1170, -1072, -982,
-899, -822, -750, -682, -618, -558, -501, -447,
-396, -347, -300, -254, -211, -170, -130, -91,
3101, 2738, 2376, 2088, 1873, 1689, 1535, 1399,
1279, 1170, 1072, 982, 899, 822, 750, 682,
618, 558, 501, 447, 396, 347, 300, 254,
211, 170, 130, 91, 54, 17, -54, -17
};
static const int16_t *low_inv_quants[3] = { low_inv_quant6, low_inv_quant5,
low_inv_quant4 };
static int g722_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
{
G722Context *c = avctx->priv_data;
int16_t *out_buf = data;
int j, out_len = 0;
const int skip = 8 - avctx->bits_per_coded_sample;
const int16_t *quantizer_table = low_inv_quants[skip];
GetBitContext gb;
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
for (j = 0; j < avpkt->size; j++) {
int ilow, ihigh, rlow;
ihigh = get_bits(&gb, 2);
ilow = get_bits(&gb, 6 - skip);
skip_bits(&gb, skip);
rlow = av_clip((c->band[0].scale_factor * quantizer_table[ilow] >> 10)
+ c->band[0].s_predictor, -16384, 16383);
update_low_predictor(&c->band[0], ilow >> (2 - skip));
if (!avctx->lowres) {
const int dhigh = c->band[1].scale_factor *
high_inv_quant[ihigh] >> 10;
const int rhigh = av_clip(dhigh + c->band[1].s_predictor,
-16384, 16383);
int xout1, xout2;
update_high_predictor(&c->band[1], dhigh, ihigh);
c->prev_samples[c->prev_samples_pos++] = rlow + rhigh;
c->prev_samples[c->prev_samples_pos++] = rlow - rhigh;
apply_qmf(c->prev_samples + c->prev_samples_pos - 24,
&xout1, &xout2);
out_buf[out_len++] = av_clip_int16(xout1 >> 12);
out_buf[out_len++] = av_clip_int16(xout2 >> 12);
if (c->prev_samples_pos >= PREV_SAMPLES_BUF_SIZE) {
memmove(c->prev_samples,
c->prev_samples + c->prev_samples_pos - 22,
22 * sizeof(c->prev_samples[0]));
c->prev_samples_pos = 22;
}
} else
out_buf[out_len++] = rlow;
}
*data_size = out_len << 1;
return avpkt->size;
}
AVCodec adpcm_g722_decoder = {
.name = "g722",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ADPCM_G722,
.priv_data_size = sizeof(G722Context),
.init = g722_init,
.decode = g722_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
.max_lowres = 1,
};
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