1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavcodec/g722dec.c
Martin Storsjö b087ce2bee g722: Fix the QMF scaling
This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.

This makes the decoder output have double the magnitude
compared to before.

The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.

Signed-off-by: Martin Storsjö <martin@martin.st>
2012-03-02 18:58:19 +02:00

155 lines
5.2 KiB
C

/*
* 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 Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* G.722 ADPCM audio decoder
*
* 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 "get_bits.h"
#include "g722.h"
#include "libavutil/opt.h"
#define OFFSET(x) offsetof(G722Context, x)
#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
static const AVOption options[] = {
{ "bits_per_codeword", "Bits per G722 codeword", OFFSET(bits_per_codeword), AV_OPT_TYPE_FLAGS, { 8 }, 6, 8, AD },
{ NULL }
};
static const AVClass g722_decoder_class = {
.class_name = "g722 decoder",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static av_cold int g722_decode_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 = AV_SAMPLE_FMT_S16;
c->band[0].scale_factor = 8;
c->band[1].scale_factor = 2;
c->prev_samples_pos = 22;
avcodec_get_frame_defaults(&c->frame);
avctx->coded_frame = &c->frame;
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_quants[3] = { ff_g722_low_inv_quant6,
low_inv_quant5,
ff_g722_low_inv_quant4 };
static int g722_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
G722Context *c = avctx->priv_data;
int16_t *out_buf;
int j, ret;
const int skip = 8 - c->bits_per_codeword;
const int16_t *quantizer_table = low_inv_quants[skip];
GetBitContext gb;
/* get output buffer */
c->frame.nb_samples = avpkt->size * 2;
if ((ret = avctx->get_buffer(avctx, &c->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
out_buf = (int16_t *)c->frame.data[0];
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
for (j = 0; j < avpkt->size; j++) {
int ilow, ihigh, rlow, rhigh, dhigh;
int xout1, xout2;
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);
ff_g722_update_low_predictor(&c->band[0], ilow >> (2 - skip));
dhigh = c->band[1].scale_factor * ff_g722_high_inv_quant[ihigh] >> 10;
rhigh = av_clip(dhigh + c->band[1].s_predictor, -16384, 16383);
ff_g722_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;
ff_g722_apply_qmf(c->prev_samples + c->prev_samples_pos - 24,
&xout1, &xout2);
*out_buf++ = av_clip_int16(xout1 >> 11);
*out_buf++ = av_clip_int16(xout2 >> 11);
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;
}
}
*got_frame_ptr = 1;
*(AVFrame *)data = c->frame;
return avpkt->size;
}
AVCodec ff_adpcm_g722_decoder = {
.name = "g722",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ADPCM_G722,
.priv_data_size = sizeof(G722Context),
.init = g722_decode_init,
.decode = g722_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("G.722 ADPCM"),
.priv_class = &g722_decoder_class,
};