ffmpeg/libavcodec/adxdec.c
Justin Ruggles b237248e29 adx: calculate correct LPC coeffs
Instead of using fixed coefficients, the correct way is to calculate the
coefficients using the highpass cutoff frequency from the ADX stream header
and the sample rate.
2011-11-26 16:25:06 -05:00

171 lines
5.0 KiB
C

/*
* ADX ADPCM codecs
* Copyright (c) 2001,2003 BERO
*
* 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
*/
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
#include "adx.h"
#include "get_bits.h"
/**
* @file
* SEGA CRI adx codecs.
*
* Reference documents:
* http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
* adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
*/
static av_cold int adx_decode_init(AVCodecContext *avctx)
{
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
return 0;
}
/**
* Decode 32 samples from 18 bytes.
*
* A 16-bit scalar value is applied to 32 residuals, which then have a
* 2nd-order LPC filter applied to it to form the output signal for a single
* channel.
*/
static void adx_decode(ADXContext *c, int16_t *out, const uint8_t *in, int ch)
{
ADXChannelState *prev = &c->prev[ch];
GetBitContext gb;
int scale = AV_RB16(in);
int i;
int s0, s1, s2, d;
init_get_bits(&gb, in + 2, (18 - 2) * 8);
s1 = prev->s1;
s2 = prev->s2;
for (i = 0; i < 32; i++) {
d = get_sbits(&gb, 4);
s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
s2 = s1;
s1 = av_clip_int16(s0);
*out = s1;
out += c->channels;
}
prev->s1 = s1;
prev->s2 = s2;
}
/**
* Decode stream header.
*
* @param avctx codec context
* @param buf packet data
* @param bufsize packet size
* @return data offset or negative error code if header is invalid
*/
static int adx_decode_header(AVCodecContext *avctx, const uint8_t *buf,
int bufsize)
{
ADXContext *c = avctx->priv_data;
int offset, cutoff;
if (AV_RB16(buf) != 0x8000)
return AVERROR_INVALIDDATA;
offset = AV_RB16(buf + 2) + 4;
if (bufsize < offset || memcmp(buf + offset - 6, "(c)CRI", 6))
return AVERROR_INVALIDDATA;
c->channels = avctx->channels = buf[7];
if (avctx->channels > 2)
return AVERROR_INVALIDDATA;
avctx->sample_rate = AV_RB32(buf + 8);
if (avctx->sample_rate < 1 ||
avctx->sample_rate > INT_MAX / (avctx->channels * 18 * 8))
return AVERROR_INVALIDDATA;
avctx->bit_rate = avctx->sample_rate * avctx->channels * 18 * 8 / 32;
cutoff = AV_RB16(buf + 16);
ff_adx_calculate_coeffs(cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
return offset;
}
static int adx_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf0 = avpkt->data;
int buf_size = avpkt->size;
ADXContext *c = avctx->priv_data;
int16_t *samples = data;
const uint8_t *buf = buf0;
int rest = buf_size;
if (!c->header_parsed) {
int hdrsize = adx_decode_header(avctx, buf, rest);
if (hdrsize < 0) {
av_log(avctx, AV_LOG_ERROR, "invalid stream header\n");
return hdrsize;
}
c->header_parsed = 1;
buf += hdrsize;
rest -= hdrsize;
}
/* 18 bytes of data are expanded into 32*2 bytes of audio,
so guard against buffer overflows */
if (rest / 18 > *data_size / 64)
rest = (*data_size / 64) * 18;
if (c->in_temp) {
int copysize = 18 * avctx->channels - c->in_temp;
memcpy(c->dec_temp + c->in_temp, buf, copysize);
rest -= copysize;
buf += copysize;
adx_decode(c, samples, c->dec_temp, 0);
if (avctx->channels == 2)
adx_decode(c, samples + 1, c->dec_temp + 18, 1);
samples += 32 * c->channels;
}
while (rest >= 18 * c->channels) {
adx_decode(c, samples, buf, 0);
if (c->channels == 2)
adx_decode(c, samples + 1, buf + 18, 1);
rest -= 18 * c->channels;
buf += 18 * c->channels;
samples += 32 * c->channels;
}
c->in_temp = rest;
if (rest) {
memcpy(c->dec_temp, buf, rest);
buf += rest;
}
*data_size = (uint8_t*)samples - (uint8_t*)data;
return buf - buf0;
}
AVCodec ff_adpcm_adx_decoder = {
.name = "adpcm_adx",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_ADPCM_ADX,
.priv_data_size = sizeof(ADXContext),
.init = adx_decode_init,
.decode = adx_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
};