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/*
* Copyright (C) 2008 Jaikrishnan Menon
* Copyright (C) 2011 Stefano Sabatini
*
* 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
* 8svx audio decoder
* @author Jaikrishnan Menon
*
* supports: fibonacci delta encoding
* : exponential encoding
*
* For more information about the 8SVX format:
* http://netghost.narod.ru/gff/vendspec/iff/iff.txt
* http://sox.sourceforge.net/AudioFormats-11.html
* http://aminet.net/package/mus/misc/wavepak
* http://amigan.1emu.net/reg/8SVX.txt
*
* Samples can be found here:
* http://aminet.net/mods/smpl/
*/
#include "libavutil/avassert.h"
#include "avcodec.h"
#include "internal.h"
#include "libavutil/common.h"
/** decoder context */
typedef struct EightSvxContext {
AVFrame frame;
uint8_t fib_acc[2];
const int8_t *table;
/* buffer used to store the whole first packet.
data is only sent as one large packet */
uint8_t *data[2];
int data_size;
int data_idx;
} EightSvxContext;
static const int8_t fibonacci[16] = { -34, -21, -13, -8, -5, -3, -2, -1, 0, 1, 2, 3, 5, 8, 13, 21 };
static const int8_t exponential[16] = { -128, -64, -32, -16, -8, -4, -2, -1, 0, 1, 2, 4, 8, 16, 32, 64 };
#define MAX_FRAME_SIZE 2048
/**
* Delta decode the compressed values in src, and put the resulting
* decoded samples in dst.
*
* @param[in,out] state starting value. it is saved for use in the next call.
* @param table delta sequence table
*/
static void delta_decode(uint8_t *dst, const uint8_t *src, int src_size,
uint8_t *state, const int8_t *table)
{
uint8_t val = *state;
while (src_size--) {
uint8_t d = *src++;
val = av_clip_uint8(val + table[d & 0xF]);
*dst++ = val;
val = av_clip_uint8(val + table[d >> 4]);
*dst++ = val;
}
*state = val;
}
/** decode a frame */
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
EightSvxContext *esc = avctx->priv_data;
int buf_size;
int ch, ret;
int hdr_size = 2;
/* decode and interleave the first packet */
if (!esc->data[0] && avpkt) {
int chan_size = avpkt->size / avctx->channels - hdr_size;
if (avpkt->size % avctx->channels) {
av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
}
if (avpkt->size < (hdr_size + 1) * avctx->channels) {
av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
return AVERROR(EINVAL);
}
esc->fib_acc[0] = avpkt->data[1] + 128;
if (avctx->channels == 2)
esc->fib_acc[1] = avpkt->data[2+chan_size+1] + 128;
esc->data_idx = 0;
esc->data_size = chan_size;
if (!(esc->data[0] = av_malloc(chan_size)))
return AVERROR(ENOMEM);
if (avctx->channels == 2) {
if (!(esc->data[1] = av_malloc(chan_size))) {
av_freep(&esc->data[0]);
return AVERROR(ENOMEM);
}
}
memcpy(esc->data[0], &avpkt->data[hdr_size], chan_size);
if (avctx->channels == 2)
memcpy(esc->data[1], &avpkt->data[2*hdr_size+chan_size], chan_size);
}
if (!esc->data[0]) {
av_log(avctx, AV_LOG_ERROR, "unexpected empty packet\n");
return AVERROR(EINVAL);
}
/* decode next piece of data from the buffer */
buf_size = FFMIN(MAX_FRAME_SIZE, esc->data_size - esc->data_idx);
if (buf_size <= 0) {
*got_frame_ptr = 0;
return avpkt->size;
}
/* get output buffer */
esc->frame.nb_samples = buf_size * 2;
if ((ret = ff_get_buffer(avctx, &esc->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
for (ch = 0; ch < avctx->channels; ch++) {
delta_decode(esc->frame.data[ch], &esc->data[ch][esc->data_idx],
buf_size, &esc->fib_acc[ch], esc->table);
}
esc->data_idx += buf_size;
*got_frame_ptr = 1;
*(AVFrame *)data = esc->frame;
return ((avctx->frame_number == 0)*hdr_size + buf_size)*avctx->channels;
}
static av_cold int eightsvx_decode_init(AVCodecContext *avctx)
{
EightSvxContext *esc = avctx->priv_data;
if (avctx->channels < 1 || avctx->channels > 2) {
av_log(avctx, AV_LOG_ERROR, "8SVX does not support more than 2 channels\n");
return AVERROR_INVALIDDATA;
}
switch (avctx->codec->id) {
case AV_CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
case AV_CODEC_ID_8SVX_EXP: esc->table = exponential; break;
default:
av_log(avctx, AV_LOG_ERROR, "Invalid codec id %d.\n", avctx->codec->id);
return AVERROR_INVALIDDATA;
}
avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
avcodec_get_frame_defaults(&esc->frame);
avctx->coded_frame = &esc->frame;
return 0;
}
static av_cold int eightsvx_decode_close(AVCodecContext *avctx)
{
EightSvxContext *esc = avctx->priv_data;
av_freep(&esc->data[0]);
av_freep(&esc->data[1]);
esc->data_size = 0;
esc->data_idx = 0;
return 0;
}
#if CONFIG_EIGHTSVX_FIB_DECODER
AVCodec ff_eightsvx_fib_decoder = {
.name = "8svx_fib",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_8SVX_FIB,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
.close = eightsvx_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX fibonacci"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_EIGHTSVX_EXP_DECODER
AVCodec ff_eightsvx_exp_decoder = {
.name = "8svx_exp",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_8SVX_EXP,
.priv_data_size = sizeof (EightSvxContext),
.init = eightsvx_decode_init,
.decode = eightsvx_decode_frame,
.close = eightsvx_decode_close,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("8SVX exponential"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_NONE },
};
#endif
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