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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"
/** decoder context */
typedef struct EightSvxContext {
AVFrame frame;
const int8_t *table;
/* buffer used to store the whole audio decoded/interleaved chunk,
* which is sent with the first packet */
uint8_t *samples;
int64_t samples_size;
int samples_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
/**
* Interleave samples in buffer containing all left channel samples
* at the beginning, and right channel samples at the end.
* Each sample is assumed to be in signed 8-bit format.
*
* @param size the size in bytes of the dst and src buffer
*/
static void interleave_stereo(uint8_t *dst, const uint8_t *src, int size)
{
uint8_t *dst_end = dst + size;
size = size>>1;
while (dst < dst_end) {
*dst++ = *src;
*dst++ = *(src+size);
src++;
}
}
/**
* Delta decode the compressed values in src, and put the resulting
* decoded n samples in dst.
*
* @param val starting value assumed by the delta sequence
* @param table delta sequence table
* @return size in bytes of the decoded data, must be src_size*2
*/
static int delta_decode(int8_t *dst, const uint8_t *src, int src_size,
int8_t val, const int8_t *table)
{
int n = src_size;
int8_t *dst0 = dst;
while (n--) {
uint8_t d = *src++;
val = av_clip(val + table[d & 0x0f], -127, 128);
*dst++ = val;
val = av_clip(val + table[d >> 4] , -127, 128);
*dst++ = val;
}
return dst-dst0;
}
/** decode a frame */
static int eightsvx_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
EightSvxContext *esc = avctx->priv_data;
int n, out_data_size, ret;
uint8_t *src, *dst;
/* decode and interleave the first packet */
if (!esc->samples && avpkt) {
uint8_t *deinterleaved_samples, *p = NULL;
int packet_size = avpkt->size;
if (packet_size % avctx->channels) {
av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n");
if (packet_size < avctx->channels)
return packet_size;
packet_size -= packet_size % avctx->channels;
}
esc->samples_size = !esc->table ?
packet_size : avctx->channels + (packet_size-avctx->channels) * 2;
if (!(esc->samples = av_malloc(esc->samples_size)))
return AVERROR(ENOMEM);
/* decompress */
if (esc->table) {
const uint8_t *buf = avpkt->data;
uint8_t *dst;
int buf_size = avpkt->size;
int i, n = esc->samples_size;
if (buf_size < 2) {
av_log(avctx, AV_LOG_ERROR, "packet size is too small\n");
return AVERROR(EINVAL);
}
if (!(deinterleaved_samples = av_mallocz(n)))
return AVERROR(ENOMEM);
dst = p = deinterleaved_samples;
/* the uncompressed starting value is contained in the first byte */
dst = deinterleaved_samples;
for (i = 0; i < avctx->channels; i++) {
delta_decode(dst, buf + 1, buf_size / avctx->channels - 1, buf[0], esc->table);
buf += buf_size / avctx->channels;
dst += n / avctx->channels - 1;
}
} else {
deinterleaved_samples = avpkt->data;
}
if (avctx->channels == 2)
interleave_stereo(esc->samples, deinterleaved_samples, esc->samples_size);
else
memcpy(esc->samples, deinterleaved_samples, esc->samples_size);
av_freep(&p);
}
/* get output buffer */
av_assert1(!(esc->samples_size % avctx->channels || esc->samples_idx % avctx->channels));
esc->frame.nb_samples = FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) / avctx->channels;
if ((ret = avctx->get_buffer(avctx, &esc->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
*got_frame_ptr = 1;
*(AVFrame *)data = esc->frame;
dst = esc->frame.data[0];
src = esc->samples + esc->samples_idx;
out_data_size = esc->frame.nb_samples * avctx->channels;
for (n = out_data_size; n > 0; n--)
*dst++ = *src++ + 128;
esc->samples_idx += out_data_size;
return esc->table ?
(avctx->frame_number == 0)*2 + out_data_size / 2 :
out_data_size;
}
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 CODEC_ID_8SVX_FIB: esc->table = fibonacci; break;
case CODEC_ID_8SVX_EXP: esc->table = exponential; break;
case CODEC_ID_PCM_S8_PLANAR:
case CODEC_ID_8SVX_RAW: esc->table = NULL; 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_U8;
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->samples);
esc->samples_size = 0;
esc->samples_idx = 0;
return 0;
}
#if CONFIG_EIGHTSVX_FIB_DECODER
AVCodec ff_eightsvx_fib_decoder = {
.name = "8svx_fib",
.type = AVMEDIA_TYPE_AUDIO,
.id = 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"),
};
#endif
#if CONFIG_EIGHTSVX_EXP_DECODER
AVCodec ff_eightsvx_exp_decoder = {
.name = "8svx_exp",
.type = AVMEDIA_TYPE_AUDIO,
.id = 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"),
};
#endif
#if CONFIG_PCM_S8_PLANAR_DECODER
AVCodec ff_pcm_s8_planar_decoder = {
.name = "pcm_s8_planar",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_PCM_S8_PLANAR,
.priv_data_size = sizeof(EightSvxContext),
.init = eightsvx_decode_init,
.close = eightsvx_decode_close,
.decode = eightsvx_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("PCM signed 8-bit planar"),
};
#endif
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