mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-25 00:32:31 +00:00
246 lines
8.0 KiB
C
246 lines
8.0 KiB
C
/*
|
|
* 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 "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;
|
|
size_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;
|
|
|
|
esc->samples_size = avctx->codec->id == CODEC_ID_8SVX_RAW || avctx->codec->id ==CODEC_ID_PCM_S8_PLANAR?
|
|
avpkt->size : avctx->channels + (avpkt->size-avctx->channels) * 2;
|
|
if (!(esc->samples = av_malloc(esc->samples_size)))
|
|
return AVERROR(ENOMEM);
|
|
|
|
/* decompress */
|
|
if (avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP) {
|
|
const uint8_t *buf = avpkt->data;
|
|
int buf_size = avpkt->size;
|
|
int 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);
|
|
p = deinterleaved_samples;
|
|
|
|
/* the uncompressed starting value is contained in the first byte */
|
|
if (avctx->channels == 2) {
|
|
delta_decode(deinterleaved_samples , buf+1, buf_size/2-1, buf[0], esc->table);
|
|
buf += buf_size/2;
|
|
delta_decode(deinterleaved_samples+n/2-1, buf+1, buf_size/2-1, buf[0], esc->table);
|
|
} else
|
|
delta_decode(deinterleaved_samples , buf+1, buf_size-1 , buf[0], esc->table);
|
|
} 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 */
|
|
esc->frame.nb_samples = (FFMIN(MAX_FRAME_SIZE, esc->samples_size - esc->samples_idx) +avctx->channels-1) / 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 avctx->codec->id == CODEC_ID_8SVX_FIB || avctx->codec->id == CODEC_ID_8SVX_EXP ?
|
|
(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;
|
|
}
|
|
|
|
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"),
|
|
};
|
|
|
|
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"),
|
|
};
|
|
|
|
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"),
|
|
};
|