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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/ | |
*/ | |
/** decoder context */ | |
typedef struct EightSvxContext { | |
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 }; | |
/** | |
* 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, AVFrame *frame, | |
int *got_frame_ptr, AVPacket *avpkt) | |
{ | |
EightSvxContext *esc = avctx->priv_data; | |
int channels = avctx->ch_layout.nb_channels; | |
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 / channels - hdr_size; | |
if (avpkt->size % channels) { | |
av_log(avctx, AV_LOG_WARNING, "Packet with odd size, ignoring last byte\n"); | |
} | |
if (avpkt->size < (hdr_size + 1) * channels) { | |
av_log(avctx, AV_LOG_ERROR, "packet size is too small\n"); | |
return AVERROR_INVALIDDATA; | |
} | |
esc->fib_acc[0] = avpkt->data[1] + 128; | |
if (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 (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 (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_INVALIDDATA; | |
} | |
/* 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 */ | |
frame->nb_samples = buf_size * 2; | |
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) | |
return ret; | |
for (ch = 0; ch < channels; ch++) { | |
delta_decode(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; | |
return ((avctx->frame_num == 0) * hdr_size + buf_size) * channels; | |
} | |
static av_cold int eightsvx_decode_init(AVCodecContext *avctx) | |
{ | |
EightSvxContext *esc = avctx->priv_data; | |
if (avctx->ch_layout.nb_channels < 1 || avctx->ch_layout.nb_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_assert1(0); | |
} | |
avctx->sample_fmt = AV_SAMPLE_FMT_U8P; | |
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; | |
} | |
const FFCodec ff_eightsvx_fib_decoder = { | |
.p.name = "8svx_fib", | |
CODEC_LONG_NAME("8SVX fibonacci"), | |
.p.type = AVMEDIA_TYPE_AUDIO, | |
.p.id = AV_CODEC_ID_8SVX_FIB, | |
.priv_data_size = sizeof (EightSvxContext), | |
.init = eightsvx_decode_init, | |
FF_CODEC_DECODE_CB(eightsvx_decode_frame), | |
.close = eightsvx_decode_close, | |
.p.capabilities = AV_CODEC_CAP_DR1, | |
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P, | |
AV_SAMPLE_FMT_NONE }, | |
}; | |
const FFCodec ff_eightsvx_exp_decoder = { | |
.p.name = "8svx_exp", | |
CODEC_LONG_NAME("8SVX exponential"), | |
.p.type = AVMEDIA_TYPE_AUDIO, | |
.p.id = AV_CODEC_ID_8SVX_EXP, | |
.priv_data_size = sizeof (EightSvxContext), | |
.init = eightsvx_decode_init, | |
FF_CODEC_DECODE_CB(eightsvx_decode_frame), | |
.close = eightsvx_decode_close, | |
.p.capabilities = AV_CODEC_CAP_DR1, | |
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_U8P, | |
AV_SAMPLE_FMT_NONE }, | |
}; | |