FFmpeg/libavcodec/ws-snd1.c
Michael Niedermayer 7c1aba4f01 Merge remote-tracking branch 'qatar/master'
* qatar/master: (21 commits)
  fate: allow testing with libavfilter disabled
  x86: XOP/FMA4 CPU detection support
  ws_snd: misc cosmetic clean-ups
  ws_snd: remove the 2-bit ADPCM table and just subtract 2 instead.
  ws_snd: use memcpy() and memset() instead of loops
  ws_snd: use samples pointer for loop termination instead of a separate iterator variable.
  ws_snd: make sure number of channels is 1
  ws_snd: add some checks to prevent buffer overread or overwrite.
  ws_snd: decode to AV_SAMPLE_FMT_U8 instead of S16.
  flacdec: fix buffer size checking in get_metadata_size()
  rtp: Simplify ff_rtp_get_payload_type
  rtpenc: Add a payload type private option
  rtp: Correct ff_rtp_get_payload_type documentation
  avconv: replace all fprintf() by av_log().
  avconv: change av_log verbosity from ERROR to FATAL for fatal errors.
  cmdutils: replace fprintf() by av_log()
  avtools: parse loglevel before all the other options.
  oggdec: add support for Xiph's CELT codec
  sol: return error if av_get_packet() fails.
  cosmetics: reindent and pretty-print
  ...

Conflicts:
	avconv.c
	cmdutils.c
	libavcodec/avcodec.h
	libavcodec/version.h
	libavformat/oggparsecelt.c
	libavformat/utils.c
	libavutil/avutil.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-09-27 02:14:37 +02:00

175 lines
5.2 KiB
C

/*
* Westwood SNDx codecs
* Copyright (c) 2005 Konstantin Shishkov
*
* 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
*/
#include <stdint.h>
#include "libavutil/intreadwrite.h"
#include "avcodec.h"
/**
* @file
* Westwood SNDx codecs
*
* Reference documents about VQA format and its audio codecs
* can be found here:
* http://www.multimedia.cx
*/
static const int8_t ws_adpcm_4bit[] = {
-9, -8, -6, -5, -4, -3, -2, -1,
0, 1, 2, 3, 4, 5, 6, 8
};
static av_cold int ws_snd_decode_init(AVCodecContext *avctx)
{
if (avctx->channels != 1) {
av_log_ask_for_sample(avctx, "unsupported number of channels\n");
return AVERROR(EINVAL);
}
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
return 0;
}
static int ws_snd_decode_frame(AVCodecContext *avctx, void *data,
int *data_size, AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int in_size, out_size;
int sample = 128;
uint8_t *samples = data;
uint8_t *samples_end;
if (!buf_size)
return 0;
if (buf_size < 4) {
av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
return AVERROR(EINVAL);
}
out_size = AV_RL16(&buf[0]);
in_size = AV_RL16(&buf[2]);
buf += 4;
if (out_size > *data_size) {
av_log(avctx, AV_LOG_ERROR, "Frame is too large to fit in buffer\n");
return -1;
}
if (in_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "Frame data is larger than input buffer\n");
return -1;
}
samples_end = samples + out_size;
if (in_size == out_size) {
memcpy(samples, buf, out_size);
*data_size = out_size;
return buf_size;
}
while (samples < samples_end && buf - avpkt->data < buf_size) {
int code, smp, size;
uint8_t count;
code = *buf >> 6;
count = *buf & 0x3F;
buf++;
/* make sure we don't write past the output buffer */
switch (code) {
case 0: smp = 4; break;
case 1: smp = 2; break;
case 2: smp = (count & 0x20) ? 1 : count + 1; break;
default: smp = count + 1; break;
}
if (samples_end - samples < smp)
break;
/* make sure we don't read past the input buffer */
size = ((code == 2 && (count & 0x20)) || code == 3) ? 0 : count + 1;
if ((buf - avpkt->data) + size > buf_size)
break;
switch (code) {
case 0: /* ADPCM 2-bit */
for (count++; count > 0; count--) {
code = *buf++;
sample += ( code & 0x3) - 2;
sample = av_clip_uint8(sample);
*samples++ = sample;
sample += ((code >> 2) & 0x3) - 2;
sample = av_clip_uint8(sample);
*samples++ = sample;
sample += ((code >> 4) & 0x3) - 2;
sample = av_clip_uint8(sample);
*samples++ = sample;
sample += (code >> 6) - 2;
sample = av_clip_uint8(sample);
*samples++ = sample;
}
break;
case 1: /* ADPCM 4-bit */
for (count++; count > 0; count--) {
code = *buf++;
sample += ws_adpcm_4bit[code & 0xF];
sample = av_clip_uint8(sample);
*samples++ = sample;
sample += ws_adpcm_4bit[code >> 4];
sample = av_clip_uint8(sample);
*samples++ = sample;
}
break;
case 2: /* no compression */
if (count & 0x20) { /* big delta */
int8_t t;
t = count;
t <<= 3;
sample += t >> 3;
sample = av_clip_uint8(sample);
*samples++ = sample;
} else { /* copy */
memcpy(samples, buf, smp);
samples += smp;
buf += smp;
sample = buf[-1];
}
break;
default: /* run */
memset(samples, sample, smp);
samples += smp;
}
}
*data_size = samples - (uint8_t *)data;
return buf_size;
}
AVCodec ff_ws_snd1_decoder = {
.name = "ws_snd1",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_WESTWOOD_SND1,
.init = ws_snd_decode_init,
.decode = ws_snd_decode_frame,
.long_name = NULL_IF_CONFIG_SMALL("Westwood Audio (SND1)"),
};