ffmpeg/libavcodec/ws-snd1.c
Michael Niedermayer 8551c6bec0 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  dv1394: Swap the min and max values of the 'standard' option
  rtpdec_vp8: Don't parse fields that aren't used
  lavc: add some AVPacket doxy.
  audiointerleave: deobfuscate a function call.
  rtpdec: factorize identical code used in several handlers
  a64: remove interleaved mode.
  doc: Point to the new location of the c99-to-c89 tool
  decode_audio3: initialize AVFrame
  ws-snd1: set channel layout
  wmavoice: set channel layout
  wmapro: use AVCodecContext.channels instead of keeping a private copy
  wma: do not keep private copies of some AVCodecContext fields

Conflicts:
	libavcodec/wmadec.c
	libavcodec/wmaenc.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-11-02 14:57:36 +01:00

195 lines
5.8 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/audioconvert.h"
#include "libavutil/common.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
};
typedef struct WSSndContext {
AVFrame frame;
} WSSndContext;
static av_cold int ws_snd_decode_init(AVCodecContext *avctx)
{
WSSndContext *s = avctx->priv_data;
avctx->channels = 1;
avctx->channel_layout = AV_CH_LAYOUT_MONO;
avctx->sample_fmt = AV_SAMPLE_FMT_U8;
avcodec_get_frame_defaults(&s->frame);
avctx->coded_frame = &s->frame;
return 0;
}
static int ws_snd_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
WSSndContext *s = avctx->priv_data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
int in_size, out_size, ret;
int sample = 128;
uint8_t *samples;
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 (in_size > buf_size) {
av_log(avctx, AV_LOG_ERROR, "Frame data is larger than input buffer\n");
return -1;
}
/* get output buffer */
s->frame.nb_samples = out_size;
if ((ret = avctx->get_buffer(avctx, &s->frame)) < 0) {
av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
return ret;
}
samples = s->frame.data[0];
samples_end = samples + out_size;
if (in_size == out_size) {
memcpy(samples, buf, out_size);
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
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 * (count + 1); break;
case 1: smp = 2 * (count + 1); 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;
}
}
s->frame.nb_samples = samples - s->frame.data[0];
*got_frame_ptr = 1;
*(AVFrame *)data = s->frame;
return buf_size;
}
AVCodec ff_ws_snd1_decoder = {
.name = "ws_snd1",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_WESTWOOD_SND1,
.priv_data_size = sizeof(WSSndContext),
.init = ws_snd_decode_init,
.decode = ws_snd_decode_frame,
.capabilities = CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Westwood Audio (SND1)"),
};