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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-02 03:06:28 +02:00
FFmpeg/libavformat/vocdec.c
Michael Niedermayer 6cb2085278 Merge remote-tracking branch 'qatar/master'
* qatar/master: (27 commits)
  ppc: Add ff_ prefix to nonstatic symbols
  sh4: Add ff_ prefix to nonstatic symbols
  mpegvideo: Add ff_ prefix to nonstatic functions
  rtjpeg: Add ff_ prefix to nonstatic symbols
  rv: Add ff_ prefix to nonstatic symbols
  vp56: Add ff_ prefix to nonstatic symbols
  vorbis: Add ff_ prefix to nonstatic symbols
  msmpeg4: Add ff_ prefix to nonstatic symbols
  vc1: Add ff_ prefix to nonstatic symbols
  msmpeg4: Add ff_ prefixes to nonstatic symbols
  snow: Add ff_ prefix to nonstatic symbols
  mpeg12: Add ff_ prefix to nonstatic symbols
  mpeg4: Add ff_ prefixes to nonstatic symbols
  lagarith: Add ff_ prefix to lag_rac_init
  libavcodec: Add ff_ prefix to j_rev_dct*
  dsputil: Add ff_ prefix to inv_zigzag_direct16
  libavcodec: Prefix fdct_ifast, fdct_ifast248
  dsputil: Add ff_ prefix to the dsputil*_init* functions
  libavcodec: Add ff_ prefix to some nonstatic symbols
  vlc/rl: Add ff_ prefix to the nonstatic symbols
  ...

Conflicts:
	libavcodec/Makefile
	libavcodec/allcodecs.c
	libavcodec/dnxhddec.c
	libavcodec/ffv1.c
	libavcodec/h263.h
	libavcodec/h263dec.c
	libavcodec/h264.c
	libavcodec/mpegvideo.c
	libavcodec/mpegvideo_enc.c
	libavcodec/nuv.c
	libavcodec/ppc/dsputil_ppc.c
	libavcodec/proresdsp.c
	libavcodec/svq3.c
	libavcodec/version.h
	libavformat/dv.h
	libavformat/dvenc.c
	libavformat/matroskadec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-02-16 01:34:37 +01:00

168 lines
5.0 KiB
C

/*
* Creative Voice File demuxer.
* Copyright (c) 2006 Aurelien Jacobs <aurel@gnuage.org>
*
* 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 "libavutil/intreadwrite.h"
#include "voc.h"
#include "internal.h"
static int voc_probe(AVProbeData *p)
{
int version, check;
if (memcmp(p->buf, ff_voc_magic, sizeof(ff_voc_magic) - 1))
return 0;
version = AV_RL16(p->buf + 22);
check = AV_RL16(p->buf + 24);
if (~version + 0x1234 != check)
return 10;
return AVPROBE_SCORE_MAX;
}
static int voc_read_header(AVFormatContext *s)
{
VocDecContext *voc = s->priv_data;
AVIOContext *pb = s->pb;
int header_size;
AVStream *st;
avio_skip(pb, 20);
header_size = avio_rl16(pb) - 22;
if (header_size != 4) {
av_log(s, AV_LOG_ERROR, "unknown header size: %d\n", header_size);
return AVERROR(ENOSYS);
}
avio_skip(pb, header_size);
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
voc->remaining_size = 0;
return 0;
}
int
ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
{
VocDecContext *voc = s->priv_data;
AVCodecContext *dec = st->codec;
AVIOContext *pb = s->pb;
VocType type;
int size, tmp_codec=-1;
int sample_rate = 0;
int channels = 1;
while (!voc->remaining_size) {
type = avio_r8(pb);
if (type == VOC_TYPE_EOF)
return AVERROR(EIO);
voc->remaining_size = avio_rl24(pb);
if (!voc->remaining_size) {
if (!s->pb->seekable)
return AVERROR(EIO);
voc->remaining_size = avio_size(pb) - avio_tell(pb);
}
max_size -= 4;
switch (type) {
case VOC_TYPE_VOICE_DATA:
dec->sample_rate = 1000000 / (256 - avio_r8(pb));
if (sample_rate)
dec->sample_rate = sample_rate;
dec->channels = channels;
tmp_codec = avio_r8(pb);
dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
voc->remaining_size -= 2;
max_size -= 2;
channels = 1;
break;
case VOC_TYPE_VOICE_DATA_CONT:
break;
case VOC_TYPE_EXTENDED:
sample_rate = avio_rl16(pb);
avio_r8(pb);
channels = avio_r8(pb) + 1;
sample_rate = 256000000 / (channels * (65536 - sample_rate));
voc->remaining_size = 0;
max_size -= 4;
break;
case VOC_TYPE_NEW_VOICE_DATA:
dec->sample_rate = avio_rl32(pb);
dec->bits_per_coded_sample = avio_r8(pb);
dec->channels = avio_r8(pb);
tmp_codec = avio_rl16(pb);
avio_skip(pb, 4);
voc->remaining_size -= 12;
max_size -= 12;
break;
default:
avio_skip(pb, voc->remaining_size);
max_size -= voc->remaining_size;
voc->remaining_size = 0;
break;
}
}
if (tmp_codec >= 0) {
tmp_codec = ff_codec_get_id(ff_voc_codec_tags, tmp_codec);
if (dec->codec_id == CODEC_ID_NONE)
dec->codec_id = tmp_codec;
else if (dec->codec_id != tmp_codec)
av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in audio codec\n");
if (dec->codec_id == CODEC_ID_NONE) {
if (s->audio_codec_id == CODEC_ID_NONE) {
av_log(s, AV_LOG_ERROR, "unknown codec tag\n");
return AVERROR(EINVAL);
}
av_log(s, AV_LOG_WARNING, "unknown codec tag\n");
}
}
dec->bit_rate = dec->sample_rate * dec->channels * dec->bits_per_coded_sample;
if (max_size <= 0)
max_size = 2048;
size = FFMIN(voc->remaining_size, max_size);
voc->remaining_size -= size;
return av_get_packet(pb, pkt, size);
}
static int voc_read_packet(AVFormatContext *s, AVPacket *pkt)
{
return ff_voc_get_packet(s, pkt, s->streams[0], 0);
}
AVInputFormat ff_voc_demuxer = {
.name = "voc",
.long_name = NULL_IF_CONFIG_SMALL("Creative Voice file format"),
.priv_data_size = sizeof(VocDecContext),
.read_probe = voc_probe,
.read_header = voc_read_header,
.read_packet = voc_read_packet,
.codec_tag=(const AVCodecTag* const []){ff_voc_codec_tags, 0},
};