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mirror of https://git.videolan.org/git/ffmpeg.git synced 2024-09-10 01:30:21 +02:00
ffmpeg/libavformat/vocdec.c
Martin Storsjö 5bbfe193a0 vocdec: Don't update codec parameters mid-stream
If we really want to support parameter changes, they need to be
signalled along with the AVPackets as parameter change side data,
not just changing the AVCodecContext parameters when a packet
is demuxed (since there may be other earlier packets yet undecoded).

Something similar was already done for the sample rate in 0883109b2,
but some parameters were left changeable.

This avoids having to recheck the channel count for validity for
each decoded frame in (ad)pcm decoders, unless the decoders
explicitly say that they accept parameter changes.

Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
2013-09-16 11:04:52 +03:00

176 lines
5.4 KiB
C

/*
* Creative Voice File demuxer.
* Copyright (c) 2006 Aurelien Jacobs <aurel@gnuage.org>
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; 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:
if (!dec->sample_rate) {
dec->sample_rate = 1000000 / (256 - avio_r8(pb));
if (sample_rate)
dec->sample_rate = sample_rate;
avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
dec->channels = channels;
dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
} else
avio_skip(pb, 1);
tmp_codec = avio_r8(pb);
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:
if (!dec->sample_rate) {
dec->sample_rate = avio_rl32(pb);
avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
dec->bits_per_coded_sample = avio_r8(pb);
dec->channels = avio_r8(pb);
} else
avio_skip(pb, 6);
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 == AV_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 == AV_CODEC_ID_NONE) {
if (s->audio_codec_id == AV_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->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"),
.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 },
};