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/*
* 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 "avformat.h"
#include "internal.h"
#include "voc.h"
int
ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
{
VocDecContext *voc = s->priv_data;
AVCodecParameters *par = st->codecpar;
AVIOContext *pb = s->pb;
VocType type;
int size, tmp_codec=-1;
int sample_rate = 0;
int channels = 1;
int64_t duration;
int ret;
av_add_index_entry(st,
avio_tell(pb),
voc->pts,
voc->remaining_size,
0,
AVINDEX_KEYFRAME);
while (!voc->remaining_size) {
type = avio_r8(pb);
if (type == VOC_TYPE_EOF)
return AVERROR_EOF;
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 (!par->sample_rate) {
par->sample_rate = 1000000 / (256 - avio_r8(pb));
if (sample_rate)
par->sample_rate = sample_rate;
avpriv_set_pts_info(st, 64, 1, par->sample_rate);
par->channels = channels;
par->bits_per_coded_sample = av_get_bits_per_sample(par->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 (!par->sample_rate) {
par->sample_rate = avio_rl32(pb);
avpriv_set_pts_info(st, 64, 1, par->sample_rate);
par->bits_per_coded_sample = avio_r8(pb);
par->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 (par->sample_rate <= 0) {
av_log(s, AV_LOG_ERROR, "Invalid sample rate %d\n", par->sample_rate);
return AVERROR_INVALIDDATA;
}
if (tmp_codec >= 0) {
tmp_codec = ff_codec_get_id(ff_voc_codec_tags, tmp_codec);
if (par->codec_id == AV_CODEC_ID_NONE)
par->codec_id = tmp_codec;
else if (par->codec_id != tmp_codec)
av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in audio codec\n");
if (par->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");
}
}
par->bit_rate = (int64_t)par->sample_rate * par->channels * par->bits_per_coded_sample;
if (max_size <= 0)
max_size = 2048;
size = FFMIN(voc->remaining_size, max_size);
voc->remaining_size -= size;
ret = av_get_packet(pb, pkt, size);
pkt->dts = pkt->pts = voc->pts;
duration = av_get_audio_frame_duration2(st->codecpar, size);
if (duration > 0 && voc->pts != AV_NOPTS_VALUE)
voc->pts += duration;
else
voc->pts = AV_NOPTS_VALUE;
return ret;
}
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