mirror of https://git.ffmpeg.org/ffmpeg.git
120 lines
4.0 KiB
C
120 lines
4.0 KiB
C
/*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avformat.h"
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#include "internal.h"
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#include "voc.h"
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int
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ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
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{
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VocDecContext *voc = s->priv_data;
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AVCodecParameters *par = st->codecpar;
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AVIOContext *pb = s->pb;
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VocType type;
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int size, tmp_codec=-1;
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int sample_rate = 0;
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int channels = 1;
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while (!voc->remaining_size) {
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type = avio_r8(pb);
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if (type == VOC_TYPE_EOF)
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return AVERROR(EIO);
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voc->remaining_size = avio_rl24(pb);
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if (!voc->remaining_size) {
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if (!(s->pb->seekable & AVIO_SEEKABLE_NORMAL))
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return AVERROR(EIO);
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voc->remaining_size = avio_size(pb) - avio_tell(pb);
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}
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max_size -= 4;
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switch (type) {
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case VOC_TYPE_VOICE_DATA:
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if (!par->sample_rate) {
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par->sample_rate = 1000000 / (256 - avio_r8(pb));
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if (sample_rate)
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par->sample_rate = sample_rate;
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avpriv_set_pts_info(st, 64, 1, par->sample_rate);
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par->channels = channels;
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par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id);
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} else
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avio_skip(pb, 1);
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tmp_codec = avio_r8(pb);
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voc->remaining_size -= 2;
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max_size -= 2;
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channels = 1;
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break;
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case VOC_TYPE_VOICE_DATA_CONT:
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break;
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case VOC_TYPE_EXTENDED:
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sample_rate = avio_rl16(pb);
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avio_r8(pb);
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channels = avio_r8(pb) + 1;
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sample_rate = 256000000 / (channels * (65536 - sample_rate));
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voc->remaining_size = 0;
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max_size -= 4;
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break;
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case VOC_TYPE_NEW_VOICE_DATA:
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if (!par->sample_rate) {
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par->sample_rate = avio_rl32(pb);
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avpriv_set_pts_info(st, 64, 1, par->sample_rate);
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par->bits_per_coded_sample = avio_r8(pb);
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par->channels = avio_r8(pb);
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} else
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avio_skip(pb, 6);
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tmp_codec = avio_rl16(pb);
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avio_skip(pb, 4);
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voc->remaining_size -= 12;
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max_size -= 12;
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break;
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default:
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avio_skip(pb, voc->remaining_size);
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max_size -= voc->remaining_size;
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voc->remaining_size = 0;
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break;
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}
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}
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if (tmp_codec >= 0) {
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tmp_codec = ff_codec_get_id(ff_voc_codec_tags, tmp_codec);
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if (par->codec_id == AV_CODEC_ID_NONE)
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par->codec_id = tmp_codec;
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else if (par->codec_id != tmp_codec)
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av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in audio codec\n");
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if (par->codec_id == AV_CODEC_ID_NONE) {
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if (s->audio_codec_id == AV_CODEC_ID_NONE) {
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av_log(s, AV_LOG_ERROR, "unknown codec tag\n");
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return AVERROR(EINVAL);
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}
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av_log(s, AV_LOG_WARNING, "unknown codec tag\n");
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}
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}
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par->bit_rate = par->sample_rate * par->bits_per_coded_sample;
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if (max_size <= 0)
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max_size = 2048;
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size = FFMIN(voc->remaining_size, max_size);
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voc->remaining_size -= size;
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return av_get_packet(pb, pkt, size);
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}
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