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e37f161e66
* qatar/master: (71 commits) movenc: Allow writing to a non-seekable output if using empty moov movenc: Support adding isml (smooth streaming live) metadata libavcodec: Don't crash in avcodec_encode_audio if time_base isn't set sunrast: Document the different Sun Raster file format types. sunrast: Add a check for experimental type. libspeexenc: use AVSampleFormat instead of deprecated/removed SampleFormat lavf: remove disabled FF_API_SET_PTS_INFO cruft lavf: remove disabled FF_API_OLD_INTERRUPT_CB cruft lavf: remove disabled FF_API_REORDER_PRIVATE cruft lavf: remove disabled FF_API_SEEK_PUBLIC cruft lavf: remove disabled FF_API_STREAM_COPY cruft lavf: remove disabled FF_API_PRELOAD cruft lavf: remove disabled FF_API_NEW_STREAM cruft lavf: remove disabled FF_API_RTSP_URL_OPTIONS cruft lavf: remove disabled FF_API_MUXRATE cruft lavf: remove disabled FF_API_FILESIZE cruft lavf: remove disabled FF_API_TIMESTAMP cruft lavf: remove disabled FF_API_LOOP_OUTPUT cruft lavf: remove disabled FF_API_LOOP_INPUT cruft lavf: remove disabled FF_API_AVSTREAM_QUALITY cruft ... Conflicts: doc/APIchanges libavcodec/8bps.c libavcodec/avcodec.h libavcodec/libx264.c libavcodec/mjpegbdec.c libavcodec/options.c libavcodec/sunrast.c libavcodec/utils.c libavcodec/version.h libavcodec/x86/h264_deblock.asm libavdevice/libdc1394.c libavdevice/v4l2.c libavformat/avformat.h libavformat/avio.c libavformat/avio.h libavformat/aviobuf.c libavformat/dv.c libavformat/mov.c libavformat/utils.c libavformat/version.h libavformat/wtv.c libavutil/Makefile libavutil/file.c libswscale/x86/input.asm libswscale/x86/swscale_mmx.c libswscale/x86/swscale_template.c tests/ref/lavf/ffm Merged-by: Michael Niedermayer <michaelni@gmx.at>
168 lines
5.0 KiB
C
168 lines
5.0 KiB
C
/*
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* Creative Voice File demuxer.
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* Copyright (c) 2006 Aurelien Jacobs <aurel@gnuage.org>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg 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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* FFmpeg 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 FFmpeg; 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 "libavutil/intreadwrite.h"
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#include "voc.h"
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#include "internal.h"
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static int voc_probe(AVProbeData *p)
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{
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int version, check;
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if (memcmp(p->buf, ff_voc_magic, sizeof(ff_voc_magic) - 1))
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return 0;
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version = AV_RL16(p->buf + 22);
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check = AV_RL16(p->buf + 24);
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if (~version + 0x1234 != check)
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return 10;
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return AVPROBE_SCORE_MAX;
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}
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static int voc_read_header(AVFormatContext *s)
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{
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VocDecContext *voc = s->priv_data;
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AVIOContext *pb = s->pb;
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int header_size;
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AVStream *st;
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avio_skip(pb, 20);
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header_size = avio_rl16(pb) - 22;
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if (header_size != 4) {
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av_log(s, AV_LOG_ERROR, "unknown header size: %d\n", header_size);
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return AVERROR(ENOSYS);
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}
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avio_skip(pb, header_size);
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st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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voc->remaining_size = 0;
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return 0;
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}
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int
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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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AVCodecContext *dec = st->codec;
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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)
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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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dec->sample_rate = 1000000 / (256 - avio_r8(pb));
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if (sample_rate)
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dec->sample_rate = sample_rate;
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dec->channels = channels;
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tmp_codec = avio_r8(pb);
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dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
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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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dec->sample_rate = avio_rl32(pb);
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dec->bits_per_coded_sample = avio_r8(pb);
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dec->channels = avio_r8(pb);
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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 (dec->codec_id == CODEC_ID_NONE)
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dec->codec_id = tmp_codec;
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else if (dec->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 (dec->codec_id == CODEC_ID_NONE) {
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if (s->audio_codec_id == 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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dec->bit_rate = dec->sample_rate * dec->channels * dec->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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static int voc_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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return voc_get_packet(s, pkt, s->streams[0], 0);
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}
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AVInputFormat ff_voc_demuxer = {
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.name = "voc",
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.long_name = NULL_IF_CONFIG_SMALL("Creative Voice file format"),
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.priv_data_size = sizeof(VocDecContext),
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.read_probe = voc_probe,
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.read_header = voc_read_header,
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.read_packet = voc_read_packet,
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.codec_tag=(const AVCodecTag* const []){ff_voc_codec_tags, 0},
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};
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