annotate vocdec.c @ 4723:a2390c6a35e6 libavformat

Fix index generation in the way that it was supposed to be used. See the discussion in the ML thread "[PATCH] rmdec.c: merge old/new packet reading code". Over time, this code broke somewhat, e.g. seq was never actually written into (and was thus always 1, therefore the seq condition was always true), whereas it was supposed to be set to the sequence number of the video slice in case the video frame is divided over multiple RM packets (slices). The problem of this is that packets other than those containing the beginning of a video frame would be indexed as well. Secondly, flags&2 is supposed to be true for video keyframes and for these audio packets containing the start of a block. For some codecs (e.g. AAC), that is every single packet, whereas for others (e.g. cook), that is the packet containing the first of a series of scrambled packets that are to be descrambled together. Indexing any of the following would lead to incomplete and thus useless frames. Problem here is that flags would be reset to 2 to indicate that the first packet is ready to be returned, and in addition if no data was left to be returned (which is always true for the first packet), then we wouldn't actually write the index entry anyway. All in all, the idea was good and it probably worked at some point, but that is long ago. This patch should at the very least make it likely for this code to be executed again at the right times, i.e. the way it was originally intended to be used.

/*
 * Creative Voice File demuxer.
 * Copyright (c) 2006  Aurelien Jacobs <[e-mail]>
 *
 * 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"


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, AVFormatParameters *ap)
{
    VocDecContext *voc = s->priv_data;
    ByteIOContext *pb = s->pb;
    int header_size;
    AVStream *st;

    url_fskip(pb, 20);
    header_size = get_le16(pb) - 22;
    if (header_size != 4) {
        av_log(s, AV_LOG_ERROR, "unknown header size: %d\n", header_size);
        return AVERROR(ENOSYS);
    }
    url_fskip(pb, header_size);
    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);
    st->codec->codec_type = CODEC_TYPE_AUDIO;

    voc->remaining_size = 0;
    return 0;
}

int
voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
{
    VocDecContext *voc = s->priv_data;
    AVCodecContext *dec = st->codec;
    ByteIOContext *pb = s->pb;
    VocType type;
    int size;
    int sample_rate = 0;
    int channels = 1;

    while (!voc->remaining_size) {
        type = get_byte(pb);
        if (type == VOC_TYPE_EOF)
            return AVERROR(EIO);
        voc->remaining_size = get_le24(pb);
        max_size -= 4;

        switch (type) {
        case VOC_TYPE_VOICE_DATA:
            dec->sample_rate = 1000000 / (256 - get_byte(pb));
            if (sample_rate)
                dec->sample_rate = sample_rate;
            dec->channels = channels;
            dec->codec_id = codec_get_id(ff_voc_codec_tags, get_byte(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 = get_le16(pb);
            get_byte(pb);
            channels = get_byte(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 = get_le32(pb);
            dec->bits_per_coded_sample = get_byte(pb);
            dec->channels = get_byte(pb);
            dec->codec_id = codec_get_id(ff_voc_codec_tags, get_le16(pb));
            url_fskip(pb, 4);
            voc->remaining_size -= 12;
            max_size -= 12;
            break;

        default:
            url_fskip(pb, voc->remaining_size);
            max_size -= voc->remaining_size;
            voc->remaining_size = 0;
            break;
        }
    }

    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 voc_get_packet(s, pkt, s->streams[0], 0);
}

AVInputFormat voc_demuxer = {
    "voc",
    NULL_IF_CONFIG_SMALL("Creative Voice file format"),
    sizeof(VocDecContext),
    voc_probe,
    voc_read_header,
    voc_read_packet,
    .codec_tag=(const AVCodecTag* const []){ff_voc_codec_tags, 0},
};