view iss.c @ 4633:0c69b895a01f libavformat

Don't let finalize_packet() touch pkt->stream_index. Instead, let individual payload handlers take care of that themselves at their own option. What this patch really does is "fix" a bug in MS-RTSP protocol where incoming packets are always coming in over the connection (UDP) or interleave-id (TCP) of the stream-id of the first ASF packet in the RTP packet. However, RTP packets may contain multiple ASF packets (and usually do, from what I can see), and therefore this leads to playback bugs. The intended stream-id per ASF packet is given in the respective ASF packet header. The ASF demuxer will correctly read this and set pkt->stream_index, but since the "stream" parameter can not be known to rtpdec.c or any of the RTP/RTSP code, the "st" parameter in all these functions is basically invalid. Therefore, using st->id as pkt->stream_index leads to various playback bugs. The result of this patch is that pkt->stream_index is left untouched for RTP/ASF (and possibly for other payloads that have similar behaviour). The patch was discussed in the "[PATCH] rtpdec.c: don't overwrite pkt->stream_index in finalize_packet()" thread on the mailinglist.
author rbultje
date Tue, 03 Mar 2009 13:51:34 +0000
parents 49c1d3b27727
children 536e5527c1e0
line wrap: on
line source

/*
 * ISS (.iss) file demuxer
 * Copyright (c) 2008 Jaikrishnan Menon <realityman@gmx.net>
 *
 * 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
 */

/**
 * @file libavformat/iss.c
 * Funcom ISS file demuxer
 * @author Jaikrishnan Menon
 * for more information on the .iss file format, visit:
 * http://wiki.multimedia.cx/index.php?title=FunCom_ISS
 */

#include "avformat.h"
#include "libavutil/avstring.h"

#define ISS_SIG "IMA_ADPCM_Sound"
#define ISS_SIG_LEN 15
#define MAX_TOKEN_SIZE 20

typedef struct {
    int packet_size;
    int sample_start_pos;
} IssDemuxContext;

static void get_token(ByteIOContext *s, char *buf, int maxlen)
{
    int i = 0;
    char c;

    while ((c = get_byte(s))) {
        if(c == ' ')
            break;
        if (i < maxlen-1)
            buf[i++] = c;
    }

    if(!c)
        get_byte(s);

    buf[i] = 0; /* Ensure null terminated, but may be truncated */
}

static int iss_probe(AVProbeData *p)
{
    if (strncmp(p->buf, ISS_SIG, ISS_SIG_LEN))
        return 0;

    return AVPROBE_SCORE_MAX;
}

static av_cold int iss_read_header(AVFormatContext *s, AVFormatParameters *ap)
{
    IssDemuxContext *iss = s->priv_data;
    ByteIOContext *pb = s->pb;
    AVStream *st;
    char token[MAX_TOKEN_SIZE];
    int stereo, rate_divisor;

    get_token(pb, token, sizeof(token)); //"IMA_ADPCM_Sound"
    get_token(pb, token, sizeof(token)); //packet size
    sscanf(token, "%d", &iss->packet_size);
    get_token(pb, token, sizeof(token)); //File ID
    get_token(pb, token, sizeof(token)); //out size
    get_token(pb, token, sizeof(token)); //stereo
    sscanf(token, "%d", &stereo);
    get_token(pb, token, sizeof(token)); //Unknown1
    get_token(pb, token, sizeof(token)); //RateDivisor
    sscanf(token, "%d", &rate_divisor);
    get_token(pb, token, sizeof(token)); //Unknown2
    get_token(pb, token, sizeof(token)); //Version ID
    get_token(pb, token, sizeof(token)); //Size

    iss->sample_start_pos = url_ftell(pb);

    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);
    st->codec->codec_type = CODEC_TYPE_AUDIO;
    st->codec->codec_id = CODEC_ID_ADPCM_IMA_ISS;
    st->codec->channels = stereo ? 2 : 1;
    st->codec->sample_rate = 44100;
    if(rate_divisor > 0)
         st->codec->sample_rate /= rate_divisor;
    st->codec->bits_per_coded_sample = 4;
    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate
                                      * st->codec->bits_per_coded_sample;
    st->codec->block_align = iss->packet_size;
    av_set_pts_info(st, 32, 1, st->codec->sample_rate);

    return 0;
}

static int iss_read_packet(AVFormatContext *s, AVPacket *pkt)
{
    IssDemuxContext *iss = s->priv_data;
    int ret = av_get_packet(s->pb, pkt, iss->packet_size);

    if(ret != iss->packet_size)
        return AVERROR(EIO);

    pkt->stream_index = 0;
    pkt->pts = url_ftell(s->pb) - iss->sample_start_pos;
    if(s->streams[0]->codec->channels > 0)
        pkt->pts /= s->streams[0]->codec->channels*2;
    return 0;
}

AVInputFormat iss_demuxer = {
    "ISS",
    NULL_IF_CONFIG_SMALL("Funcom ISS format"),
    sizeof(IssDemuxContext),
    iss_probe,
    iss_read_header,
    iss_read_packet,
};