annotate iff.c @ 4874:5370b0c1653c libavformat

Increase the SDP buffer size (again!) and also increase the temporary buffer size of the fmtp parameter buffer. For Vorbis RT(S)P, these contain full Vorbis headers, which can be up to 12kb each, formatted in base64, so 16kb total. Patch required for proper Vorbis/RTP playback, submitted as GSoC qualification task in the thread "RTP/Vorbis payload implementation (GSoC qual task)" by Colin McQuillan m.niloc googlemail com.

/*
 * IFF (.iff) file demuxer
 * Copyright (c) 2008 Jaikrishnan Menon <[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
 */

/**
 * @file libavformat/iff.c
 * IFF file demuxer
 * by Jaikrishnan Menon
 * for more information on the .iff file format, visit:
 * http://wiki.multimedia.cx/index.php?title=IFF
 */

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

#define ID_8SVX       MKTAG('8','S','V','X')
#define ID_VHDR       MKTAG('V','H','D','R')
#define ID_ATAK       MKTAG('A','T','A','K')
#define ID_RLSE       MKTAG('R','L','S','E')
#define ID_CHAN       MKTAG('C','H','A','N')

#define ID_FORM       MKTAG('F','O','R','M')
#define ID_ANNO       MKTAG('A','N','N','O')
#define ID_AUTH       MKTAG('A','U','T','H')
#define ID_CHRS       MKTAG('C','H','R','S')
#define ID_COPYRIGHT  MKTAG('(','c',')',' ')
#define ID_CSET       MKTAG('C','S','E','T')
#define ID_FVER       MKTAG('F','V','E','R')
#define ID_NAME       MKTAG('N','A','M','E')
#define ID_TEXT       MKTAG('T','E','X','T')
#define ID_BODY       MKTAG('B','O','D','Y')

#define LEFT    2
#define RIGHT   4
#define STEREO  6

#define PACKET_SIZE 1024

typedef enum {COMP_NONE, COMP_FIB, COMP_EXP} svx8_compression_type;

typedef struct {
    uint32_t  body_size;
    uint32_t  sent_bytes;
    uint32_t  audio_frame_count;
} IffDemuxContext;


static void interleave_stereo(const uint8_t *src, uint8_t *dest, int size)
{
    uint8_t *end = dest + size;
    size = size>>1;

    while(dest < end) {
        *dest++ = *src;
        *dest++ = *(src+size);
        src++;
    }
}

static int iff_probe(AVProbeData *p)
{
    const uint8_t *d = p->buf;

    if ( AV_RL32(d)   == ID_FORM &&
         AV_RL32(d+8) == ID_8SVX)
        return AVPROBE_SCORE_MAX;
    return 0;
}

static int iff_read_header(AVFormatContext *s,
                           AVFormatParameters *ap)
{
    IffDemuxContext *iff = s->priv_data;
    ByteIOContext *pb = s->pb;
    AVStream *st;
    uint32_t chunk_id, data_size;
    int padding, done = 0;

    st = av_new_stream(s, 0);
    if (!st)
      return AVERROR(ENOMEM);

    st->codec->channels = 1;
    url_fskip(pb, 12);

    while(!done && !url_feof(pb)) {
        chunk_id = get_le32(pb);
        data_size = get_be32(pb);
        padding = data_size & 1;

        switch(chunk_id) {
        case ID_VHDR:
            url_fskip(pb, 12);
            st->codec->sample_rate = get_be16(pb);
            url_fskip(pb, 1);
            st->codec->codec_tag = get_byte(pb);
            url_fskip(pb, 4);
            break;

        case ID_BODY:
            iff->body_size = data_size;
            done = 1;
            break;

        case ID_CHAN:
            st->codec->channels = (get_be32(pb) < 6) ? 1 : 2;
            break;

        default:
            url_fseek(pb, data_size + padding, SEEK_CUR);
            break;
        }
    }

    if(!st->codec->sample_rate)
        return AVERROR_INVALIDDATA;

    av_set_pts_info(st, 32, 1, st->codec->sample_rate);
    st->codec->codec_type = CODEC_TYPE_AUDIO;

    switch(st->codec->codec_tag) {
    case COMP_NONE:
        st->codec->codec_id = CODEC_ID_PCM_S8;
        break;
    case COMP_FIB:
        st->codec->codec_id = CODEC_ID_8SVX_FIB;
        break;
    case COMP_EXP:
        st->codec->codec_id = CODEC_ID_8SVX_EXP;
        break;
    default:
        av_log(s, AV_LOG_ERROR, "iff: unknown compression method\n");
        return -1;
    }

    st->codec->bits_per_coded_sample = 8;
    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * st->codec->bits_per_coded_sample;
    st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample;

    return 0;
}

static int iff_read_packet(AVFormatContext *s,
                           AVPacket *pkt)
{
    IffDemuxContext *iff = s->priv_data;
    ByteIOContext *pb = s->pb;
    int ret;

    if(iff->sent_bytes > iff->body_size)
        return AVERROR(EIO);

    if(s->streams[0]->codec->channels == 2) {
        uint8_t sample_buffer[PACKET_SIZE];

        ret = get_buffer(pb, sample_buffer, PACKET_SIZE);
        if(av_new_packet(pkt, PACKET_SIZE) < 0) {
            av_log(s, AV_LOG_ERROR, "iff: cannot allocate packet \n");
            return AVERROR(ENOMEM);
        }
        interleave_stereo(sample_buffer, pkt->data, PACKET_SIZE);
    }
    else {
        ret = av_get_packet(pb, pkt, PACKET_SIZE);
    }

    if(iff->sent_bytes == 0)
        pkt->flags |= PKT_FLAG_KEY;

    iff->sent_bytes += PACKET_SIZE;
    pkt->stream_index = 0;
    pkt->pts = iff->audio_frame_count;
    iff->audio_frame_count += ret / s->streams[0]->codec->channels;
    return ret;
}

AVInputFormat iff_demuxer = {
    "IFF",
    NULL_IF_CONFIG_SMALL("IFF format"),
    sizeof(IffDemuxContext),
    iff_probe,
    iff_read_header,
    iff_read_packet,
};