view westwood.c @ 3902:5f9bec099c69 libavformat

Add dynamic payload handlers to rdt.c. These follow the same API as the ones in rtpdec.c, so that they can be shared and used in the same way in rtsp.c. The handlers, since they are specific for RDT, are registered in rdt.c and a new registration function is thus called from allformats.c. The dynamic payload handler also implements RDT-specific SDP-line parsing for OpaqueData and StartTime, which are specific for RDT and needed for proper playback. OpaqueData contains one or a list ("MLTI") of "MDPR" chunks that can be parsed by the rmdec.c function ff_rm_read_mdpr_codecdata(). To use this function, we create a new rdt_demuxer, which has the same private data as the rm_demuxer. The resulting AVFormatContext created with _open_stream() can thus be used to call functions in the RM demuxer. See discussion in "Realmedia patch" thread on ML.

/*
 * Westwood Studios Multimedia Formats Demuxer (VQA, AUD)
 * Copyright (c) 2003 The ffmpeg Project
 *
 * 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 westwood.c
 * Westwood Studios VQA & AUD file demuxers
 * by Mike Melanson ([e-mail])
 * for more information on the Westwood file formats, visit:
 *   http://www.pcisys.net/~melanson/codecs/
 *   http://www.geocities.com/SiliconValley/8682/aud3.txt
 *
 * Implementation note: There is no definite file signature for AUD files.
 * The demuxer uses a probabilistic strategy for content detection. This
 * entails performing sanity checks on certain header values in order to
 * qualify a file. Refer to wsaud_probe() for the precise parameters.
 */

#include "avformat.h"

#define AUD_HEADER_SIZE 12
#define AUD_CHUNK_PREAMBLE_SIZE 8
#define AUD_CHUNK_SIGNATURE 0x0000DEAF

#define FORM_TAG MKBETAG('F', 'O', 'R', 'M')
#define WVQA_TAG MKBETAG('W', 'V', 'Q', 'A')
#define VQHD_TAG MKBETAG('V', 'Q', 'H', 'D')
#define FINF_TAG MKBETAG('F', 'I', 'N', 'F')
#define SND0_TAG MKBETAG('S', 'N', 'D', '0')
#define SND1_TAG MKBETAG('S', 'N', 'D', '1')
#define SND2_TAG MKBETAG('S', 'N', 'D', '2')
#define VQFR_TAG MKBETAG('V', 'Q', 'F', 'R')

/* don't know what these tags are for, but acknowledge their existence */
#define CINF_TAG MKBETAG('C', 'I', 'N', 'F')
#define CINH_TAG MKBETAG('C', 'I', 'N', 'H')
#define CIND_TAG MKBETAG('C', 'I', 'N', 'D')
#define PINF_TAG MKBETAG('P', 'I', 'N', 'F')
#define PINH_TAG MKBETAG('P', 'I', 'N', 'H')
#define PIND_TAG MKBETAG('P', 'I', 'N', 'D')
#define CMDS_TAG MKBETAG('C', 'M', 'D', 'S')

#define VQA_HEADER_SIZE 0x2A
#define VQA_FRAMERATE 15
#define VQA_VIDEO_PTS_INC (90000 / VQA_FRAMERATE)
#define VQA_PREAMBLE_SIZE 8

typedef struct WsAudDemuxContext {
    int audio_samplerate;
    int audio_channels;
    int audio_bits;
    int audio_type;
    int audio_stream_index;
    int64_t audio_frame_counter;
} WsAudDemuxContext;

typedef struct WsVqaDemuxContext {
    int audio_samplerate;
    int audio_channels;
    int audio_bits;

    int audio_stream_index;
    int video_stream_index;

    int64_t audio_frame_counter;
    int64_t video_pts;
} WsVqaDemuxContext;

static int wsaud_probe(AVProbeData *p)
{
    int field;

    /* Probabilistic content detection strategy: There is no file signature
     * so perform sanity checks on various header parameters:
     *   8000 <= sample rate (16 bits) <= 48000  ==> 40001 acceptable numbers
     *   flags <= 0x03 (2 LSBs are used)         ==> 4 acceptable numbers
     *   compression type (8 bits) = 1 or 99     ==> 2 acceptable numbers
     *   first audio chunk signature (32 bits)   ==> 1 acceptable number
     * The number space contains 2^64 numbers. There are 40001 * 4 * 2 * 1 =
     * 320008 acceptable number combinations.
     */

    if (p->buf_size < AUD_HEADER_SIZE + AUD_CHUNK_PREAMBLE_SIZE)
        return 0;

    /* check sample rate */
    field = AV_RL16(&p->buf[0]);
    if ((field < 8000) || (field > 48000))
        return 0;

    /* enforce the rule that the top 6 bits of this flags field are reserved (0);
     * this might not be true, but enforce it until deemed unnecessary */
    if (p->buf[10] & 0xFC)
        return 0;

    /* note: only check for WS IMA (type 99) right now since there is no
     * support for type 1 */
    if (p->buf[11] != 99)
        return 0;

    /* read ahead to the first audio chunk and validate the first header signature */
    if (AV_RL32(&p->buf[16]) != AUD_CHUNK_SIGNATURE)
        return 0;

    /* return 1/2 certainty since this file check is a little sketchy */
    return AVPROBE_SCORE_MAX / 2;
}

static int wsaud_read_header(AVFormatContext *s,
                             AVFormatParameters *ap)
{
    WsAudDemuxContext *wsaud = s->priv_data;
    ByteIOContext *pb = s->pb;
    AVStream *st;
    unsigned char header[AUD_HEADER_SIZE];

    if (get_buffer(pb, header, AUD_HEADER_SIZE) != AUD_HEADER_SIZE)
        return AVERROR(EIO);
    wsaud->audio_samplerate = AV_RL16(&header[0]);
    if (header[11] == 99)
        wsaud->audio_type = CODEC_ID_ADPCM_IMA_WS;
    else
        return AVERROR_INVALIDDATA;

    /* flag 0 indicates stereo */
    wsaud->audio_channels = (header[10] & 0x1) + 1;
    /* flag 1 indicates 16 bit audio */
    wsaud->audio_bits = (((header[10] & 0x2) >> 1) + 1) * 8;

    /* initialize the audio decoder stream */
    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);
    av_set_pts_info(st, 33, 1, wsaud->audio_samplerate);
    st->codec->codec_type = CODEC_TYPE_AUDIO;
    st->codec->codec_id = wsaud->audio_type;
    st->codec->codec_tag = 0;  /* no tag */
    st->codec->channels = wsaud->audio_channels;
    st->codec->sample_rate = wsaud->audio_samplerate;
    st->codec->bits_per_sample = wsaud->audio_bits;
    st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
        st->codec->bits_per_sample / 4;
    st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;

    wsaud->audio_stream_index = st->index;
    wsaud->audio_frame_counter = 0;

    return 0;
}

static int wsaud_read_packet(AVFormatContext *s,
                             AVPacket *pkt)
{
    WsAudDemuxContext *wsaud = s->priv_data;
    ByteIOContext *pb = s->pb;
    unsigned char preamble[AUD_CHUNK_PREAMBLE_SIZE];
    unsigned int chunk_size;
    int ret = 0;

    if (get_buffer(pb, preamble, AUD_CHUNK_PREAMBLE_SIZE) !=
        AUD_CHUNK_PREAMBLE_SIZE)
        return AVERROR(EIO);

    /* validate the chunk */
    if (AV_RL32(&preamble[4]) != AUD_CHUNK_SIGNATURE)
        return AVERROR_INVALIDDATA;

    chunk_size = AV_RL16(&preamble[0]);
    ret= av_get_packet(pb, pkt, chunk_size);
    if (ret != chunk_size)
        return AVERROR(EIO);
    pkt->stream_index = wsaud->audio_stream_index;
    pkt->pts = wsaud->audio_frame_counter;
    pkt->pts /= wsaud->audio_samplerate;

    /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
    wsaud->audio_frame_counter += (chunk_size * 2) / wsaud->audio_channels;

    return ret;
}

static int wsvqa_probe(AVProbeData *p)
{
    /* need 12 bytes to qualify */
    if (p->buf_size < 12)
        return 0;

    /* check for the VQA signatures */
    if ((AV_RB32(&p->buf[0]) != FORM_TAG) ||
        (AV_RB32(&p->buf[8]) != WVQA_TAG))
        return 0;

    return AVPROBE_SCORE_MAX;
}

static int wsvqa_read_header(AVFormatContext *s,
                             AVFormatParameters *ap)
{
    WsVqaDemuxContext *wsvqa = s->priv_data;
    ByteIOContext *pb = s->pb;
    AVStream *st;
    unsigned char *header;
    unsigned char scratch[VQA_PREAMBLE_SIZE];
    unsigned int chunk_tag;
    unsigned int chunk_size;

    /* initialize the video decoder stream */
    st = av_new_stream(s, 0);
    if (!st)
        return AVERROR(ENOMEM);
    av_set_pts_info(st, 33, 1, VQA_FRAMERATE);
    wsvqa->video_stream_index = st->index;
    st->codec->codec_type = CODEC_TYPE_VIDEO;
    st->codec->codec_id = CODEC_ID_WS_VQA;
    st->codec->codec_tag = 0;  /* no fourcc */

    /* skip to the start of the VQA header */
    url_fseek(pb, 20, SEEK_SET);

    /* the VQA header needs to go to the decoder */
    st->codec->extradata_size = VQA_HEADER_SIZE;
    st->codec->extradata = av_mallocz(VQA_HEADER_SIZE + FF_INPUT_BUFFER_PADDING_SIZE);
    header = (unsigned char *)st->codec->extradata;
    if (get_buffer(pb, st->codec->extradata, VQA_HEADER_SIZE) !=
        VQA_HEADER_SIZE) {
        av_free(st->codec->extradata);
        return AVERROR(EIO);
    }
    st->codec->width = AV_RL16(&header[6]);
    st->codec->height = AV_RL16(&header[8]);

    /* initialize the audio decoder stream for VQA v1 or nonzero samplerate */
    if (AV_RL16(&header[24]) || (AV_RL16(&header[0]) == 1 && AV_RL16(&header[2]) == 1)) {
        st = av_new_stream(s, 0);
        if (!st)
            return AVERROR(ENOMEM);
        av_set_pts_info(st, 33, 1, VQA_FRAMERATE);
        st->codec->codec_type = CODEC_TYPE_AUDIO;
        if (AV_RL16(&header[0]) == 1)
            st->codec->codec_id = CODEC_ID_WESTWOOD_SND1;
        else
            st->codec->codec_id = CODEC_ID_ADPCM_IMA_WS;
        st->codec->codec_tag = 0;  /* no tag */
        st->codec->sample_rate = AV_RL16(&header[24]);
        if (!st->codec->sample_rate)
            st->codec->sample_rate = 22050;
        st->codec->channels = header[26];
        if (!st->codec->channels)
            st->codec->channels = 1;
        st->codec->bits_per_sample = 16;
        st->codec->bit_rate = st->codec->channels * st->codec->sample_rate *
            st->codec->bits_per_sample / 4;
        st->codec->block_align = st->codec->channels * st->codec->bits_per_sample;

        wsvqa->audio_stream_index = st->index;
        wsvqa->audio_samplerate = st->codec->sample_rate;
        wsvqa->audio_channels = st->codec->channels;
        wsvqa->audio_frame_counter = 0;
    }

    /* there are 0 or more chunks before the FINF chunk; iterate until
     * FINF has been skipped and the file will be ready to be demuxed */
    do {
        if (get_buffer(pb, scratch, VQA_PREAMBLE_SIZE) != VQA_PREAMBLE_SIZE) {
            av_free(st->codec->extradata);
            return AVERROR(EIO);
        }
        chunk_tag = AV_RB32(&scratch[0]);
        chunk_size = AV_RB32(&scratch[4]);

        /* catch any unknown header tags, for curiousity */
        switch (chunk_tag) {
        case CINF_TAG:
        case CINH_TAG:
        case CIND_TAG:
        case PINF_TAG:
        case PINH_TAG:
        case PIND_TAG:
        case FINF_TAG:
        case CMDS_TAG:
            break;

        default:
            av_log (s, AV_LOG_ERROR, " note: unknown chunk seen (%c%c%c%c)\n",
                scratch[0], scratch[1],
                scratch[2], scratch[3]);
            break;
        }

        url_fseek(pb, chunk_size, SEEK_CUR);
    } while (chunk_tag != FINF_TAG);

    wsvqa->video_pts = wsvqa->audio_frame_counter = 0;

    return 0;
}

static int wsvqa_read_packet(AVFormatContext *s,
                             AVPacket *pkt)
{
    WsVqaDemuxContext *wsvqa = s->priv_data;
    ByteIOContext *pb = s->pb;
    int ret = -1;
    unsigned char preamble[VQA_PREAMBLE_SIZE];
    unsigned int chunk_type;
    unsigned int chunk_size;
    int skip_byte;

    while (get_buffer(pb, preamble, VQA_PREAMBLE_SIZE) == VQA_PREAMBLE_SIZE) {
        chunk_type = AV_RB32(&preamble[0]);
        chunk_size = AV_RB32(&preamble[4]);
        skip_byte = chunk_size & 0x01;

        if ((chunk_type == SND1_TAG) || (chunk_type == SND2_TAG) || (chunk_type == VQFR_TAG)) {

            if (av_new_packet(pkt, chunk_size))
                return AVERROR(EIO);
            ret = get_buffer(pb, pkt->data, chunk_size);
            if (ret != chunk_size) {
                av_free_packet(pkt);
                return AVERROR(EIO);
            }

            if (chunk_type == SND2_TAG) {
                pkt->stream_index = wsvqa->audio_stream_index;
                /* 2 samples/byte, 1 or 2 samples per frame depending on stereo */
                wsvqa->audio_frame_counter += (chunk_size * 2) / wsvqa->audio_channels;
            } else if(chunk_type == SND1_TAG) {
                pkt->stream_index = wsvqa->audio_stream_index;
                /* unpacked size is stored in header */
                wsvqa->audio_frame_counter += AV_RL16(pkt->data) / wsvqa->audio_channels;
            } else {
                pkt->stream_index = wsvqa->video_stream_index;
                wsvqa->video_pts += VQA_VIDEO_PTS_INC;
            }
            /* stay on 16-bit alignment */
            if (skip_byte)
                url_fseek(pb, 1, SEEK_CUR);

            return ret;
        } else {
            switch(chunk_type){
            case CMDS_TAG:
            case SND0_TAG:
                break;
            default:
                av_log(s, AV_LOG_INFO, "Skipping unknown chunk 0x%08X\n", chunk_type);
            }
            url_fseek(pb, chunk_size + skip_byte, SEEK_CUR);
        }
    }

    return ret;
}

#ifdef CONFIG_WSAUD_DEMUXER
AVInputFormat wsaud_demuxer = {
    "wsaud",
    NULL_IF_CONFIG_SMALL("Westwood Studios audio format"),
    sizeof(WsAudDemuxContext),
    wsaud_probe,
    wsaud_read_header,
    wsaud_read_packet,
};
#endif
#ifdef CONFIG_WSVQA_DEMUXER
AVInputFormat wsvqa_demuxer = {
    "wsvqa",
    NULL_IF_CONFIG_SMALL("Westwood Studios VQA format"),
    sizeof(WsVqaDemuxContext),
    wsvqa_probe,
    wsvqa_read_header,
    wsvqa_read_packet,
};
#endif