comparison rtp.c @ 294:6091b76cfc2a libavformat

added MPEG2TS support in RTP, SDP and RTSP - replaced fake RTP demux by a specific API

293:62cec412a186

 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */
#include "avformat.h"

#include <unistd.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>

  RTCP_SDES_IMG    =  9,
  RTCP_SDES_DOOR   = 10,
  RTCP_SDES_SOURCE = 11
} rtcp_sdes_type_t;

enum RTPPayloadType {
    RTP_PT_ULAW = 0,
    RTP_PT_GSM = 3,
    RTP_PT_G723 = 4,
    RTP_PT_ALAW = 8,
    RTP_PT_S16BE_STEREO = 10,
    RTP_PT_S16BE_MONO = 11,
    RTP_PT_MPEGAUDIO = 14,
    RTP_PT_JPEG = 26,
    RTP_PT_H261 = 31,
    RTP_PT_MPEGVIDEO = 32,
    RTP_PT_MPEG2TS = 33,
    RTP_PT_H263 = 34, /* old H263 encapsulation */
    RTP_PT_PRIVATE = 96,
};

typedef struct RTPContext {
    int payload_type;
    uint32_t ssrc;
    uint16_t seq;
    uint32_t timestamp;
    uint32_t base_timestamp;
    uint32_t cur_timestamp;
    int max_payload_size;
    /* rtcp sender statistics receive */
    int64_t last_rtcp_ntp_time;
    int64_t first_rtcp_ntp_time;
    uint32_t last_rtcp_timestamp;
    /* rtcp sender statistics */

    unsigned int last_octet_count;
    int first_packet;
    /* buffer for output */
    uint8_t buf[RTP_MAX_PACKET_LENGTH];
    uint8_t *buf_ptr;
} RTPContext;

int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
    switch(payload_type) {
    case RTP_PT_ULAW:
        codec->codec_id = CODEC_ID_PCM_MULAW;
        codec->channels = 1;
        codec->sample_rate = 8000;
        break;
    case RTP_PT_ALAW:
        codec->codec_id = CODEC_ID_PCM_ALAW;
        codec->channels = 1;
        codec->sample_rate = 8000;
        break;
    case RTP_PT_S16BE_STEREO:
        codec->codec_id = CODEC_ID_PCM_S16BE;
        codec->channels = 2;
        codec->sample_rate = 44100;
        break;
    case RTP_PT_S16BE_MONO:
        codec->codec_id = CODEC_ID_PCM_S16BE;
        codec->channels = 1;
        codec->sample_rate = 44100;
        break;
    case RTP_PT_MPEGAUDIO:
        codec->codec_id = CODEC_ID_MP2;
        break;
    case RTP_PT_JPEG:
        codec->codec_id = CODEC_ID_MJPEG;
        break;
    case RTP_PT_MPEGVIDEO:
        codec->codec_id = CODEC_ID_MPEG1VIDEO;
        break;
    default:
        return -1;
    }
    return 0;

        payload_type = RTP_PT_JPEG;
        break;
    case CODEC_ID_MPEG1VIDEO:
        payload_type = RTP_PT_MPEGVIDEO;
        break;
    default:
        break;
    }
    return payload_type;
}

static inline uint64_t decode_be64(const uint8_t *p)
{
    return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
}

static int rtcp_parse_packet(AVFormatContext *s1, const unsigned char *buf, int len)
{
    RTPContext *s = s1->priv_data;

    if (buf[1] != 200)
        return -1;
    s->last_rtcp_ntp_time = decode_be64(buf + 8);
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
    s->last_rtcp_timestamp = decode_be32(buf + 16);
    return 0;
}

/**
 * Parse an RTP packet directly sent as raw data. Can only be used if
 * 'raw' is given as input file
 * @param s1 media file context
 * @param pkt returned packet
 * @param buf input buffer
 * @param len buffer len
 * @return zero if no error.
 */
int rtp_parse_packet(AVFormatContext *s1, AVPacket *pkt, 
                     const unsigned char *buf, int len)
{
    RTPContext *s = s1->priv_data;
    unsigned int ssrc, h;
    int payload_type, seq, delta_timestamp;
    AVStream *st;
    uint32_t timestamp;
    
    if (len < 12)
        return -1;

    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
        return -1;
    if (buf[1] >= 200 && buf[1] <= 204) {
        rtcp_parse_packet(s1, buf, len);
        return -1;
    }
    payload_type = buf[1] & 0x7f;
    seq  = (buf[2] << 8) | buf[3];
    timestamp = decode_be32(buf + 4);
    ssrc = decode_be32(buf + 8);
    
    if (s->payload_type < 0) {
        s->payload_type = payload_type;
        
        if (payload_type == RTP_PT_MPEG2TS) {
            /* XXX: special case : not a single codec but a whole stream */
            return -1;
        } else {
            st = av_new_stream(s1, 0);
            if (!st)
                return -1;
            rtp_get_codec_info(&st->codec, payload_type);
        }
    }

    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;
#if defined(DEBUG) || 1
    if (seq != ((s->seq + 1) & 0xffff)) {

    }
    s->seq = seq;
#endif
    len -= 12;
    buf += 12;
    st = s1->streams[0];
    switch(st->codec.codec_id) {
    case CODEC_ID_MP2:
        /* better than nothing: skip mpeg audio RTP header */
        if (len <= 4)
            return -1;
        h = decode_be32(buf);
        len -= 4;
        buf += 4;
        av_new_packet(pkt, len);
        memcpy(pkt->data, buf, len);
        break;
    case CODEC_ID_MPEG1VIDEO:
        /* better than nothing: skip mpeg audio RTP header */
        if (len <= 4)
            return -1;
        h = decode_be32(buf);
        buf += 4;
        len -= 4;
        if (h & (1 << 26)) {
            /* mpeg2 */
            if (len <= 4)
                return -1;
            buf += 4;
            len -= 4;
        }
        av_new_packet(pkt, len);
        memcpy(pkt->data, buf, len);
        break;
    default:
        av_new_packet(pkt, len);
        memcpy(pkt->data, buf, len);
        break;
    }

    switch(st->codec.codec_id) {
    case CODEC_ID_MP2:
    case CODEC_ID_MPEG1VIDEO:
        if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
            int64_t addend;
            /* XXX: is it really necessary to unify the timestamp base ? */
            /* compute pts from timestamp with received ntp_time */
            delta_timestamp = timestamp - s->last_rtcp_timestamp;
            /* convert to 90 kHz without overflow */
            addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
            addend = (addend * 5625) >> 14;
            pkt->pts = addend + delta_timestamp;
        }
        break;
    default:
        /* no timestamp info yet */
        break;
    }
    return 0;
}

static int rtp_read_header(AVFormatContext *s1,
                           AVFormatParameters *ap)
{
    RTPContext *s = s1->priv_data;
    s->payload_type = -1;
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
    return 0;
}

static int rtp_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
    char buf[RTP_MAX_PACKET_LENGTH];
    int ret;

    /* XXX: needs a better API for packet handling ? */
    for(;;) {
        ret = url_read(url_fileno(&s1->pb), buf, sizeof(buf));
        if (ret < 0)
            return AVERROR_IO;
        if (rtp_parse_packet(s1, pkt, buf, ret) == 0)
            break;
    }
    return 0;
}

static int rtp_read_close(AVFormatContext *s1)
{
    //    RTPContext *s = s1->priv_data;
    return 0;
}

static int rtp_probe(AVProbeData *p)
{
    if (strstart(p->filename, "rtp://", NULL))
        return AVPROBE_SCORE_MAX;
    return 0;
}

/* rtp output */

static int rtp_write_header(AVFormatContext *s1)
{
    RTPContext *s = s1->priv_data;
    int payload_type, max_packet_size;
    AVStream *st;

    if (s1->nb_streams != 1)
        return -1;
    st = s1->streams[0];

        s->cur_timestamp = 0;
        break;
    case CODEC_ID_MPEG1VIDEO:
        s->cur_timestamp = 0;
        break;
    default:
        s->buf_ptr = s->buf;
        break;
    }

}

/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
    RTPContext *s = s1->priv_data;
#if defined(DEBUG)
    printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
#endif
    put_byte(&s1->pb, (RTP_VERSION << 6));
    put_byte(&s1->pb, 200);

/* send an rtp packet. sequence number is incremented, but the caller
   must update the timestamp itself */
static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
{
    RTPContext *s = s1->priv_data;

#ifdef DEBUG
    printf("rtp_send_data size=%d\n", len);
#endif

/* send an integer number of samples and compute time stamp and fill
   the rtp send buffer before sending. */
static void rtp_send_samples(AVFormatContext *s1,
                             const uint8_t *buf1, int size, int sample_size)
{
    RTPContext *s = s1->priv_data;
    int len, max_packet_size, n;

    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
    /* not needed, but who nows */
    if ((size % sample_size) != 0)

/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
                               const uint8_t *buf1, int size)
{
    RTPContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, count, max_packet_size;

    max_packet_size = s->max_payload_size;

/* NOTE: a single frame must be passed with sequence header if
   needed. XXX: use slices. */
static void rtp_send_mpegvideo(AVFormatContext *s1,
                               const uint8_t *buf1, int size)
{
    RTPContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, h, max_packet_size;
    uint8_t *q;

    max_packet_size = s->max_payload_size;

}

static void rtp_send_raw(AVFormatContext *s1,
                         const uint8_t *buf1, int size)
{
    RTPContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, max_packet_size;

    max_packet_size = s->max_payload_size;

        size -= len;
    }
    s->cur_timestamp++;
}

/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, int stream_index,
                            const uint8_t *buf1, int size, int64_t pts)
{
    RTPContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int rtcp_bytes;
    int64_t ntp_time;
    
#ifdef DEBUG

        rtp_send_mpegaudio(s1, buf1, size);
        break;
    case CODEC_ID_MPEG1VIDEO:
        rtp_send_mpegvideo(s1, buf1, size);
        break;
    default:
        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, buf1, size);
        break;
    }
    return 0;
}

static int rtp_write_trailer(AVFormatContext *s1)
{
    //    RTPContext *s = s1->priv_data;
    return 0;
}

AVInputFormat rtp_demux = {
    "rtp",
    "RTP input format",
    sizeof(RTPContext),    
    rtp_probe,
    rtp_read_header,
    rtp_read_packet,
    rtp_read_close,
    .flags = AVFMT_NOHEADER,
};

AVOutputFormat rtp_mux = {
    "rtp",
    "RTP output format",
    NULL,
    NULL,
    sizeof(RTPContext),
    CODEC_ID_PCM_MULAW,
    CODEC_ID_NONE,
    rtp_write_header,
    rtp_write_packet,
    rtp_write_trailer,
};

int rtp_init(void)
{
    av_register_output_format(&rtp_mux);
    av_register_input_format(&rtp_demux);
    return 0;
}

294:6091b76cfc2a

 * You should have received a copy of the GNU Lesser General Public
 * License along with this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */
#include "avformat.h"
#include "mpegts.h"

#include <unistd.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>

  RTCP_SDES_IMG    =  9,
  RTCP_SDES_DOOR   = 10,
  RTCP_SDES_SOURCE = 11
} rtcp_sdes_type_t;

struct RTPDemuxContext {
    AVFormatContext *ic;
    AVStream *st;
    int payload_type;
    uint32_t ssrc;
    uint16_t seq;
    uint32_t timestamp;
    uint32_t base_timestamp;
    uint32_t cur_timestamp;
    int max_payload_size;
    MpegTSContext *ts; /* only used for RTP_PT_MPEG2TS payloads */
    int read_buf_index;
    int read_buf_size;
    
    /* rtcp sender statistics receive */
    int64_t last_rtcp_ntp_time;
    int64_t first_rtcp_ntp_time;
    uint32_t last_rtcp_timestamp;
    /* rtcp sender statistics */

    unsigned int last_octet_count;
    int first_packet;
    /* buffer for output */
    uint8_t buf[RTP_MAX_PACKET_LENGTH];
    uint8_t *buf_ptr;
};

int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
{
    switch(payload_type) {
    case RTP_PT_ULAW:
        codec->codec_type = CODEC_TYPE_AUDIO;
        codec->codec_id = CODEC_ID_PCM_MULAW;
        codec->channels = 1;
        codec->sample_rate = 8000;
        break;
    case RTP_PT_ALAW:
        codec->codec_type = CODEC_TYPE_AUDIO;
        codec->codec_id = CODEC_ID_PCM_ALAW;
        codec->channels = 1;
        codec->sample_rate = 8000;
        break;
    case RTP_PT_S16BE_STEREO:
        codec->codec_type = CODEC_TYPE_AUDIO;
        codec->codec_id = CODEC_ID_PCM_S16BE;
        codec->channels = 2;
        codec->sample_rate = 44100;
        break;
    case RTP_PT_S16BE_MONO:
        codec->codec_type = CODEC_TYPE_AUDIO;
        codec->codec_id = CODEC_ID_PCM_S16BE;
        codec->channels = 1;
        codec->sample_rate = 44100;
        break;
    case RTP_PT_MPEGAUDIO:
        codec->codec_type = CODEC_TYPE_AUDIO;
        codec->codec_id = CODEC_ID_MP2;
        break;
    case RTP_PT_JPEG:
        codec->codec_type = CODEC_TYPE_VIDEO;
        codec->codec_id = CODEC_ID_MJPEG;
        break;
    case RTP_PT_MPEGVIDEO:
        codec->codec_type = CODEC_TYPE_VIDEO;
        codec->codec_id = CODEC_ID_MPEG1VIDEO;
        break;
    case RTP_PT_MPEG2TS:
        codec->codec_type = CODEC_TYPE_DATA;
        codec->codec_id = CODEC_ID_MPEG2TS;
        break;
    default:
        return -1;
    }
    return 0;

        payload_type = RTP_PT_JPEG;
        break;
    case CODEC_ID_MPEG1VIDEO:
        payload_type = RTP_PT_MPEGVIDEO;
        break;
    case CODEC_ID_MPEG2TS:
        payload_type = RTP_PT_MPEG2TS;
        break;
    default:
        break;
    }
    return payload_type;
}

static inline uint64_t decode_be64(const uint8_t *p)
{
    return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
}

static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
    if (buf[1] != 200)
        return -1;
    s->last_rtcp_ntp_time = decode_be64(buf + 8);
    if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
    s->last_rtcp_timestamp = decode_be32(buf + 16);
    return 0;
}

/**
 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
 * MPEG2TS streams to indicate that they should be demuxed inside the
 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) 
 */
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type)
{
    RTPDemuxContext *s;

    s = av_mallocz(sizeof(RTPDemuxContext));
    if (!s)
        return NULL;
    s->payload_type = payload_type;
    s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
    s->ic = s1;
    s->st = st;
    if (payload_type == RTP_PT_MPEG2TS) {
        s->ts = mpegts_parse_open(s->ic);
        if (s->ts == NULL) {
            av_free(s);
            return NULL;
        }
    }
    return s;
}

/**
 * Parse an RTP or RTCP packet directly sent as a buffer. 
 * @param s RTP parse context.
 * @param pkt returned packet
 * @param buf input buffer or NULL to read the next packets
 * @param len buffer len
 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow 
 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
 */
int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, 
                     const uint8_t *buf, int len)
{
    unsigned int ssrc, h;
    int payload_type, seq, delta_timestamp, ret;
    AVStream *st;
    uint32_t timestamp;
    
    if (!buf) {
        /* return the next packets, if any */
        if (s->read_buf_index >= s->read_buf_size)
            return -1;
        ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, 
                                  s->read_buf_size - s->read_buf_index);
        if (ret < 0)
            return -1;
        s->read_buf_index += ret;
        if (s->read_buf_index < s->read_buf_size)
            return 1;
        else
            return 0;
    }

    if (len < 12)
        return -1;

    if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
        return -1;
    if (buf[1] >= 200 && buf[1] <= 204) {
        rtcp_parse_packet(s, buf, len);
        return -1;
    }
    payload_type = buf[1] & 0x7f;
    seq  = (buf[2] << 8) | buf[3];
    timestamp = decode_be32(buf + 4);
    ssrc = decode_be32(buf + 8);
    
    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;
#if defined(DEBUG) || 1
    if (seq != ((s->seq + 1) & 0xffff)) {

    }
    s->seq = seq;
#endif
    len -= 12;
    buf += 12;

    st = s->st;
    if (!st) {
        /* specific MPEG2TS demux support */
        ret = mpegts_parse_packet(s->ts, pkt, buf, len);
        if (ret < 0)
            return -1;
        if (ret < len) {
            s->read_buf_size = len - ret;
            memcpy(s->buf, buf + ret, s->read_buf_size);
            s->read_buf_index = 0;
            return 1;
        }
    } else {
        switch(st->codec.codec_id) {
        case CODEC_ID_MP2:
            /* better than nothing: skip mpeg audio RTP header */
            if (len <= 4)
                return -1;
            h = decode_be32(buf);
            len -= 4;
            buf += 4;
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        case CODEC_ID_MPEG1VIDEO:
            /* better than nothing: skip mpeg audio RTP header */
            if (len <= 4)
                return -1;
            h = decode_be32(buf);
            buf += 4;
            len -= 4;
            if (h & (1 << 26)) {
                /* mpeg2 */
                if (len <= 4)
                    return -1;
                buf += 4;
                len -= 4;
            }
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        default:
            av_new_packet(pkt, len);
            memcpy(pkt->data, buf, len);
            break;
        }
        
        switch(st->codec.codec_id) {
        case CODEC_ID_MP2:
        case CODEC_ID_MPEG1VIDEO:
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
                int64_t addend;
                /* XXX: is it really necessary to unify the timestamp base ? */
                /* compute pts from timestamp with received ntp_time */
                delta_timestamp = timestamp - s->last_rtcp_timestamp;
                /* convert to 90 kHz without overflow */
                addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
                addend = (addend * 5625) >> 14;
                pkt->pts = addend + delta_timestamp;
            }
            break;
        default:
            /* no timestamp info yet */
            break;
        }
        pkt->stream_index = s->st->index;
    }
    return 0;
}

void rtp_parse_close(RTPDemuxContext *s)
{
    if (s->payload_type == RTP_PT_MPEG2TS) {
        mpegts_parse_close(s->ts);
    }
    av_free(s);
}

/* rtp output */

static int rtp_write_header(AVFormatContext *s1)
{
    RTPDemuxContext *s = s1->priv_data;
    int payload_type, max_packet_size, n;
    AVStream *st;

    if (s1->nb_streams != 1)
        return -1;
    st = s1->streams[0];

        s->cur_timestamp = 0;
        break;
    case CODEC_ID_MPEG1VIDEO:
        s->cur_timestamp = 0;
        break;
    case CODEC_ID_MPEG2TS:
        n = s->max_payload_size / TS_PACKET_SIZE;
        if (n < 1)
            n = 1;
        s->max_payload_size = n * TS_PACKET_SIZE;
        s->buf_ptr = s->buf;
        break;
    default:
        s->buf_ptr = s->buf;
        break;
    }

}

/* send an rtcp sender report packet */
static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
{
    RTPDemuxContext *s = s1->priv_data;
#if defined(DEBUG)
    printf("RTCP: %02x %Lx %x\n", s->payload_type, ntp_time, s->timestamp);
#endif
    put_byte(&s1->pb, (RTP_VERSION << 6));
    put_byte(&s1->pb, 200);

/* send an rtp packet. sequence number is incremented, but the caller
   must update the timestamp itself */
static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
{
    RTPDemuxContext *s = s1->priv_data;

#ifdef DEBUG
    printf("rtp_send_data size=%d\n", len);
#endif

/* send an integer number of samples and compute time stamp and fill
   the rtp send buffer before sending. */
static void rtp_send_samples(AVFormatContext *s1,
                             const uint8_t *buf1, int size, int sample_size)
{
    RTPDemuxContext *s = s1->priv_data;
    int len, max_packet_size, n;

    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
    /* not needed, but who nows */
    if ((size % sample_size) != 0)

/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
static void rtp_send_mpegaudio(AVFormatContext *s1,
                               const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, count, max_packet_size;

    max_packet_size = s->max_payload_size;

/* NOTE: a single frame must be passed with sequence header if
   needed. XXX: use slices. */
static void rtp_send_mpegvideo(AVFormatContext *s1,
                               const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, h, max_packet_size;
    uint8_t *q;

    max_packet_size = s->max_payload_size;

}

static void rtp_send_raw(AVFormatContext *s1,
                         const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int len, max_packet_size;

    max_packet_size = s->max_payload_size;

        size -= len;
    }
    s->cur_timestamp++;
}

/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
static void rtp_send_mpegts_raw(AVFormatContext *s1,
                                const uint8_t *buf1, int size)
{
    RTPDemuxContext *s = s1->priv_data;
    int len, out_len;

    while (size >= TS_PACKET_SIZE) {
        len = s->max_payload_size - (s->buf_ptr - s->buf);
        if (len > size)
            len = size;
        memcpy(s->buf_ptr, buf1, len);
        buf1 += len;
        size -= len;
        s->buf_ptr += len;
        
        out_len = s->buf_ptr - s->buf;
        if (out_len >= s->max_payload_size) {
            rtp_send_data(s1, s->buf, out_len);
            s->buf_ptr = s->buf;
        }
    }
}

/* write an RTP packet. 'buf1' must contain a single specific frame. */
static int rtp_write_packet(AVFormatContext *s1, int stream_index,
                            const uint8_t *buf1, int size, int64_t pts)
{
    RTPDemuxContext *s = s1->priv_data;
    AVStream *st = s1->streams[0];
    int rtcp_bytes;
    int64_t ntp_time;
    
#ifdef DEBUG

        rtp_send_mpegaudio(s1, buf1, size);
        break;
    case CODEC_ID_MPEG1VIDEO:
        rtp_send_mpegvideo(s1, buf1, size);
        break;
    case CODEC_ID_MPEG2TS:
        rtp_send_mpegts_raw(s1, buf1, size);
        break;
    default:
        /* better than nothing : send the codec raw data */
        rtp_send_raw(s1, buf1, size);
        break;
    }
    return 0;
}

static int rtp_write_trailer(AVFormatContext *s1)
{
    //    RTPDemuxContext *s = s1->priv_data;
    return 0;
}

AVOutputFormat rtp_mux = {
    "rtp",
    "RTP output format",
    NULL,
    NULL,
    sizeof(RTPDemuxContext),
    CODEC_ID_PCM_MULAW,
    CODEC_ID_NONE,
    rtp_write_header,
    rtp_write_packet,
    rtp_write_trailer,
};

int rtp_init(void)
{
    av_register_output_format(&rtp_mux);
    return 0;
}