rtp.c

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        return -1;

    /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
    /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
    s->octet_count += count;
    rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
        RTCP_TX_RATIO_DEN;
    rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
    if (rtcp_bytes < 28)
        return -1;
    s->last_octet_count = s->octet_count;

    if (url_open_dyn_buf(&pb) < 0)
        return -1;

    // Receiver Report
    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(&pb, 201);
    put_be16(&pb, 7); /* length in words - 1 */
    put_be32(&pb, s->ssrc); // our own SSRC
    put_be32(&pb, s->ssrc); // XXX: should be the server's here!
    // some placeholders we should really fill...
    // RFC 1889/p64
    extended_max= stats->cycles + stats->max_seq;
    expected= extended_max - stats->base_seq + 1;
    lost= expected - stats->received;
    lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
    expected_interval= expected - stats->expected_prior;
    stats->expected_prior= expected;
    received_interval= stats->received - stats->received_prior;
    stats->received_prior= stats->received;
    lost_interval= expected_interval - received_interval;
    if (expected_interval==0 || lost_interval<=0) fraction= 0;
    else fraction = (lost_interval<<8)/expected_interval;

    fraction= (fraction<<24) | lost;

    put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
    put_be32(&pb, extended_max); /* max sequence received */
    put_be32(&pb, stats->jitter>>4); /* jitter */

    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
    {
        put_be32(&pb, 0); /* last SR timestamp */
        put_be32(&pb, 0); /* delay since last SR */
    } else {
        uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
        uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;

        put_be32(&pb, middle_32_bits); /* last SR timestamp */
        put_be32(&pb, delay_since_last); /* delay since last SR */
    }

    // CNAME
    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(&pb, 202);
    len = strlen(s->hostname);
    put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
    put_be32(&pb, s->ssrc);
    put_byte(&pb, 0x01);
    put_byte(&pb, len);
    put_buffer(&pb, s->hostname, len);
    // padding
    for (len = (6 + len) % 4; len % 4; len++) {
        put_byte(&pb, 0);
    }

    put_flush_packet(&pb);
    len = url_close_dyn_buf(&pb, &buf);
    if ((len > 0) && buf) {
        int result;
#if defined(DEBUG)
        printf("sending %d bytes of RR\n", len);
#endif
        result= url_write(s->rtp_ctx, buf, len);
#if defined(DEBUG)
        printf("result from url_write: %d\n", result);
#endif
        av_free(buf);
    }
    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)
 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
 */
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
    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;
    s->rtp_payload_data = rtp_payload_data;
    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
    if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
        s->ts = mpegts_parse_open(s->ic);
        if (s->ts == NULL) {
            av_free(s);
            return NULL;
        }
    } else {
        switch(st->codec->codec_id) {
        case CODEC_ID_MPEG1VIDEO:
        case CODEC_ID_MPEG2VIDEO:
        case CODEC_ID_MP2:
        case CODEC_ID_MP3:
        case CODEC_ID_MPEG4:
        case CODEC_ID_H264:
            st->need_parsing = AVSTREAM_PARSE_FULL;
            break;
        default:
            break;
        }
    }
    // needed to send back RTCP RR in RTSP sessions
    s->rtp_ctx = rtpc;
    gethostname(s->hostname, sizeof(s->hostname));
    return s;
}

static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
{
    int au_headers_length, au_header_size, i;
    GetBitContext getbitcontext;
    rtp_payload_data_t *infos;

    infos = s->rtp_payload_data;

    if (infos == NULL)
        return -1;

    /* decode the first 2 bytes where are stored the AUHeader sections
       length in bits */
    au_headers_length = AV_RB16(buf);

    if (au_headers_length > RTP_MAX_PACKET_LENGTH)
      return -1;

    infos->au_headers_length_bytes = (au_headers_length + 7) / 8;

    /* skip AU headers length section (2 bytes) */
    buf += 2;

    init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);

    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
    au_header_size = infos->sizelength + infos->indexlength;
    if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
        return -1;

    infos->nb_au_headers = au_headers_length / au_header_size;
    infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);

    /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
       In my test, the FAAD decoder does not behave correctly when sending each AU one by one
       but does when sending the whole as one big packet...  */
    infos->au_headers[0].size = 0;
    infos->au_headers[0].index = 0;
    for (i = 0; i < infos->nb_au_headers; ++i) {
        infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
        infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
    }

    infos->nb_au_headers = 1;

    return 0;
}

/**
 * This was the second switch in rtp_parse packet.  Normalizes time, if required, sets stream_index, etc.
 */
static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
{
    switch(s->st->codec->codec_id) {
        case CODEC_ID_MP2:
        case CODEC_ID_MPEG1VIDEO:
            if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
                int64_t addend;

                int delta_timestamp;
                /* 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;
        case CODEC_ID_AAC:
        case CODEC_ID_H264:
        case CODEC_ID_MPEG4:
            pkt->pts = timestamp;
            break;
        default:
            /* no timestamp info yet */
            break;
    }
    pkt->stream_index = s->st->index;
}

/**
 * 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, ret;
    AVStream *st;
    uint32_t timestamp;
    int rv= 0;

    if (!buf) {
        /* return the next packets, if any */
        if(s->st && s->parse_packet) {
            timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
            rv= s->parse_packet(s, pkt, &timestamp, NULL, 0);
            finalize_packet(s, pkt, timestamp);
            return rv;
        } else {
            // TODO: Move to a dynamic packet handler (like above)
            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  = AV_RB16(buf + 2);
    timestamp = AV_RB32(buf + 4);
    ssrc = AV_RB32(buf + 8);
    /* store the ssrc in the RTPDemuxContext */
    s->ssrc = ssrc;

    /* NOTE: we can handle only one payload type */
    if (s->payload_type != payload_type)
        return -1;

    st = s->st;
    // only do something with this if all the rtp checks pass...
    if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
    {
        av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
               payload_type, seq, ((s->seq + 1) & 0xffff));
        return -1;
    }

    s->seq = seq;
    len -= 12;
    buf += 12;

    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 {
        // at this point, the RTP header has been stripped;  This is ASSUMING that there is only 1 CSRC, which in't wise.
        switch(st->codec->codec_id) {
        case CODEC_ID_MP2:
            /* better than nothing: skip mpeg audio RTP header */
            if (len <= 4)
                return -1;
            h = AV_RB32(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 video RTP header */
            if (len <= 4)
                return -1;
            h = AV_RB32(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;
            // moved from below, verbatim.  this is because this section handles packets, and the lower switch handles
            // timestamps.
            // TODO: Put this into a dynamic packet handler...
        case CODEC_ID_AAC:
            if (rtp_parse_mp4_au(s, buf))
                return -1;
            {
                rtp_payload_data_t *infos = s->rtp_payload_data;
                if (infos == NULL)
                    return -1;
                buf += infos->au_headers_length_bytes + 2;
                len -= infos->au_headers_length_bytes + 2;

                /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
                    one au_header */
                av_new_packet(pkt, infos->au_headers[0].size);
                memcpy(pkt->data, buf, infos->au_headers[0].size);
                buf += infos->au_headers[0].size;
                len -= infos->au_headers[0].size;
            }
            s->read_buf_size = len;
            rv= 0;