comparison rtp.c @ 1445:db97355877b1 libavformat

add valid statistics for the RTCP receiver report. Basically taken verbatim from RFC 1889. Patch by Ryan Martell % rdm4 A martellventures P com % Original thread: Date: Oct 31, 2006 12:43 AM Subject: [Ffmpeg-devel] [PATCH] RTCP valid receiver statistics....

1444:74cb68ad9dce

        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
    s->last_rtcp_timestamp = decode_be32(buf + 16);
    return 0;
}

/**
 * some rtp servers assume client is dead if they don't hear from them...
 * so we send a Receiver Report to the provided ByteIO context
 * (we don't have access to the rtcp handle from here)
 */

{
    ByteIOContext pb;
    uint8_t *buf;
    int len;
    int rtcp_bytes;

    if (!s->rtp_ctx || (count < 1))
        return -1;

    /* 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 !?

    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...
    put_be32(&pb, ((0 << 24) | (0 & 0x0ffffff))); /* 0% lost, total 0 lost */
    put_be32(&pb, (0 << 16) | s->seq);
    put_be32(&pb, 0x68); /* jitter */
    put_be32(&pb, -1); /* last SR timestamp */
    put_be32(&pb, 1); /* delay since last SR */

    // CNAME
    put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
    put_byte(&pb, 202);
    len = strlen(s->hostname);

    }

    put_flush_packet(&pb);
    len = url_close_dyn_buf(&pb, &buf);
    if ((len > 0) && buf) {
#if defined(DEBUG)
        printf("sending %d bytes of RR\n", len);
#endif
        url_write(s->rtp_ctx, buf, len);
        av_free(buf);
    }
    return 0;
}

    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;
    if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
        s->ts = mpegts_parse_open(s->ic);
        if (s->ts == NULL) {
            av_free(s);
            return NULL;

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

    st = s->st;
#if defined(DEBUG) || 1
    if (seq != ((s->seq + 1) & 0xffff)) {
        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));
    }
#endif
    s->seq = seq;
    len -= 12;
    buf += 12;

    if (!st) {

1445:db97355877b1

        s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
    s->last_rtcp_timestamp = decode_be32(buf + 16);
    return 0;
}

#define RTP_SEQ_MOD (1<<16)

/**
* called on parse open packet
*/
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
{
    memset(s, 0, sizeof(RTPStatistics));
    s->max_seq= base_sequence;
    s->probation= 1;
}

/**
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
*/
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
{
    s->max_seq= seq;
    s->cycles= 0;
    s->base_seq= seq -1;
    s->bad_seq= RTP_SEQ_MOD + 1;
    s->received= 0;
    s->expected_prior= 0;
    s->received_prior= 0;
    s->jitter= 0;
    s->transit= 0;
}

/**
* returns 1 if we should handle this packet.
*/
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
{
    uint16_t udelta= seq - s->max_seq;
    const int MAX_DROPOUT= 3000;
    const int MAX_MISORDER = 100;
    const int MIN_SEQUENTIAL = 2;

    /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
    if(s->probation)
    {
        if(seq==s->max_seq + 1) {
            s->probation--;
            s->max_seq= seq;
            if(s->probation==0) {
                rtp_init_sequence(s, seq);
                s->received++;
                return 1;
            }
        } else {
            s->probation= MIN_SEQUENTIAL - 1;
            s->max_seq = seq;
        }
    } else if (udelta < MAX_DROPOUT) {
        // in order, with permissible gap
        if(seq < s->max_seq) {
            //sequence number wrapped; count antother 64k cycles
            s->cycles += RTP_SEQ_MOD;
        }
        s->max_seq= seq;
    } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
        // sequence made a large jump...
        if(seq==s->bad_seq) {
            // two sequential packets-- assume that the other side restarted without telling us; just resync.
            rtp_init_sequence(s, seq);
        } else {
            s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
            return 0;
        }
    } else {
        // duplicate or reordered packet...
    }
    s->received++;
    return 1;
}

#if 0
/**
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
* never change.  I left this in in case someone else can see a way. (rdm)
*/
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
{
    uint32_t transit= arrival_timestamp - sent_timestamp;
    int d;
    s->transit= transit;
    d= FFABS(transit - s->transit);
    s->jitter += d - ((s->jitter + 8)>>4);
}
#endif

/**
 * some rtp servers assume client is dead if they don't hear from them...
 * so we send a Receiver Report to the provided ByteIO context
 * (we don't have access to the rtcp handle from here)
 */

{
    ByteIOContext pb;
    uint8_t *buf;
    int len;
    int rtcp_bytes;
    RTPStatistics *stats= &s->statistics;
    uint32_t lost;
    uint32_t extended_max;
    uint32_t expected_interval;
    uint32_t received_interval;
    uint32_t lost_interval;
    uint32_t expected;
    uint32_t fraction;
    uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?

    if (!s->rtp_ctx || (count < 1))
        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 !?

    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_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;
}

    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;

    /* 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) {