PeerStreamer

/*

 * RTP input format

 * Copyright (c) 2002 Fabrice Bellard

 *

 * 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

 */

/* needed for gethostname() */

#define _XOPEN_SOURCE 600

#include "libavcodec/get_bits.h"

#include "avformat.h"

#include "mpegts.h"

#include <unistd.h>

#include <strings.h>

#include "network.h"

#include "rtpdec.h"

#include "rtpdec_formats.h"

//#define DEBUG

/* TODO: - add RTCP statistics reporting (should be optional).

         - add support for h263/mpeg4 packetized output : IDEA: send a

         buffer to 'rtp_write_packet' contains all the packets for ONE

         frame. Each packet should have a four byte header containing

         the length in big endian format (same trick as

         'url_open_dyn_packet_buf')

*/

static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {

    .enc_name           = "X-MP3-draft-00",

    .codec_type         = AVMEDIA_TYPE_AUDIO,

    .codec_id           = CODEC_ID_MP3ADU,

};

/* statistics functions */

static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;

void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)

{

    handler->next= RTPFirstDynamicPayloadHandler;

    RTPFirstDynamicPayloadHandler= handler;

}

void av_register_rtp_dynamic_payload_handlers(void)

{

    ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);

    ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);

    ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);

    ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);

    ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);

    ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);

    ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);

}

RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,

                                                  enum AVMediaType codec_type)

{

    RTPDynamicProtocolHandler *handler;

    for (handler = RTPFirstDynamicPayloadHandler;

         handler; handler = handler->next)

        if (!strcasecmp(name, handler->enc_name) &&

            codec_type == handler->codec_type)

            return handler;

    return NULL;

}

RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,

                                                enum AVMediaType codec_type)

{

    RTPDynamicProtocolHandler *handler;

    for (handler = RTPFirstDynamicPayloadHandler;

         handler; handler = handler->next)

        if (handler->static_payload_id && handler->static_payload_id == id &&

            codec_type == handler->codec_type)

            return handler;

    return NULL;

}

static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)

{

    int payload_len;

    while (len >= 2) {

        switch (buf[1]) {

        case RTCP_SR:

            if (len < 16) {

                av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");

                return AVERROR_INVALIDDATA;

            }

            payload_len = (AV_RB16(buf + 2) + 1) * 4;

            s->last_rtcp_ntp_time = AV_RB64(buf + 8);

            s->last_rtcp_timestamp = AV_RB32(buf + 16);

            if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {

                s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;

                if (!s->base_timestamp)

                    s->base_timestamp = s->last_rtcp_timestamp;

                s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;

            }

            buf += payload_len;

            len -= payload_len;

            break;

        case RTCP_BYE:

            return -RTCP_BYE;

        default:

            return -1;

        }

    }

    return -1;

}

#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

int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)

{

    AVIOContext *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 !?

    if (rtcp_bytes < 28)

        return -1;

    s->last_octet_count = s->octet_count;

    if (url_open_dyn_buf(&pb) < 0)

        return -1;

    // Receiver Report

    avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */

    avio_w8(pb, RTCP_RR);

    avio_wb16(pb, 7); /* length in words - 1 */

    // our own SSRC: we use the server's SSRC + 1 to avoid conflicts

    avio_wb32(pb, s->ssrc + 1);

    avio_wb32(pb, s->ssrc); // server SSRC

    // 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;

    avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */

    avio_wb32(pb, extended_max); /* max sequence received */

    avio_wb32(pb, stats->jitter>>4); /* jitter */

    if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)

    {

        avio_wb32(pb, 0); /* last SR timestamp */

        avio_wb32(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;

        avio_wb32(pb, middle_32_bits); /* last SR timestamp */

        avio_wb32(pb, delay_since_last); /* delay since last SR */

    }

    // CNAME

    avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */

    avio_w8(pb, RTCP_SDES);

    len = strlen(s->hostname);

    avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */

    avio_wb32(pb, s->ssrc);

    avio_w8(pb, 0x01);

    avio_w8(pb, len);

    avio_write(pb, s->hostname, len);

    // padding

    for (len = (6 + len) % 4; len % 4; len++) {

        avio_w8(pb, 0);

    }

    avio_flush(pb);

    len = url_close_dyn_buf(pb, &buf);

    if ((len > 0) && buf) {

        int result;

        av_dlog(s->ic, "sending %d bytes of RR\n", len);

        result= url_write(s->rtp_ctx, buf, len);

        av_dlog(s->ic, "result from url_write: %d\n", result);

        av_free(buf);

    }

    return 0;

}

void rtp_send_punch_packets(URLContext* rtp_handle)

{

    AVIOContext *pb;

    uint8_t *buf;

    int len;

    /* Send a small RTP packet */

    if (url_open_dyn_buf(&pb) < 0)

        return;

    avio_w8(pb, (RTP_VERSION << 6));

    avio_w8(pb, 0); /* Payload type */

    avio_wb16(pb, 0); /* Seq */

    avio_wb32(pb, 0); /* Timestamp */

    avio_wb32(pb, 0); /* SSRC */

    avio_flush(pb);

    len = url_close_dyn_buf(pb, &buf);

    if ((len > 0) && buf)

        url_write(rtp_handle, buf, len);

    av_free(buf);

    /* Send a minimal RTCP RR */

    if (url_open_dyn_buf(&pb) < 0)

        return;

    avio_w8(pb, (RTP_VERSION << 6));

    avio_w8(pb, RTCP_RR); /* receiver report */

    avio_wb16(pb, 1); /* length in words - 1 */

    avio_wb32(pb, 0); /* our own SSRC */

    avio_flush(pb);

    len = url_close_dyn_buf(pb, &buf);

    if ((len > 0) && buf)

        url_write(rtp_handle, buf, len);

    av_free(buf);

}

/**

 * 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, URLContext *rtpc, int payload_type, int queue_size)

{

    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->queue_size = queue_size;

    rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?

    if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {

        s->ts = ff_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_H263:

        case CODEC_ID_H264:

            st->need_parsing = AVSTREAM_PARSE_FULL;

            break;

        case CODEC_ID_ADPCM_G722:

            /* According to RFC 3551, the stream clock rate is 8000

             * even if the sample rate is 16000. */

            if (st->codec->sample_rate == 8000)

                st->codec->sample_rate = 16000;

            break;

        default:

            break;

        }

    }

    // needed to send back RTCP RR in RTSP sessions

    s->rtp_ctx = rtpc;

    gethostname(s->hostname, sizeof(s->hostname));

    return s;

}

void

rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,

                               RTPDynamicProtocolHandler *handler)

{

    s->dynamic_protocol_context = ctx;

    s->parse_packet = handler->parse_packet;

}

/**

 * 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)

{

    if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)

        return; /* Timestamp already set by depacketizer */

    if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {

        int64_t addend;

        int delta_timestamp;

        /* compute pts from timestamp with received ntp_time */

        delta_timestamp = timestamp - s->last_rtcp_timestamp;

        /* convert to the PTS timebase */

        addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);

        pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +

                   delta_timestamp;

        return;

    }

    if (timestamp == RTP_NOTS_VALUE)

        return;

    if (!s->base_timestamp)

        s->base_timestamp = timestamp;

    pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;

}

static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,

                                     const uint8_t *buf, int len)

{

    unsigned int ssrc, h;

    int payload_type, seq, ret, flags = 0;

    int ext;

    AVStream *st;

    uint32_t timestamp;

    int rv= 0;

    ext = buf[0] & 0x10;

    payload_type = buf[1] & 0x7f;

    if (buf[1] & 0x80)

        flags |= RTP_FLAG_MARKER;

    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;

    }

    if (buf[0] & 0x20) {

        int padding = buf[len - 1];

        if (len >= 12 + padding)

            len -= padding;

    }

    s->seq = seq;

    len -= 12;

    buf += 12;

    /* RFC 3550 Section 5.3.1 RTP Header Extension handling */

    if (ext) {

        if (len < 4)

            return -1;

        /* calculate the header extension length (stored as number

         * of 32-bit words) */

        ext = (AV_RB16(buf + 2) + 1) << 2;

        if (len < ext)

            return -1;

        // skip past RTP header extension

        len -= ext;

        buf += ext;

    }

    if (!st) {

        /* specific MPEG2TS demux support */

        ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);

        /* The only error that can be returned from ff_mpegts_parse_packet

         * is "no more data to return from the provided buffer", so return

         * AVERROR(EAGAIN) for all errors */

        if (ret < 0)

            return AVERROR(EAGAIN);

        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;

        }

        return 0;

    } else if (s->parse_packet) {

        rv = s->parse_packet(s->ic, s->dynamic_protocol_context,

                             s->st, pkt, &timestamp, buf, len, flags);

    } 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:

        case CODEC_ID_MP3:

            /* 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:

        case CODEC_ID_MPEG2VIDEO:

            /* 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;

        default:

            av_new_packet(pkt, len);

            memcpy(pkt->data, buf, len);

            break;

        }

        pkt->stream_index = st->index;

    }

    // now perform timestamp things....

    finalize_packet(s, pkt, timestamp);

    return rv;

}

void ff_rtp_reset_packet_queue(RTPDemuxContext *s)

{

    while (s->queue) {

        RTPPacket *next = s->queue->next;

        av_free(s->queue->buf);

        av_free(s->queue);

        s->queue = next;

    }

    s->seq       = 0;

    s->queue_len = 0;

    s->prev_ret  = 0;

}

static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)

{

    uint16_t seq = AV_RB16(buf + 2);

    RTPPacket *cur = s->queue, *prev = NULL, *packet;

    /* Find the correct place in the queue to insert the packet */

    while (cur) {

        int16_t diff = seq - cur->seq;

        if (diff < 0)

            break;

        prev = cur;

        cur = cur->next;

    }

    packet = av_mallocz(sizeof(*packet));

    if (!packet)

        return;

    packet->recvtime = av_gettime();

    packet->seq = seq;

    packet->len = len;

    packet->buf = buf;

    packet->next = cur;

    if (prev)

        prev->next = packet;

    else

        s->queue = packet;

    s->queue_len++;

}

static int has_next_packet(RTPDemuxContext *s)

{

    return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);

}

int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)

{

    return s->queue ? s->queue->recvtime : 0;

}

static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)

{

    int rv;

    RTPPacket *next;

    if (s->queue_len <= 0)

        return -1;

    if (!has_next_packet(s))

        av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,

               "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);

    /* Parse the first packet in the queue, and dequeue it */

    rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);

    next = s->queue->next;

    av_free(s->queue->buf);

    av_free(s->queue);

    s->queue = next;

    s->queue_len--;

    return rv;

}

static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,

                     uint8_t **bufptr, int len)

{

    uint8_t* buf = bufptr ? *bufptr : NULL;

    int ret, flags = 0;

    uint32_t timestamp;

    int rv= 0;

    if (!buf) {

        /* If parsing of the previous packet actually returned 0 or an error,

         * there's nothing more to be parsed from that packet, but we may have

         * indicated that we can return the next enqueued packet. */

        if (s->prev_ret <= 0)

            return rtp_parse_queued_packet(s, pkt);

        /* return the next packets, if any */

        if(s->st && s->parse_packet) {

            /* timestamp should be overwritten by parse_packet, if not,

             * the packet is left with pts == AV_NOPTS_VALUE */

            timestamp = RTP_NOTS_VALUE;

            rv= s->parse_packet(s->ic, s->dynamic_protocol_context,

                                s->st, pkt, &timestamp, NULL, 0, flags);

            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 AVERROR(EAGAIN);

            ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,

                                      s->read_buf_size - s->read_buf_index);

            if (ret < 0)

                return AVERROR(EAGAIN);

            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] >= RTCP_SR && buf[1] <= RTCP_APP) {

        return rtcp_parse_packet(s, buf, len);

    }

    if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {

        /* First packet, or no reordering */

        return rtp_parse_packet_internal(s, pkt, buf, len);

    } else {

        uint16_t seq = AV_RB16(buf + 2);

        int16_t diff = seq - s->seq;

        if (diff < 0) {

            /* Packet older than the previously emitted one, drop */

            av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,

                   "RTP: dropping old packet received too late\n");

            return -1;

        } else if (diff <= 1) {

            /* Correct packet */

            rv = rtp_parse_packet_internal(s, pkt, buf, len);

            return rv;

        } else {

            /* Still missing some packet, enqueue this one. */

            enqueue_packet(s, buf, len);

            *bufptr = NULL;

            /* Return the first enqueued packet if the queue is full,

             * even if we're missing something */

            if (s->queue_len >= s->queue_size)

                return rtp_parse_queued_packet(s, pkt);

            return -1;

        }

    }

}

/**

 * Parse an RTP or RTCP packet directly sent as a buffer.

 * @param s RTP parse context.

 * @param pkt returned packet

 * @param bufptr pointer to the 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,

                     uint8_t **bufptr, int len)

{

    int rv = rtp_parse_one_packet(s, pkt, bufptr, len);

    s->prev_ret = rv;

    while (rv == AVERROR(EAGAIN) && has_next_packet(s))

        rv = rtp_parse_queued_packet(s, pkt);

    return rv ? rv : has_next_packet(s);

}

void rtp_parse_close(RTPDemuxContext *s)

{

    ff_rtp_reset_packet_queue(s);

    if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {

        ff_mpegts_parse_close(s->ts);

    }

    av_free(s);

}

int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,

                  int (*parse_fmtp)(AVStream *stream,

                                    PayloadContext *data,

                                    char *attr, char *value))

{

    char attr[256];

    char *value;

    int res;

    int value_size = strlen(p) + 1;

    if (!(value = av_malloc(value_size))) {

        av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");

        return AVERROR(ENOMEM);

    }

    // remove protocol identifier

    while (*p && *p == ' ') p++; // strip spaces

    while (*p && *p != ' ') p++; // eat protocol identifier

    while (*p && *p == ' ') p++; // strip trailing spaces

    while (ff_rtsp_next_attr_and_value(&p,

                                       attr, sizeof(attr),

                                       value, value_size)) {

        res = parse_fmtp(stream, data, attr, value);

        if (res < 0 && res != AVERROR_PATCHWELCOME) {

            av_free(value);

            return res;

        }

    }

    av_free(value);

    return 0;

}