/*
  * 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
  */

 #include "libavcodec/get_bits.h"

 #include "avformat.h"
 #include "mpegts.h"

 #include "url.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

          'ffio_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);

 
     ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
     ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_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 >= 4) {
         payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
 

         switch (buf[1]) {
         case RTCP_SR:

             if (payload_len < 20) {

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

             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;
             }

 
             break;

         case RTCP_BYE:
             return -RTCP_BYE;

         }

 
         buf += payload_len;
         len -= payload_len;

     }

     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 (avio_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 = avio_close_dyn_buf(pb, &buf);

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

         int av_unused result;

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

         result= ffurl_write(s->rtp_ctx, buf, len);
         av_dlog(s->ic, "result from ffurl_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 (avio_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 = avio_close_dyn_buf(pb, &buf);

     if ((len > 0) && buf)

         ffurl_write(rtp_handle, buf, len);

     av_free(buf);
 
     /* Send a minimal RTCP RR */

     if (avio_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 = avio_close_dyn_buf(pb, &buf);

     if ((len > 0) && buf)

         ffurl_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;
 }