2 * RTP input/output format
3 * Copyright (c) 2002 Fabrice Bellard.
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 #include "bitstream.h"
26 #include <sys/types.h>
27 #include <sys/socket.h>
28 #include <netinet/in.h>
30 # include <arpa/inet.h>
32 # include "barpainet.h"
36 #include "rtp_internal.h"
42 /* TODO: - add RTCP statistics reporting (should be optional).
44 - add support for h263/mpeg4 packetized output : IDEA: send a
45 buffer to 'rtp_write_packet' contains all the packets for ONE
46 frame. Each packet should have a four byte header containing
47 the length in big endian format (same trick as
48 'url_open_dyn_packet_buf')
51 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
52 AVRtpPayloadType_t AVRtpPayloadTypes[]=
54 {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1},
55 {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
56 {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
57 {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
58 {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
59 {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
60 {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1},
61 {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
62 {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1},
63 {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
64 {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2},
65 {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1},
66 {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1},
67 {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
68 {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1},
69 {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
70 {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1},
71 {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1},
72 {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
73 {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
74 {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
75 {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
76 {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
77 {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
78 {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
79 {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
80 {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1},
81 {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
82 {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
83 {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
84 {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
85 {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1},
86 {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1},
87 {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1},
88 {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1},
89 {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
90 {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
91 {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
92 {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
93 {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
94 {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
95 {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
96 {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
97 {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
98 {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
99 {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
100 {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
101 {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
102 {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
103 {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
104 {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
105 {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
106 {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
107 {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
108 {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
109 {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
110 {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
111 {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
112 {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
113 {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
114 {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
115 {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
116 {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
117 {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
118 {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
119 {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
120 {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
121 {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
122 {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
123 {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
124 {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
125 {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
126 {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
127 {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
128 {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
129 {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
130 {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
131 {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
132 {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
133 {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
134 {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
135 {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
136 {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
137 {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
138 {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
139 {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
140 {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
141 {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
142 {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
143 {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
144 {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
145 {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
146 {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
147 {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
148 {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
149 {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
150 {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
151 {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
152 {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
153 {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
154 {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
155 {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
156 {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
157 {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
158 {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
159 {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
160 {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
161 {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
162 {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
163 {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
164 {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
165 {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
166 {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
167 {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
168 {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
169 {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
170 {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
171 {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
172 {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
173 {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
174 {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
175 {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
176 {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
177 {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
178 {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
179 {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
180 {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
181 {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
182 {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
185 /* statistics functions */
186 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
188 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
189 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
191 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
193 handler->next= RTPFirstDynamicPayloadHandler;
194 RTPFirstDynamicPayloadHandler= handler;
197 void av_register_rtp_dynamic_payload_handlers()
199 register_dynamic_payload_handler(&mp4v_es_handler);
200 register_dynamic_payload_handler(&mpeg4_generic_handler);
201 register_dynamic_payload_handler(&ff_h264_dynamic_handler);
204 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
206 if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
207 codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
208 codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
209 if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
210 codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
211 if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
212 codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
218 /* return < 0 if unknown payload type */
219 int rtp_get_payload_type(AVCodecContext *codec)
223 /* compute the payload type */
224 for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
225 if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
226 if (codec->codec_id == CODEC_ID_PCM_S16BE)
227 if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
229 payload_type = AVRtpPayloadTypes[i].pt;
234 static inline uint32_t decode_be32(const uint8_t *p)
236 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
239 static inline uint64_t decode_be64(const uint8_t *p)
241 return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
244 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
248 s->last_rtcp_ntp_time = decode_be64(buf + 8);
249 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
250 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
251 s->last_rtcp_timestamp = decode_be32(buf + 16);
255 #define RTP_SEQ_MOD (1<<16)
258 * called on parse open packet
260 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
262 memset(s, 0, sizeof(RTPStatistics));
263 s->max_seq= base_sequence;
268 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
270 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
275 s->bad_seq= RTP_SEQ_MOD + 1;
277 s->expected_prior= 0;
278 s->received_prior= 0;
284 * returns 1 if we should handle this packet.
286 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
288 uint16_t udelta= seq - s->max_seq;
289 const int MAX_DROPOUT= 3000;
290 const int MAX_MISORDER = 100;
291 const int MIN_SEQUENTIAL = 2;
293 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
296 if(seq==s->max_seq + 1) {
299 if(s->probation==0) {
300 rtp_init_sequence(s, seq);
305 s->probation= MIN_SEQUENTIAL - 1;
308 } else if (udelta < MAX_DROPOUT) {
309 // in order, with permissible gap
310 if(seq < s->max_seq) {
311 //sequence number wrapped; count antother 64k cycles
312 s->cycles += RTP_SEQ_MOD;
315 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
316 // sequence made a large jump...
317 if(seq==s->bad_seq) {
318 // two sequential packets-- assume that the other side restarted without telling us; just resync.
319 rtp_init_sequence(s, seq);
321 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
325 // duplicate or reordered packet...
333 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
334 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
335 * never change. I left this in in case someone else can see a way. (rdm)
337 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
339 uint32_t transit= arrival_timestamp - sent_timestamp;
342 d= FFABS(transit - s->transit);
343 s->jitter += d - ((s->jitter + 8)>>4);
348 * some rtp servers assume client is dead if they don't hear from them...
349 * so we send a Receiver Report to the provided ByteIO context
350 * (we don't have access to the rtcp handle from here)
352 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
358 RTPStatistics *stats= &s->statistics;
360 uint32_t extended_max;
361 uint32_t expected_interval;
362 uint32_t received_interval;
363 uint32_t lost_interval;
366 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
368 if (!s->rtp_ctx || (count < 1))
371 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
372 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
373 s->octet_count += count;
374 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
376 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
379 s->last_octet_count = s->octet_count;
381 if (url_open_dyn_buf(&pb) < 0)
385 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
387 put_be16(&pb, 7); /* length in words - 1 */
388 put_be32(&pb, s->ssrc); // our own SSRC
389 put_be32(&pb, s->ssrc); // XXX: should be the server's here!
390 // some placeholders we should really fill...
392 extended_max= stats->cycles + stats->max_seq;
393 expected= extended_max - stats->base_seq + 1;
394 lost= expected - stats->received;
395 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
396 expected_interval= expected - stats->expected_prior;
397 stats->expected_prior= expected;
398 received_interval= stats->received - stats->received_prior;
399 stats->received_prior= stats->received;
400 lost_interval= expected_interval - received_interval;
401 if (expected_interval==0 || lost_interval<=0) fraction= 0;
402 else fraction = (lost_interval<<8)/expected_interval;
404 fraction= (fraction<<24) | lost;
406 put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
407 put_be32(&pb, extended_max); /* max sequence received */
408 put_be32(&pb, stats->jitter>>4); /* jitter */
410 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
412 put_be32(&pb, 0); /* last SR timestamp */
413 put_be32(&pb, 0); /* delay since last SR */
415 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
416 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
418 put_be32(&pb, middle_32_bits); /* last SR timestamp */
419 put_be32(&pb, delay_since_last); /* delay since last SR */
423 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
425 len = strlen(s->hostname);
426 put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
427 put_be32(&pb, s->ssrc);
430 put_buffer(&pb, s->hostname, len);
432 for (len = (6 + len) % 4; len % 4; len++) {
436 put_flush_packet(&pb);
437 len = url_close_dyn_buf(&pb, &buf);
438 if ((len > 0) && buf) {
441 printf("sending %d bytes of RR\n", len);
443 result= url_write(s->rtp_ctx, buf, len);
445 printf("result from url_write: %d\n", result);
453 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
454 * MPEG2TS streams to indicate that they should be demuxed inside the
455 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
456 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
458 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
462 s = av_mallocz(sizeof(RTPDemuxContext));
465 s->payload_type = payload_type;
466 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
467 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
470 s->rtp_payload_data = rtp_payload_data;
471 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
472 if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
473 s->ts = mpegts_parse_open(s->ic);
479 switch(st->codec->codec_id) {
480 case CODEC_ID_MPEG1VIDEO:
481 case CODEC_ID_MPEG2VIDEO:
486 st->need_parsing = 1;
492 // needed to send back RTCP RR in RTSP sessions
494 gethostname(s->hostname, sizeof(s->hostname));
498 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
500 int au_headers_length, au_header_size, i;
501 GetBitContext getbitcontext;
502 rtp_payload_data_t *infos;
504 infos = s->rtp_payload_data;
509 /* decode the first 2 bytes where are stored the AUHeader sections
511 au_headers_length = BE_16(buf);
513 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
516 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
518 /* skip AU headers length section (2 bytes) */
521 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
523 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
524 au_header_size = infos->sizelength + infos->indexlength;
525 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
528 infos->nb_au_headers = au_headers_length / au_header_size;
529 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
531 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
532 In my test, the faad decoder doesnt behave correctly when sending each AU one by one
533 but does when sending the whole as one big packet... */
534 infos->au_headers[0].size = 0;
535 infos->au_headers[0].index = 0;
536 for (i = 0; i < infos->nb_au_headers; ++i) {
537 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
538 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
541 infos->nb_au_headers = 1;
547 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
549 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
551 switch(s->st->codec->codec_id) {
553 case CODEC_ID_MPEG1VIDEO:
554 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
558 /* XXX: is it really necessary to unify the timestamp base ? */
559 /* compute pts from timestamp with received ntp_time */
560 delta_timestamp = timestamp - s->last_rtcp_timestamp;
561 /* convert to 90 kHz without overflow */
562 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
563 addend = (addend * 5625) >> 14;
564 pkt->pts = addend + delta_timestamp;
570 pkt->pts = timestamp;
573 /* no timestamp info yet */
576 pkt->stream_index = s->st->index;
580 * Parse an RTP or RTCP packet directly sent as a buffer.
581 * @param s RTP parse context.
582 * @param pkt returned packet
583 * @param buf input buffer or NULL to read the next packets
584 * @param len buffer len
585 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
586 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
588 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
589 const uint8_t *buf, int len)
591 unsigned int ssrc, h;
592 int payload_type, seq, ret;
598 /* return the next packets, if any */
599 if(s->st && s->parse_packet) {
600 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
601 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0);
602 finalize_packet(s, pkt, timestamp);
605 // TODO: Move to a dynamic packet handler (like above)
606 if (s->read_buf_index >= s->read_buf_size)
608 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
609 s->read_buf_size - s->read_buf_index);
612 s->read_buf_index += ret;
613 if (s->read_buf_index < s->read_buf_size)
623 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
625 if (buf[1] >= 200 && buf[1] <= 204) {
626 rtcp_parse_packet(s, buf, len);
629 payload_type = buf[1] & 0x7f;
630 seq = (buf[2] << 8) | buf[3];
631 timestamp = decode_be32(buf + 4);
632 ssrc = decode_be32(buf + 8);
633 /* store the ssrc in the RTPDemuxContext */
636 /* NOTE: we can handle only one payload type */
637 if (s->payload_type != payload_type)
641 // only do something with this if all the rtp checks pass...
642 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
644 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
645 payload_type, seq, ((s->seq + 1) & 0xffff));
654 /* specific MPEG2TS demux support */
655 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
659 s->read_buf_size = len - ret;
660 memcpy(s->buf, buf + ret, s->read_buf_size);
661 s->read_buf_index = 0;
665 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
666 switch(st->codec->codec_id) {
668 /* better than nothing: skip mpeg audio RTP header */
671 h = decode_be32(buf);
674 av_new_packet(pkt, len);
675 memcpy(pkt->data, buf, len);
677 case CODEC_ID_MPEG1VIDEO:
678 /* better than nothing: skip mpeg video RTP header */
681 h = decode_be32(buf);
691 av_new_packet(pkt, len);
692 memcpy(pkt->data, buf, len);
694 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
696 // TODO: Put this into a dynamic packet handler...
698 if (rtp_parse_mp4_au(s, buf))
701 rtp_payload_data_t *infos = s->rtp_payload_data;
704 buf += infos->au_headers_length_bytes + 2;
705 len -= infos->au_headers_length_bytes + 2;
707 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
709 av_new_packet(pkt, infos->au_headers[0].size);
710 memcpy(pkt->data, buf, infos->au_headers[0].size);
711 buf += infos->au_headers[0].size;
712 len -= infos->au_headers[0].size;
714 s->read_buf_size = len;
719 if(s->parse_packet) {
720 rv= s->parse_packet(s, pkt, ×tamp, buf, len);
722 av_new_packet(pkt, len);
723 memcpy(pkt->data, buf, len);
728 // now perform timestamp things....
729 finalize_packet(s, pkt, timestamp);
734 void rtp_parse_close(RTPDemuxContext *s)
736 // TODO: fold this into the protocol specific data fields.
737 if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
738 mpegts_parse_close(s->ts);
745 static int rtp_write_header(AVFormatContext *s1)
747 RTPDemuxContext *s = s1->priv_data;
748 int payload_type, max_packet_size, n;
751 if (s1->nb_streams != 1)
755 payload_type = rtp_get_payload_type(st->codec);
756 if (payload_type < 0)
757 payload_type = RTP_PT_PRIVATE; /* private payload type */
758 s->payload_type = payload_type;
760 // following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
761 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
762 s->timestamp = s->base_timestamp;
763 s->ssrc = 0; /* FIXME: was random(), what should this be? */
766 max_packet_size = url_fget_max_packet_size(&s1->pb);
767 if (max_packet_size <= 12)
769 s->max_payload_size = max_packet_size - 12;
771 switch(st->codec->codec_id) {
774 s->buf_ptr = s->buf + 4;
775 s->cur_timestamp = 0;
777 case CODEC_ID_MPEG1VIDEO:
778 s->cur_timestamp = 0;
780 case CODEC_ID_MPEG2TS:
781 n = s->max_payload_size / TS_PACKET_SIZE;
784 s->max_payload_size = n * TS_PACKET_SIZE;
795 /* send an rtcp sender report packet */
796 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
798 RTPDemuxContext *s = s1->priv_data;
800 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
802 put_byte(&s1->pb, (RTP_VERSION << 6));
803 put_byte(&s1->pb, 200);
804 put_be16(&s1->pb, 6); /* length in words - 1 */
805 put_be32(&s1->pb, s->ssrc);
806 put_be64(&s1->pb, ntp_time);
807 put_be32(&s1->pb, s->timestamp);
808 put_be32(&s1->pb, s->packet_count);
809 put_be32(&s1->pb, s->octet_count);
810 put_flush_packet(&s1->pb);
813 /* send an rtp packet. sequence number is incremented, but the caller
814 must update the timestamp itself */
815 static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
817 RTPDemuxContext *s = s1->priv_data;
820 printf("rtp_send_data size=%d\n", len);
823 /* build the RTP header */
824 put_byte(&s1->pb, (RTP_VERSION << 6));
825 put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
826 put_be16(&s1->pb, s->seq);
827 put_be32(&s1->pb, s->timestamp);
828 put_be32(&s1->pb, s->ssrc);
830 put_buffer(&s1->pb, buf1, len);
831 put_flush_packet(&s1->pb);
834 s->octet_count += len;
838 /* send an integer number of samples and compute time stamp and fill
839 the rtp send buffer before sending. */
840 static void rtp_send_samples(AVFormatContext *s1,
841 const uint8_t *buf1, int size, int sample_size)
843 RTPDemuxContext *s = s1->priv_data;
844 int len, max_packet_size, n;
846 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
847 /* not needed, but who nows */
848 if ((size % sample_size) != 0)
851 len = (max_packet_size - (s->buf_ptr - s->buf));
856 memcpy(s->buf_ptr, buf1, len);
860 n = (s->buf_ptr - s->buf);
861 /* if buffer full, then send it */
862 if (n >= max_packet_size) {
863 rtp_send_data(s1, s->buf, n, 0);
865 /* update timestamp */
866 s->timestamp += n / sample_size;
871 /* NOTE: we suppose that exactly one frame is given as argument here */
873 static void rtp_send_mpegaudio(AVFormatContext *s1,
874 const uint8_t *buf1, int size)
876 RTPDemuxContext *s = s1->priv_data;
877 AVStream *st = s1->streams[0];
878 int len, count, max_packet_size;
880 max_packet_size = s->max_payload_size;
882 /* test if we must flush because not enough space */
883 len = (s->buf_ptr - s->buf);
884 if ((len + size) > max_packet_size) {
886 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
887 s->buf_ptr = s->buf + 4;
888 /* 90 KHz time stamp */
889 s->timestamp = s->base_timestamp +
890 (s->cur_timestamp * 90000LL) / st->codec->sample_rate;
895 if (size > max_packet_size) {
896 /* big packet: fragment */
899 len = max_packet_size - 4;
902 /* build fragmented packet */
905 s->buf[2] = count >> 8;
907 memcpy(s->buf + 4, buf1, len);
908 rtp_send_data(s1, s->buf, len + 4, 0);
914 if (s->buf_ptr == s->buf + 4) {
915 /* no fragmentation possible */
921 memcpy(s->buf_ptr, buf1, size);
924 s->cur_timestamp += st->codec->frame_size;
927 /* NOTE: a single frame must be passed with sequence header if
928 needed. XXX: use slices. */
929 static void rtp_send_mpegvideo(AVFormatContext *s1,
930 const uint8_t *buf1, int size)
932 RTPDemuxContext *s = s1->priv_data;
933 AVStream *st = s1->streams[0];
934 int len, h, max_packet_size;
937 max_packet_size = s->max_payload_size;
940 /* XXX: more correct headers */
942 if (st->codec->sub_id == 2)
943 h |= 1 << 26; /* mpeg 2 indicator */
950 if (st->codec->sub_id == 2) {
958 len = max_packet_size - (q - s->buf);
962 memcpy(q, buf1, len);
965 /* 90 KHz time stamp */
966 s->timestamp = s->base_timestamp +
967 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
968 rtp_send_data(s1, s->buf, q - s->buf, (len == size));
976 static void rtp_send_raw(AVFormatContext *s1,
977 const uint8_t *buf1, int size)
979 RTPDemuxContext *s = s1->priv_data;
980 AVStream *st = s1->streams[0];
981 int len, max_packet_size;
983 max_packet_size = s->max_payload_size;
986 len = max_packet_size;
990 /* 90 KHz time stamp */
991 s->timestamp = s->base_timestamp +
992 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
993 rtp_send_data(s1, buf1, len, (len == size));
1001 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
1002 static void rtp_send_mpegts_raw(AVFormatContext *s1,
1003 const uint8_t *buf1, int size)
1005 RTPDemuxContext *s = s1->priv_data;
1008 while (size >= TS_PACKET_SIZE) {
1009 len = s->max_payload_size - (s->buf_ptr - s->buf);
1012 memcpy(s->buf_ptr, buf1, len);
1017 out_len = s->buf_ptr - s->buf;
1018 if (out_len >= s->max_payload_size) {
1019 rtp_send_data(s1, s->buf, out_len, 0);
1020 s->buf_ptr = s->buf;
1025 /* write an RTP packet. 'buf1' must contain a single specific frame. */
1026 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
1028 RTPDemuxContext *s = s1->priv_data;
1029 AVStream *st = s1->streams[0];
1032 int size= pkt->size;
1033 uint8_t *buf1= pkt->data;
1036 printf("%d: write len=%d\n", pkt->stream_index, size);
1039 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
1040 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
1042 if (s->first_packet || rtcp_bytes >= 28) {
1043 /* compute NTP time */
1044 /* XXX: 90 kHz timestamp hardcoded */
1045 ntp_time = (pkt->pts << 28) / 5625;
1046 rtcp_send_sr(s1, ntp_time);
1047 s->last_octet_count = s->octet_count;
1048 s->first_packet = 0;
1051 switch(st->codec->codec_id) {
1052 case CODEC_ID_PCM_MULAW:
1053 case CODEC_ID_PCM_ALAW:
1054 case CODEC_ID_PCM_U8:
1055 case CODEC_ID_PCM_S8:
1056 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
1058 case CODEC_ID_PCM_U16BE:
1059 case CODEC_ID_PCM_U16LE:
1060 case CODEC_ID_PCM_S16BE:
1061 case CODEC_ID_PCM_S16LE:
1062 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
1066 rtp_send_mpegaudio(s1, buf1, size);
1068 case CODEC_ID_MPEG1VIDEO:
1069 rtp_send_mpegvideo(s1, buf1, size);
1071 case CODEC_ID_MPEG2TS:
1072 rtp_send_mpegts_raw(s1, buf1, size);
1075 /* better than nothing : send the codec raw data */
1076 rtp_send_raw(s1, buf1, size);
1082 static int rtp_write_trailer(AVFormatContext *s1)
1084 // RTPDemuxContext *s = s1->priv_data;
1088 AVOutputFormat rtp_muxer = {
1090 "RTP output format",
1093 sizeof(RTPDemuxContext),