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"
28 #include "rtp_internal.h"
34 /* TODO: - add RTCP statistics reporting (should be optional).
36 - add support for h263/mpeg4 packetized output : IDEA: send a
37 buffer to 'rtp_write_packet' contains all the packets for ONE
38 frame. Each packet should have a four byte header containing
39 the length in big endian format (same trick as
40 'url_open_dyn_packet_buf')
43 /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
44 AVRtpPayloadType_t AVRtpPayloadTypes[]=
46 {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1},
47 {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
48 {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
49 {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
50 {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
51 {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
52 {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1},
53 {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
54 {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1},
55 {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
56 {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2},
57 {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1},
58 {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1},
59 {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
60 {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1},
61 {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
62 {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1},
63 {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1},
64 {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
65 {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
66 {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
67 {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
68 {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
69 {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1},
70 {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
71 {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
72 {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1},
73 {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
74 {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1},
75 {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
76 {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1},
77 {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1},
78 {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1},
79 {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1},
80 {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1},
81 {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
82 {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
83 {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
84 {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
85 {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
86 {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
87 {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
88 {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
89 {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
90 {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
91 {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
92 {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
93 {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
94 {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
95 {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
96 {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
97 {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
98 {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
99 {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
100 {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
101 {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
102 {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
103 {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
104 {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
105 {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
106 {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
107 {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
108 {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
109 {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
110 {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
111 {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
112 {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
113 {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
114 {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
115 {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
116 {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
117 {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
118 {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
119 {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
120 {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
121 {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
122 {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
123 {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
124 {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
125 {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
126 {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
127 {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
128 {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
129 {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
130 {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
131 {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
132 {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
133 {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
134 {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
135 {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
136 {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
137 {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
138 {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
139 {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
140 {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
141 {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
142 {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
143 {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
144 {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
145 {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
146 {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
147 {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
148 {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
149 {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
150 {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
151 {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
152 {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
153 {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
154 {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
155 {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
156 {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
157 {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
158 {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
159 {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
160 {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
161 {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
162 {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
163 {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
164 {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
165 {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
166 {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
167 {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
168 {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
169 {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
170 {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
171 {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
172 {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
173 {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1},
174 {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}
177 /* statistics functions */
178 RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
180 static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
181 static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
183 static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
185 handler->next= RTPFirstDynamicPayloadHandler;
186 RTPFirstDynamicPayloadHandler= handler;
189 void av_register_rtp_dynamic_payload_handlers()
191 register_dynamic_payload_handler(&mp4v_es_handler);
192 register_dynamic_payload_handler(&mpeg4_generic_handler);
193 register_dynamic_payload_handler(&ff_h264_dynamic_handler);
196 int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
198 if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
199 codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
200 codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
201 if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
202 codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
203 if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
204 codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate;
210 /* return < 0 if unknown payload type */
211 int rtp_get_payload_type(AVCodecContext *codec)
215 /* compute the payload type */
216 for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i)
217 if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) {
218 if (codec->codec_id == CODEC_ID_PCM_S16BE)
219 if (codec->channels != AVRtpPayloadTypes[i].audio_channels)
221 payload_type = AVRtpPayloadTypes[i].pt;
226 static inline uint32_t decode_be32(const uint8_t *p)
228 return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3];
231 static inline uint64_t decode_be64(const uint8_t *p)
233 return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4);
236 static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
240 s->last_rtcp_ntp_time = decode_be64(buf + 8);
241 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
242 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
243 s->last_rtcp_timestamp = decode_be32(buf + 16);
247 #define RTP_SEQ_MOD (1<<16)
250 * called on parse open packet
252 static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
254 memset(s, 0, sizeof(RTPStatistics));
255 s->max_seq= base_sequence;
260 * called whenever there is a large jump in sequence numbers, or when they get out of probation...
262 static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
267 s->bad_seq= RTP_SEQ_MOD + 1;
269 s->expected_prior= 0;
270 s->received_prior= 0;
276 * returns 1 if we should handle this packet.
278 static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
280 uint16_t udelta= seq - s->max_seq;
281 const int MAX_DROPOUT= 3000;
282 const int MAX_MISORDER = 100;
283 const int MIN_SEQUENTIAL = 2;
285 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
288 if(seq==s->max_seq + 1) {
291 if(s->probation==0) {
292 rtp_init_sequence(s, seq);
297 s->probation= MIN_SEQUENTIAL - 1;
300 } else if (udelta < MAX_DROPOUT) {
301 // in order, with permissible gap
302 if(seq < s->max_seq) {
303 //sequence number wrapped; count antother 64k cycles
304 s->cycles += RTP_SEQ_MOD;
307 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
308 // sequence made a large jump...
309 if(seq==s->bad_seq) {
310 // two sequential packets-- assume that the other side restarted without telling us; just resync.
311 rtp_init_sequence(s, seq);
313 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
317 // duplicate or reordered packet...
325 * This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
326 * difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
327 * never change. I left this in in case someone else can see a way. (rdm)
329 static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
331 uint32_t transit= arrival_timestamp - sent_timestamp;
334 d= FFABS(transit - s->transit);
335 s->jitter += d - ((s->jitter + 8)>>4);
340 * some rtp servers assume client is dead if they don't hear from them...
341 * so we send a Receiver Report to the provided ByteIO context
342 * (we don't have access to the rtcp handle from here)
344 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
350 RTPStatistics *stats= &s->statistics;
352 uint32_t extended_max;
353 uint32_t expected_interval;
354 uint32_t received_interval;
355 uint32_t lost_interval;
358 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
360 if (!s->rtp_ctx || (count < 1))
363 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
364 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
365 s->octet_count += count;
366 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
368 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
371 s->last_octet_count = s->octet_count;
373 if (url_open_dyn_buf(&pb) < 0)
377 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
379 put_be16(&pb, 7); /* length in words - 1 */
380 put_be32(&pb, s->ssrc); // our own SSRC
381 put_be32(&pb, s->ssrc); // XXX: should be the server's here!
382 // some placeholders we should really fill...
384 extended_max= stats->cycles + stats->max_seq;
385 expected= extended_max - stats->base_seq + 1;
386 lost= expected - stats->received;
387 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
388 expected_interval= expected - stats->expected_prior;
389 stats->expected_prior= expected;
390 received_interval= stats->received - stats->received_prior;
391 stats->received_prior= stats->received;
392 lost_interval= expected_interval - received_interval;
393 if (expected_interval==0 || lost_interval<=0) fraction= 0;
394 else fraction = (lost_interval<<8)/expected_interval;
396 fraction= (fraction<<24) | lost;
398 put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
399 put_be32(&pb, extended_max); /* max sequence received */
400 put_be32(&pb, stats->jitter>>4); /* jitter */
402 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
404 put_be32(&pb, 0); /* last SR timestamp */
405 put_be32(&pb, 0); /* delay since last SR */
407 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
408 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
410 put_be32(&pb, middle_32_bits); /* last SR timestamp */
411 put_be32(&pb, delay_since_last); /* delay since last SR */
415 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
417 len = strlen(s->hostname);
418 put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */
419 put_be32(&pb, s->ssrc);
422 put_buffer(&pb, s->hostname, len);
424 for (len = (6 + len) % 4; len % 4; len++) {
428 put_flush_packet(&pb);
429 len = url_close_dyn_buf(&pb, &buf);
430 if ((len > 0) && buf) {
433 printf("sending %d bytes of RR\n", len);
435 result= url_write(s->rtp_ctx, buf, len);
437 printf("result from url_write: %d\n", result);
445 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
446 * MPEG2TS streams to indicate that they should be demuxed inside the
447 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
448 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
450 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
454 s = av_mallocz(sizeof(RTPDemuxContext));
457 s->payload_type = payload_type;
458 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
459 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
462 s->rtp_payload_data = rtp_payload_data;
463 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
464 if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
465 s->ts = mpegts_parse_open(s->ic);
471 switch(st->codec->codec_id) {
472 case CODEC_ID_MPEG1VIDEO:
473 case CODEC_ID_MPEG2VIDEO:
478 st->need_parsing = 1;
484 // needed to send back RTCP RR in RTSP sessions
486 gethostname(s->hostname, sizeof(s->hostname));
490 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
492 int au_headers_length, au_header_size, i;
493 GetBitContext getbitcontext;
494 rtp_payload_data_t *infos;
496 infos = s->rtp_payload_data;
501 /* decode the first 2 bytes where are stored the AUHeader sections
503 au_headers_length = AV_RB16(buf);
505 if (au_headers_length > RTP_MAX_PACKET_LENGTH)
508 infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
510 /* skip AU headers length section (2 bytes) */
513 init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
515 /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
516 au_header_size = infos->sizelength + infos->indexlength;
517 if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
520 infos->nb_au_headers = au_headers_length / au_header_size;
521 infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
523 /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
524 In my test, the faad decoder doesnt behave correctly when sending each AU one by one
525 but does when sending the whole as one big packet... */
526 infos->au_headers[0].size = 0;
527 infos->au_headers[0].index = 0;
528 for (i = 0; i < infos->nb_au_headers; ++i) {
529 infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
530 infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
533 infos->nb_au_headers = 1;
539 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
541 static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
543 switch(s->st->codec->codec_id) {
545 case CODEC_ID_MPEG1VIDEO:
546 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
550 /* XXX: is it really necessary to unify the timestamp base ? */
551 /* compute pts from timestamp with received ntp_time */
552 delta_timestamp = timestamp - s->last_rtcp_timestamp;
553 /* convert to 90 kHz without overflow */
554 addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
555 addend = (addend * 5625) >> 14;
556 pkt->pts = addend + delta_timestamp;
562 pkt->pts = timestamp;
565 /* no timestamp info yet */
568 pkt->stream_index = s->st->index;
572 * Parse an RTP or RTCP packet directly sent as a buffer.
573 * @param s RTP parse context.
574 * @param pkt returned packet
575 * @param buf input buffer or NULL to read the next packets
576 * @param len buffer len
577 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
578 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
580 int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
581 const uint8_t *buf, int len)
583 unsigned int ssrc, h;
584 int payload_type, seq, ret;
590 /* return the next packets, if any */
591 if(s->st && s->parse_packet) {
592 timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
593 rv= s->parse_packet(s, pkt, ×tamp, NULL, 0);
594 finalize_packet(s, pkt, timestamp);
597 // TODO: Move to a dynamic packet handler (like above)
598 if (s->read_buf_index >= s->read_buf_size)
600 ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
601 s->read_buf_size - s->read_buf_index);
604 s->read_buf_index += ret;
605 if (s->read_buf_index < s->read_buf_size)
615 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
617 if (buf[1] >= 200 && buf[1] <= 204) {
618 rtcp_parse_packet(s, buf, len);
621 payload_type = buf[1] & 0x7f;
622 seq = (buf[2] << 8) | buf[3];
623 timestamp = decode_be32(buf + 4);
624 ssrc = decode_be32(buf + 8);
625 /* store the ssrc in the RTPDemuxContext */
628 /* NOTE: we can handle only one payload type */
629 if (s->payload_type != payload_type)
633 // only do something with this if all the rtp checks pass...
634 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
636 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
637 payload_type, seq, ((s->seq + 1) & 0xffff));
646 /* specific MPEG2TS demux support */
647 ret = mpegts_parse_packet(s->ts, pkt, buf, len);
651 s->read_buf_size = len - ret;
652 memcpy(s->buf, buf + ret, s->read_buf_size);
653 s->read_buf_index = 0;
657 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
658 switch(st->codec->codec_id) {
660 /* better than nothing: skip mpeg audio RTP header */
663 h = decode_be32(buf);
666 av_new_packet(pkt, len);
667 memcpy(pkt->data, buf, len);
669 case CODEC_ID_MPEG1VIDEO:
670 /* better than nothing: skip mpeg video RTP header */
673 h = decode_be32(buf);
683 av_new_packet(pkt, len);
684 memcpy(pkt->data, buf, len);
686 // moved from below, verbatim. this is because this section handles packets, and the lower switch handles
688 // TODO: Put this into a dynamic packet handler...
690 if (rtp_parse_mp4_au(s, buf))
693 rtp_payload_data_t *infos = s->rtp_payload_data;
696 buf += infos->au_headers_length_bytes + 2;
697 len -= infos->au_headers_length_bytes + 2;
699 /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
701 av_new_packet(pkt, infos->au_headers[0].size);
702 memcpy(pkt->data, buf, infos->au_headers[0].size);
703 buf += infos->au_headers[0].size;
704 len -= infos->au_headers[0].size;
706 s->read_buf_size = len;
711 if(s->parse_packet) {
712 rv= s->parse_packet(s, pkt, ×tamp, buf, len);
714 av_new_packet(pkt, len);
715 memcpy(pkt->data, buf, len);
720 // now perform timestamp things....
721 finalize_packet(s, pkt, timestamp);
726 void rtp_parse_close(RTPDemuxContext *s)
728 // TODO: fold this into the protocol specific data fields.
729 if (!strcmp(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) {
730 mpegts_parse_close(s->ts);
737 static int rtp_write_header(AVFormatContext *s1)
739 RTPDemuxContext *s = s1->priv_data;
740 int payload_type, max_packet_size, n;
743 if (s1->nb_streams != 1)
747 payload_type = rtp_get_payload_type(st->codec);
748 if (payload_type < 0)
749 payload_type = RTP_PT_PRIVATE; /* private payload type */
750 s->payload_type = payload_type;
752 // following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
753 s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
754 s->timestamp = s->base_timestamp;
755 s->ssrc = 0; /* FIXME: was random(), what should this be? */
758 max_packet_size = url_fget_max_packet_size(&s1->pb);
759 if (max_packet_size <= 12)
761 s->max_payload_size = max_packet_size - 12;
763 switch(st->codec->codec_id) {
766 s->buf_ptr = s->buf + 4;
767 s->cur_timestamp = 0;
769 case CODEC_ID_MPEG1VIDEO:
770 s->cur_timestamp = 0;
772 case CODEC_ID_MPEG2TS:
773 n = s->max_payload_size / TS_PACKET_SIZE;
776 s->max_payload_size = n * TS_PACKET_SIZE;
787 /* send an rtcp sender report packet */
788 static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
790 RTPDemuxContext *s = s1->priv_data;
792 printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp);
794 put_byte(&s1->pb, (RTP_VERSION << 6));
795 put_byte(&s1->pb, 200);
796 put_be16(&s1->pb, 6); /* length in words - 1 */
797 put_be32(&s1->pb, s->ssrc);
798 put_be64(&s1->pb, ntp_time);
799 put_be32(&s1->pb, s->timestamp);
800 put_be32(&s1->pb, s->packet_count);
801 put_be32(&s1->pb, s->octet_count);
802 put_flush_packet(&s1->pb);
805 /* send an rtp packet. sequence number is incremented, but the caller
806 must update the timestamp itself */
807 static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
809 RTPDemuxContext *s = s1->priv_data;
812 printf("rtp_send_data size=%d\n", len);
815 /* build the RTP header */
816 put_byte(&s1->pb, (RTP_VERSION << 6));
817 put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
818 put_be16(&s1->pb, s->seq);
819 put_be32(&s1->pb, s->timestamp);
820 put_be32(&s1->pb, s->ssrc);
822 put_buffer(&s1->pb, buf1, len);
823 put_flush_packet(&s1->pb);
826 s->octet_count += len;
830 /* send an integer number of samples and compute time stamp and fill
831 the rtp send buffer before sending. */
832 static void rtp_send_samples(AVFormatContext *s1,
833 const uint8_t *buf1, int size, int sample_size)
835 RTPDemuxContext *s = s1->priv_data;
836 int len, max_packet_size, n;
838 max_packet_size = (s->max_payload_size / sample_size) * sample_size;
839 /* not needed, but who nows */
840 if ((size % sample_size) != 0)
843 len = (max_packet_size - (s->buf_ptr - s->buf));
848 memcpy(s->buf_ptr, buf1, len);
852 n = (s->buf_ptr - s->buf);
853 /* if buffer full, then send it */
854 if (n >= max_packet_size) {
855 rtp_send_data(s1, s->buf, n, 0);
857 /* update timestamp */
858 s->timestamp += n / sample_size;
863 /* NOTE: we suppose that exactly one frame is given as argument here */
865 static void rtp_send_mpegaudio(AVFormatContext *s1,
866 const uint8_t *buf1, int size)
868 RTPDemuxContext *s = s1->priv_data;
869 AVStream *st = s1->streams[0];
870 int len, count, max_packet_size;
872 max_packet_size = s->max_payload_size;
874 /* test if we must flush because not enough space */
875 len = (s->buf_ptr - s->buf);
876 if ((len + size) > max_packet_size) {
878 rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
879 s->buf_ptr = s->buf + 4;
880 /* 90 KHz time stamp */
881 s->timestamp = s->base_timestamp +
882 (s->cur_timestamp * 90000LL) / st->codec->sample_rate;
887 if (size > max_packet_size) {
888 /* big packet: fragment */
891 len = max_packet_size - 4;
894 /* build fragmented packet */
897 s->buf[2] = count >> 8;
899 memcpy(s->buf + 4, buf1, len);
900 rtp_send_data(s1, s->buf, len + 4, 0);
906 if (s->buf_ptr == s->buf + 4) {
907 /* no fragmentation possible */
913 memcpy(s->buf_ptr, buf1, size);
916 s->cur_timestamp += st->codec->frame_size;
919 /* NOTE: a single frame must be passed with sequence header if
920 needed. XXX: use slices. */
921 static void rtp_send_mpegvideo(AVFormatContext *s1,
922 const uint8_t *buf1, int size)
924 RTPDemuxContext *s = s1->priv_data;
925 AVStream *st = s1->streams[0];
926 int len, h, max_packet_size;
929 max_packet_size = s->max_payload_size;
932 /* XXX: more correct headers */
934 if (st->codec->sub_id == 2)
935 h |= 1 << 26; /* mpeg 2 indicator */
942 if (st->codec->sub_id == 2) {
950 len = max_packet_size - (q - s->buf);
954 memcpy(q, buf1, len);
957 /* 90 KHz time stamp */
958 s->timestamp = s->base_timestamp +
959 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
960 rtp_send_data(s1, s->buf, q - s->buf, (len == size));
968 static void rtp_send_raw(AVFormatContext *s1,
969 const uint8_t *buf1, int size)
971 RTPDemuxContext *s = s1->priv_data;
972 AVStream *st = s1->streams[0];
973 int len, max_packet_size;
975 max_packet_size = s->max_payload_size;
978 len = max_packet_size;
982 /* 90 KHz time stamp */
983 s->timestamp = s->base_timestamp +
984 av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
985 rtp_send_data(s1, buf1, len, (len == size));
993 /* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */
994 static void rtp_send_mpegts_raw(AVFormatContext *s1,
995 const uint8_t *buf1, int size)
997 RTPDemuxContext *s = s1->priv_data;
1000 while (size >= TS_PACKET_SIZE) {
1001 len = s->max_payload_size - (s->buf_ptr - s->buf);
1004 memcpy(s->buf_ptr, buf1, len);
1009 out_len = s->buf_ptr - s->buf;
1010 if (out_len >= s->max_payload_size) {
1011 rtp_send_data(s1, s->buf, out_len, 0);
1012 s->buf_ptr = s->buf;
1017 /* write an RTP packet. 'buf1' must contain a single specific frame. */
1018 static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
1020 RTPDemuxContext *s = s1->priv_data;
1021 AVStream *st = s1->streams[0];
1024 int size= pkt->size;
1025 uint8_t *buf1= pkt->data;
1028 printf("%d: write len=%d\n", pkt->stream_index, size);
1031 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
1032 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
1034 if (s->first_packet || rtcp_bytes >= 28) {
1035 /* compute NTP time */
1036 /* XXX: 90 kHz timestamp hardcoded */
1037 ntp_time = (pkt->pts << 28) / 5625;
1038 rtcp_send_sr(s1, ntp_time);
1039 s->last_octet_count = s->octet_count;
1040 s->first_packet = 0;
1043 switch(st->codec->codec_id) {
1044 case CODEC_ID_PCM_MULAW:
1045 case CODEC_ID_PCM_ALAW:
1046 case CODEC_ID_PCM_U8:
1047 case CODEC_ID_PCM_S8:
1048 rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
1050 case CODEC_ID_PCM_U16BE:
1051 case CODEC_ID_PCM_U16LE:
1052 case CODEC_ID_PCM_S16BE:
1053 case CODEC_ID_PCM_S16LE:
1054 rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
1058 rtp_send_mpegaudio(s1, buf1, size);
1060 case CODEC_ID_MPEG1VIDEO:
1061 rtp_send_mpegvideo(s1, buf1, size);
1063 case CODEC_ID_MPEG2TS:
1064 rtp_send_mpegts_raw(s1, buf1, size);
1067 /* better than nothing : send the codec raw data */
1068 rtp_send_raw(s1, buf1, size);
1074 static int rtp_write_trailer(AVFormatContext *s1)
1076 // RTPDemuxContext *s = s1->priv_data;
1080 AVOutputFormat rtp_muxer = {
1082 "RTP output format",
1085 sizeof(RTPDemuxContext),