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2ba45a60 DM |
1 | /* |
2 | * RTP input format | |
3 | * Copyright (c) 2002 Fabrice Bellard | |
4 | * | |
5 | * This file is part of FFmpeg. | |
6 | * | |
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. | |
11 | * | |
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. | |
16 | * | |
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 | |
20 | */ | |
21 | ||
22 | #include "libavutil/mathematics.h" | |
23 | #include "libavutil/avstring.h" | |
24 | #include "libavutil/time.h" | |
25 | #include "libavcodec/get_bits.h" | |
26 | #include "avformat.h" | |
27 | #include "network.h" | |
28 | #include "srtp.h" | |
29 | #include "url.h" | |
30 | #include "rtpdec.h" | |
31 | #include "rtpdec_formats.h" | |
32 | ||
33 | #define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */ | |
34 | ||
35 | static RTPDynamicProtocolHandler gsm_dynamic_handler = { | |
36 | .enc_name = "GSM", | |
37 | .codec_type = AVMEDIA_TYPE_AUDIO, | |
38 | .codec_id = AV_CODEC_ID_GSM, | |
39 | }; | |
40 | ||
41 | static RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = { | |
42 | .enc_name = "X-MP3-draft-00", | |
43 | .codec_type = AVMEDIA_TYPE_AUDIO, | |
44 | .codec_id = AV_CODEC_ID_MP3ADU, | |
45 | }; | |
46 | ||
47 | static RTPDynamicProtocolHandler speex_dynamic_handler = { | |
48 | .enc_name = "speex", | |
49 | .codec_type = AVMEDIA_TYPE_AUDIO, | |
50 | .codec_id = AV_CODEC_ID_SPEEX, | |
51 | }; | |
52 | ||
53 | static RTPDynamicProtocolHandler opus_dynamic_handler = { | |
54 | .enc_name = "opus", | |
55 | .codec_type = AVMEDIA_TYPE_AUDIO, | |
56 | .codec_id = AV_CODEC_ID_OPUS, | |
57 | }; | |
58 | ||
59 | static RTPDynamicProtocolHandler *rtp_first_dynamic_payload_handler = NULL; | |
60 | ||
61 | void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) | |
62 | { | |
63 | handler->next = rtp_first_dynamic_payload_handler; | |
64 | rtp_first_dynamic_payload_handler = handler; | |
65 | } | |
66 | ||
67 | void ff_register_rtp_dynamic_payload_handlers(void) | |
68 | { | |
69 | ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler); | |
70 | ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler); | |
71 | ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler); | |
72 | ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler); | |
73 | ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler); | |
74 | ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler); | |
75 | ff_register_dynamic_payload_handler(&ff_h261_dynamic_handler); | |
76 | ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler); | |
77 | ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler); | |
78 | ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler); | |
79 | ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler); | |
80 | ff_register_dynamic_payload_handler(&ff_h265_dynamic_handler); | |
81 | ff_register_dynamic_payload_handler(&ff_hevc_dynamic_handler); | |
82 | ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler); | |
83 | ff_register_dynamic_payload_handler(&ff_jpeg_dynamic_handler); | |
84 | ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler); | |
85 | ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler); | |
86 | ff_register_dynamic_payload_handler(&ff_mpeg_audio_dynamic_handler); | |
87 | ff_register_dynamic_payload_handler(&ff_mpeg_video_dynamic_handler); | |
88 | ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler); | |
89 | ff_register_dynamic_payload_handler(&ff_mpegts_dynamic_handler); | |
90 | ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler); | |
91 | ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler); | |
92 | ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler); | |
93 | ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler); | |
94 | ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler); | |
95 | ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler); | |
96 | ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler); | |
97 | ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler); | |
98 | ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler); | |
99 | ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler); | |
100 | ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler); | |
101 | ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler); | |
102 | ff_register_dynamic_payload_handler(&gsm_dynamic_handler); | |
103 | ff_register_dynamic_payload_handler(&opus_dynamic_handler); | |
104 | ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler); | |
105 | ff_register_dynamic_payload_handler(&speex_dynamic_handler); | |
106 | } | |
107 | ||
108 | RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name, | |
109 | enum AVMediaType codec_type) | |
110 | { | |
111 | RTPDynamicProtocolHandler *handler; | |
112 | for (handler = rtp_first_dynamic_payload_handler; | |
113 | handler; handler = handler->next) | |
114 | if (!av_strcasecmp(name, handler->enc_name) && | |
115 | codec_type == handler->codec_type) | |
116 | return handler; | |
117 | return NULL; | |
118 | } | |
119 | ||
120 | RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id, | |
121 | enum AVMediaType codec_type) | |
122 | { | |
123 | RTPDynamicProtocolHandler *handler; | |
124 | for (handler = rtp_first_dynamic_payload_handler; | |
125 | handler; handler = handler->next) | |
126 | if (handler->static_payload_id && handler->static_payload_id == id && | |
127 | codec_type == handler->codec_type) | |
128 | return handler; | |
129 | return NULL; | |
130 | } | |
131 | ||
132 | static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, | |
133 | int len) | |
134 | { | |
135 | int payload_len; | |
136 | while (len >= 4) { | |
137 | payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4); | |
138 | ||
139 | switch (buf[1]) { | |
140 | case RTCP_SR: | |
141 | if (payload_len < 20) { | |
142 | av_log(NULL, AV_LOG_ERROR, | |
143 | "Invalid length for RTCP SR packet\n"); | |
144 | return AVERROR_INVALIDDATA; | |
145 | } | |
146 | ||
147 | s->last_rtcp_reception_time = av_gettime(); | |
148 | s->last_rtcp_ntp_time = AV_RB64(buf + 8); | |
149 | s->last_rtcp_timestamp = AV_RB32(buf + 16); | |
150 | if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) { | |
151 | s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; | |
152 | if (!s->base_timestamp) | |
153 | s->base_timestamp = s->last_rtcp_timestamp; | |
154 | s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp; | |
155 | } | |
156 | ||
157 | break; | |
158 | case RTCP_BYE: | |
159 | return -RTCP_BYE; | |
160 | } | |
161 | ||
162 | buf += payload_len; | |
163 | len -= payload_len; | |
164 | } | |
165 | return -1; | |
166 | } | |
167 | ||
168 | #define RTP_SEQ_MOD (1 << 16) | |
169 | ||
170 | static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) | |
171 | { | |
172 | memset(s, 0, sizeof(RTPStatistics)); | |
173 | s->max_seq = base_sequence; | |
174 | s->probation = 1; | |
175 | } | |
176 | ||
177 | /* | |
178 | * Called whenever there is a large jump in sequence numbers, | |
179 | * or when they get out of probation... | |
180 | */ | |
181 | static void rtp_init_sequence(RTPStatistics *s, uint16_t seq) | |
182 | { | |
183 | s->max_seq = seq; | |
184 | s->cycles = 0; | |
185 | s->base_seq = seq - 1; | |
186 | s->bad_seq = RTP_SEQ_MOD + 1; | |
187 | s->received = 0; | |
188 | s->expected_prior = 0; | |
189 | s->received_prior = 0; | |
190 | s->jitter = 0; | |
191 | s->transit = 0; | |
192 | } | |
193 | ||
194 | /* Returns 1 if we should handle this packet. */ | |
195 | static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq) | |
196 | { | |
197 | uint16_t udelta = seq - s->max_seq; | |
198 | const int MAX_DROPOUT = 3000; | |
199 | const int MAX_MISORDER = 100; | |
200 | const int MIN_SEQUENTIAL = 2; | |
201 | ||
202 | /* source not valid until MIN_SEQUENTIAL packets with sequence | |
203 | * seq. numbers have been received */ | |
204 | if (s->probation) { | |
205 | if (seq == s->max_seq + 1) { | |
206 | s->probation--; | |
207 | s->max_seq = seq; | |
208 | if (s->probation == 0) { | |
209 | rtp_init_sequence(s, seq); | |
210 | s->received++; | |
211 | return 1; | |
212 | } | |
213 | } else { | |
214 | s->probation = MIN_SEQUENTIAL - 1; | |
215 | s->max_seq = seq; | |
216 | } | |
217 | } else if (udelta < MAX_DROPOUT) { | |
218 | // in order, with permissible gap | |
219 | if (seq < s->max_seq) { | |
220 | // sequence number wrapped; count another 64k cycles | |
221 | s->cycles += RTP_SEQ_MOD; | |
222 | } | |
223 | s->max_seq = seq; | |
224 | } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) { | |
225 | // sequence made a large jump... | |
226 | if (seq == s->bad_seq) { | |
227 | /* two sequential packets -- assume that the other side | |
228 | * restarted without telling us; just resync. */ | |
229 | rtp_init_sequence(s, seq); | |
230 | } else { | |
231 | s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1); | |
232 | return 0; | |
233 | } | |
234 | } else { | |
235 | // duplicate or reordered packet... | |
236 | } | |
237 | s->received++; | |
238 | return 1; | |
239 | } | |
240 | ||
241 | static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, | |
242 | uint32_t arrival_timestamp) | |
243 | { | |
244 | // Most of this is pretty straight from RFC 3550 appendix A.8 | |
245 | uint32_t transit = arrival_timestamp - sent_timestamp; | |
246 | uint32_t prev_transit = s->transit; | |
247 | int32_t d = transit - prev_transit; | |
248 | // Doing the FFABS() call directly on the "transit - prev_transit" | |
249 | // expression doesn't work, since it's an unsigned expression. Doing the | |
250 | // transit calculation in unsigned is desired though, since it most | |
251 | // probably will need to wrap around. | |
252 | d = FFABS(d); | |
253 | s->transit = transit; | |
254 | if (!prev_transit) | |
255 | return; | |
256 | s->jitter += d - (int32_t) ((s->jitter + 8) >> 4); | |
257 | } | |
258 | ||
259 | int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd, | |
260 | AVIOContext *avio, int count) | |
261 | { | |
262 | AVIOContext *pb; | |
263 | uint8_t *buf; | |
264 | int len; | |
265 | int rtcp_bytes; | |
266 | RTPStatistics *stats = &s->statistics; | |
267 | uint32_t lost; | |
268 | uint32_t extended_max; | |
269 | uint32_t expected_interval; | |
270 | uint32_t received_interval; | |
271 | int32_t lost_interval; | |
272 | uint32_t expected; | |
273 | uint32_t fraction; | |
274 | ||
275 | if ((!fd && !avio) || (count < 1)) | |
276 | return -1; | |
277 | ||
278 | /* TODO: I think this is way too often; RFC 1889 has algorithm for this */ | |
279 | /* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */ | |
280 | s->octet_count += count; | |
281 | rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / | |
282 | RTCP_TX_RATIO_DEN; | |
283 | rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? | |
284 | if (rtcp_bytes < 28) | |
285 | return -1; | |
286 | s->last_octet_count = s->octet_count; | |
287 | ||
288 | if (!fd) | |
289 | pb = avio; | |
290 | else if (avio_open_dyn_buf(&pb) < 0) | |
291 | return -1; | |
292 | ||
293 | // Receiver Report | |
294 | avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ | |
295 | avio_w8(pb, RTCP_RR); | |
296 | avio_wb16(pb, 7); /* length in words - 1 */ | |
297 | // our own SSRC: we use the server's SSRC + 1 to avoid conflicts | |
298 | avio_wb32(pb, s->ssrc + 1); | |
299 | avio_wb32(pb, s->ssrc); // server SSRC | |
300 | // some placeholders we should really fill... | |
301 | // RFC 1889/p64 | |
302 | extended_max = stats->cycles + stats->max_seq; | |
303 | expected = extended_max - stats->base_seq; | |
304 | lost = expected - stats->received; | |
305 | lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits... | |
306 | expected_interval = expected - stats->expected_prior; | |
307 | stats->expected_prior = expected; | |
308 | received_interval = stats->received - stats->received_prior; | |
309 | stats->received_prior = stats->received; | |
310 | lost_interval = expected_interval - received_interval; | |
311 | if (expected_interval == 0 || lost_interval <= 0) | |
312 | fraction = 0; | |
313 | else | |
314 | fraction = (lost_interval << 8) / expected_interval; | |
315 | ||
316 | fraction = (fraction << 24) | lost; | |
317 | ||
318 | avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */ | |
319 | avio_wb32(pb, extended_max); /* max sequence received */ | |
320 | avio_wb32(pb, stats->jitter >> 4); /* jitter */ | |
321 | ||
322 | if (s->last_rtcp_ntp_time == AV_NOPTS_VALUE) { | |
323 | avio_wb32(pb, 0); /* last SR timestamp */ | |
324 | avio_wb32(pb, 0); /* delay since last SR */ | |
325 | } else { | |
326 | uint32_t middle_32_bits = s->last_rtcp_ntp_time >> 16; // this is valid, right? do we need to handle 64 bit values special? | |
327 | uint32_t delay_since_last = av_rescale(av_gettime() - s->last_rtcp_reception_time, | |
328 | 65536, AV_TIME_BASE); | |
329 | ||
330 | avio_wb32(pb, middle_32_bits); /* last SR timestamp */ | |
331 | avio_wb32(pb, delay_since_last); /* delay since last SR */ | |
332 | } | |
333 | ||
334 | // CNAME | |
335 | avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ | |
336 | avio_w8(pb, RTCP_SDES); | |
337 | len = strlen(s->hostname); | |
338 | avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */ | |
339 | avio_wb32(pb, s->ssrc + 1); | |
340 | avio_w8(pb, 0x01); | |
341 | avio_w8(pb, len); | |
342 | avio_write(pb, s->hostname, len); | |
343 | avio_w8(pb, 0); /* END */ | |
344 | // padding | |
345 | for (len = (7 + len) % 4; len % 4; len++) | |
346 | avio_w8(pb, 0); | |
347 | ||
348 | avio_flush(pb); | |
349 | if (!fd) | |
350 | return 0; | |
351 | len = avio_close_dyn_buf(pb, &buf); | |
352 | if ((len > 0) && buf) { | |
353 | int av_unused result; | |
354 | av_dlog(s->ic, "sending %d bytes of RR\n", len); | |
355 | result = ffurl_write(fd, buf, len); | |
356 | av_dlog(s->ic, "result from ffurl_write: %d\n", result); | |
357 | av_free(buf); | |
358 | } | |
359 | return 0; | |
360 | } | |
361 | ||
362 | void ff_rtp_send_punch_packets(URLContext *rtp_handle) | |
363 | { | |
364 | AVIOContext *pb; | |
365 | uint8_t *buf; | |
366 | int len; | |
367 | ||
368 | /* Send a small RTP packet */ | |
369 | if (avio_open_dyn_buf(&pb) < 0) | |
370 | return; | |
371 | ||
372 | avio_w8(pb, (RTP_VERSION << 6)); | |
373 | avio_w8(pb, 0); /* Payload type */ | |
374 | avio_wb16(pb, 0); /* Seq */ | |
375 | avio_wb32(pb, 0); /* Timestamp */ | |
376 | avio_wb32(pb, 0); /* SSRC */ | |
377 | ||
378 | avio_flush(pb); | |
379 | len = avio_close_dyn_buf(pb, &buf); | |
380 | if ((len > 0) && buf) | |
381 | ffurl_write(rtp_handle, buf, len); | |
382 | av_free(buf); | |
383 | ||
384 | /* Send a minimal RTCP RR */ | |
385 | if (avio_open_dyn_buf(&pb) < 0) | |
386 | return; | |
387 | ||
388 | avio_w8(pb, (RTP_VERSION << 6)); | |
389 | avio_w8(pb, RTCP_RR); /* receiver report */ | |
390 | avio_wb16(pb, 1); /* length in words - 1 */ | |
391 | avio_wb32(pb, 0); /* our own SSRC */ | |
392 | ||
393 | avio_flush(pb); | |
394 | len = avio_close_dyn_buf(pb, &buf); | |
395 | if ((len > 0) && buf) | |
396 | ffurl_write(rtp_handle, buf, len); | |
397 | av_free(buf); | |
398 | } | |
399 | ||
400 | static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing, | |
401 | uint16_t *missing_mask) | |
402 | { | |
403 | int i; | |
404 | uint16_t next_seq = s->seq + 1; | |
405 | RTPPacket *pkt = s->queue; | |
406 | ||
407 | if (!pkt || pkt->seq == next_seq) | |
408 | return 0; | |
409 | ||
410 | *missing_mask = 0; | |
411 | for (i = 1; i <= 16; i++) { | |
412 | uint16_t missing_seq = next_seq + i; | |
413 | while (pkt) { | |
414 | int16_t diff = pkt->seq - missing_seq; | |
415 | if (diff >= 0) | |
416 | break; | |
417 | pkt = pkt->next; | |
418 | } | |
419 | if (!pkt) | |
420 | break; | |
421 | if (pkt->seq == missing_seq) | |
422 | continue; | |
423 | *missing_mask |= 1 << (i - 1); | |
424 | } | |
425 | ||
426 | *first_missing = next_seq; | |
427 | return 1; | |
428 | } | |
429 | ||
430 | int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd, | |
431 | AVIOContext *avio) | |
432 | { | |
433 | int len, need_keyframe, missing_packets; | |
434 | AVIOContext *pb; | |
435 | uint8_t *buf; | |
436 | int64_t now; | |
437 | uint16_t first_missing = 0, missing_mask = 0; | |
438 | ||
439 | if (!fd && !avio) | |
440 | return -1; | |
441 | ||
442 | need_keyframe = s->handler && s->handler->need_keyframe && | |
443 | s->handler->need_keyframe(s->dynamic_protocol_context); | |
444 | missing_packets = find_missing_packets(s, &first_missing, &missing_mask); | |
445 | ||
446 | if (!need_keyframe && !missing_packets) | |
447 | return 0; | |
448 | ||
449 | /* Send new feedback if enough time has elapsed since the last | |
450 | * feedback packet. */ | |
451 | ||
452 | now = av_gettime(); | |
453 | if (s->last_feedback_time && | |
454 | (now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL) | |
455 | return 0; | |
456 | s->last_feedback_time = now; | |
457 | ||
458 | if (!fd) | |
459 | pb = avio; | |
460 | else if (avio_open_dyn_buf(&pb) < 0) | |
461 | return -1; | |
462 | ||
463 | if (need_keyframe) { | |
464 | avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */ | |
465 | avio_w8(pb, RTCP_PSFB); | |
466 | avio_wb16(pb, 2); /* length in words - 1 */ | |
467 | // our own SSRC: we use the server's SSRC + 1 to avoid conflicts | |
468 | avio_wb32(pb, s->ssrc + 1); | |
469 | avio_wb32(pb, s->ssrc); // server SSRC | |
470 | } | |
471 | ||
472 | if (missing_packets) { | |
473 | avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */ | |
474 | avio_w8(pb, RTCP_RTPFB); | |
475 | avio_wb16(pb, 3); /* length in words - 1 */ | |
476 | avio_wb32(pb, s->ssrc + 1); | |
477 | avio_wb32(pb, s->ssrc); // server SSRC | |
478 | ||
479 | avio_wb16(pb, first_missing); | |
480 | avio_wb16(pb, missing_mask); | |
481 | } | |
482 | ||
483 | avio_flush(pb); | |
484 | if (!fd) | |
485 | return 0; | |
486 | len = avio_close_dyn_buf(pb, &buf); | |
487 | if (len > 0 && buf) { | |
488 | ffurl_write(fd, buf, len); | |
489 | av_free(buf); | |
490 | } | |
491 | return 0; | |
492 | } | |
493 | ||
494 | /** | |
495 | * open a new RTP parse context for stream 'st'. 'st' can be NULL for | |
496 | * MPEG2-TS streams. | |
497 | */ | |
498 | RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, | |
499 | int payload_type, int queue_size) | |
500 | { | |
501 | RTPDemuxContext *s; | |
502 | ||
503 | s = av_mallocz(sizeof(RTPDemuxContext)); | |
504 | if (!s) | |
505 | return NULL; | |
506 | s->payload_type = payload_type; | |
507 | s->last_rtcp_ntp_time = AV_NOPTS_VALUE; | |
508 | s->first_rtcp_ntp_time = AV_NOPTS_VALUE; | |
509 | s->ic = s1; | |
510 | s->st = st; | |
511 | s->queue_size = queue_size; | |
512 | rtp_init_statistics(&s->statistics, 0); | |
513 | if (st) { | |
514 | switch (st->codec->codec_id) { | |
515 | case AV_CODEC_ID_ADPCM_G722: | |
516 | /* According to RFC 3551, the stream clock rate is 8000 | |
517 | * even if the sample rate is 16000. */ | |
518 | if (st->codec->sample_rate == 8000) | |
519 | st->codec->sample_rate = 16000; | |
520 | break; | |
521 | default: | |
522 | break; | |
523 | } | |
524 | } | |
525 | // needed to send back RTCP RR in RTSP sessions | |
526 | gethostname(s->hostname, sizeof(s->hostname)); | |
527 | return s; | |
528 | } | |
529 | ||
530 | void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, | |
531 | RTPDynamicProtocolHandler *handler) | |
532 | { | |
533 | s->dynamic_protocol_context = ctx; | |
534 | s->handler = handler; | |
535 | } | |
536 | ||
537 | void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite, | |
538 | const char *params) | |
539 | { | |
540 | if (!ff_srtp_set_crypto(&s->srtp, suite, params)) | |
541 | s->srtp_enabled = 1; | |
542 | } | |
543 | ||
544 | /** | |
545 | * This was the second switch in rtp_parse packet. | |
546 | * Normalizes time, if required, sets stream_index, etc. | |
547 | */ | |
548 | static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) | |
549 | { | |
550 | if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE) | |
551 | return; /* Timestamp already set by depacketizer */ | |
552 | if (timestamp == RTP_NOTS_VALUE) | |
553 | return; | |
554 | ||
555 | if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) { | |
556 | int64_t addend; | |
557 | int delta_timestamp; | |
558 | ||
559 | /* compute pts from timestamp with received ntp_time */ | |
560 | delta_timestamp = timestamp - s->last_rtcp_timestamp; | |
561 | /* convert to the PTS timebase */ | |
562 | addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, | |
563 | s->st->time_base.den, | |
564 | (uint64_t) s->st->time_base.num << 32); | |
565 | pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend + | |
566 | delta_timestamp; | |
567 | return; | |
568 | } | |
569 | ||
570 | if (!s->base_timestamp) | |
571 | s->base_timestamp = timestamp; | |
572 | /* assume that the difference is INT32_MIN < x < INT32_MAX, | |
573 | * but allow the first timestamp to exceed INT32_MAX */ | |
574 | if (!s->timestamp) | |
575 | s->unwrapped_timestamp += timestamp; | |
576 | else | |
577 | s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp); | |
578 | s->timestamp = timestamp; | |
579 | pkt->pts = s->unwrapped_timestamp + s->range_start_offset - | |
580 | s->base_timestamp; | |
581 | } | |
582 | ||
583 | static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, | |
584 | const uint8_t *buf, int len) | |
585 | { | |
586 | unsigned int ssrc; | |
587 | int payload_type, seq, flags = 0; | |
588 | int ext, csrc; | |
589 | AVStream *st; | |
590 | uint32_t timestamp; | |
591 | int rv = 0; | |
592 | ||
593 | csrc = buf[0] & 0x0f; | |
594 | ext = buf[0] & 0x10; | |
595 | payload_type = buf[1] & 0x7f; | |
596 | if (buf[1] & 0x80) | |
597 | flags |= RTP_FLAG_MARKER; | |
598 | seq = AV_RB16(buf + 2); | |
599 | timestamp = AV_RB32(buf + 4); | |
600 | ssrc = AV_RB32(buf + 8); | |
601 | /* store the ssrc in the RTPDemuxContext */ | |
602 | s->ssrc = ssrc; | |
603 | ||
604 | /* NOTE: we can handle only one payload type */ | |
605 | if (s->payload_type != payload_type) | |
606 | return -1; | |
607 | ||
608 | st = s->st; | |
609 | // only do something with this if all the rtp checks pass... | |
610 | if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) { | |
611 | av_log(st ? st->codec : NULL, AV_LOG_ERROR, | |
612 | "RTP: PT=%02x: bad cseq %04x expected=%04x\n", | |
613 | payload_type, seq, ((s->seq + 1) & 0xffff)); | |
614 | return -1; | |
615 | } | |
616 | ||
617 | if (buf[0] & 0x20) { | |
618 | int padding = buf[len - 1]; | |
619 | if (len >= 12 + padding) | |
620 | len -= padding; | |
621 | } | |
622 | ||
623 | s->seq = seq; | |
624 | len -= 12; | |
625 | buf += 12; | |
626 | ||
627 | len -= 4 * csrc; | |
628 | buf += 4 * csrc; | |
629 | if (len < 0) | |
630 | return AVERROR_INVALIDDATA; | |
631 | ||
632 | /* RFC 3550 Section 5.3.1 RTP Header Extension handling */ | |
633 | if (ext) { | |
634 | if (len < 4) | |
635 | return -1; | |
636 | /* calculate the header extension length (stored as number | |
637 | * of 32-bit words) */ | |
638 | ext = (AV_RB16(buf + 2) + 1) << 2; | |
639 | ||
640 | if (len < ext) | |
641 | return -1; | |
642 | // skip past RTP header extension | |
643 | len -= ext; | |
644 | buf += ext; | |
645 | } | |
646 | ||
647 | if (s->handler && s->handler->parse_packet) { | |
648 | rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context, | |
649 | s->st, pkt, ×tamp, buf, len, seq, | |
650 | flags); | |
651 | } else if (st) { | |
652 | if ((rv = av_new_packet(pkt, len)) < 0) | |
653 | return rv; | |
654 | memcpy(pkt->data, buf, len); | |
655 | pkt->stream_index = st->index; | |
656 | } else { | |
657 | return AVERROR(EINVAL); | |
658 | } | |
659 | ||
660 | // now perform timestamp things.... | |
661 | finalize_packet(s, pkt, timestamp); | |
662 | ||
663 | return rv; | |
664 | } | |
665 | ||
666 | void ff_rtp_reset_packet_queue(RTPDemuxContext *s) | |
667 | { | |
668 | while (s->queue) { | |
669 | RTPPacket *next = s->queue->next; | |
670 | av_free(s->queue->buf); | |
671 | av_free(s->queue); | |
672 | s->queue = next; | |
673 | } | |
674 | s->seq = 0; | |
675 | s->queue_len = 0; | |
676 | s->prev_ret = 0; | |
677 | } | |
678 | ||
679 | static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len) | |
680 | { | |
681 | uint16_t seq = AV_RB16(buf + 2); | |
682 | RTPPacket **cur = &s->queue, *packet; | |
683 | ||
684 | /* Find the correct place in the queue to insert the packet */ | |
685 | while (*cur) { | |
686 | int16_t diff = seq - (*cur)->seq; | |
687 | if (diff < 0) | |
688 | break; | |
689 | cur = &(*cur)->next; | |
690 | } | |
691 | ||
692 | packet = av_mallocz(sizeof(*packet)); | |
693 | if (!packet) | |
694 | return; | |
695 | packet->recvtime = av_gettime(); | |
696 | packet->seq = seq; | |
697 | packet->len = len; | |
698 | packet->buf = buf; | |
699 | packet->next = *cur; | |
700 | *cur = packet; | |
701 | s->queue_len++; | |
702 | } | |
703 | ||
704 | static int has_next_packet(RTPDemuxContext *s) | |
705 | { | |
706 | return s->queue && s->queue->seq == (uint16_t) (s->seq + 1); | |
707 | } | |
708 | ||
709 | int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s) | |
710 | { | |
711 | return s->queue ? s->queue->recvtime : 0; | |
712 | } | |
713 | ||
714 | static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt) | |
715 | { | |
716 | int rv; | |
717 | RTPPacket *next; | |
718 | ||
719 | if (s->queue_len <= 0) | |
720 | return -1; | |
721 | ||
722 | if (!has_next_packet(s)) | |
723 | av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING, | |
724 | "RTP: missed %d packets\n", s->queue->seq - s->seq - 1); | |
725 | ||
726 | /* Parse the first packet in the queue, and dequeue it */ | |
727 | rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len); | |
728 | next = s->queue->next; | |
729 | av_free(s->queue->buf); | |
730 | av_free(s->queue); | |
731 | s->queue = next; | |
732 | s->queue_len--; | |
733 | return rv; | |
734 | } | |
735 | ||
736 | static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt, | |
737 | uint8_t **bufptr, int len) | |
738 | { | |
739 | uint8_t *buf = bufptr ? *bufptr : NULL; | |
740 | int flags = 0; | |
741 | uint32_t timestamp; | |
742 | int rv = 0; | |
743 | ||
744 | if (!buf) { | |
745 | /* If parsing of the previous packet actually returned 0 or an error, | |
746 | * there's nothing more to be parsed from that packet, but we may have | |
747 | * indicated that we can return the next enqueued packet. */ | |
748 | if (s->prev_ret <= 0) | |
749 | return rtp_parse_queued_packet(s, pkt); | |
750 | /* return the next packets, if any */ | |
751 | if (s->handler && s->handler->parse_packet) { | |
752 | /* timestamp should be overwritten by parse_packet, if not, | |
753 | * the packet is left with pts == AV_NOPTS_VALUE */ | |
754 | timestamp = RTP_NOTS_VALUE; | |
755 | rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context, | |
756 | s->st, pkt, ×tamp, NULL, 0, 0, | |
757 | flags); | |
758 | finalize_packet(s, pkt, timestamp); | |
759 | return rv; | |
760 | } | |
761 | } | |
762 | ||
763 | if (len < 12) | |
764 | return -1; | |
765 | ||
766 | if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) | |
767 | return -1; | |
768 | if (RTP_PT_IS_RTCP(buf[1])) { | |
769 | return rtcp_parse_packet(s, buf, len); | |
770 | } | |
771 | ||
772 | if (s->st) { | |
773 | int64_t received = av_gettime(); | |
774 | uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q, | |
775 | s->st->time_base); | |
776 | timestamp = AV_RB32(buf + 4); | |
777 | // Calculate the jitter immediately, before queueing the packet | |
778 | // into the reordering queue. | |
779 | rtcp_update_jitter(&s->statistics, timestamp, arrival_ts); | |
780 | } | |
781 | ||
782 | if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) { | |
783 | /* First packet, or no reordering */ | |
784 | return rtp_parse_packet_internal(s, pkt, buf, len); | |
785 | } else { | |
786 | uint16_t seq = AV_RB16(buf + 2); | |
787 | int16_t diff = seq - s->seq; | |
788 | if (diff < 0) { | |
789 | /* Packet older than the previously emitted one, drop */ | |
790 | av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING, | |
791 | "RTP: dropping old packet received too late\n"); | |
792 | return -1; | |
793 | } else if (diff <= 1) { | |
794 | /* Correct packet */ | |
795 | rv = rtp_parse_packet_internal(s, pkt, buf, len); | |
796 | return rv; | |
797 | } else { | |
798 | /* Still missing some packet, enqueue this one. */ | |
799 | enqueue_packet(s, buf, len); | |
800 | *bufptr = NULL; | |
801 | /* Return the first enqueued packet if the queue is full, | |
802 | * even if we're missing something */ | |
803 | if (s->queue_len >= s->queue_size) | |
804 | return rtp_parse_queued_packet(s, pkt); | |
805 | return -1; | |
806 | } | |
807 | } | |
808 | } | |
809 | ||
810 | /** | |
811 | * Parse an RTP or RTCP packet directly sent as a buffer. | |
812 | * @param s RTP parse context. | |
813 | * @param pkt returned packet | |
814 | * @param bufptr pointer to the input buffer or NULL to read the next packets | |
815 | * @param len buffer len | |
816 | * @return 0 if a packet is returned, 1 if a packet is returned and more can follow | |
817 | * (use buf as NULL to read the next). -1 if no packet (error or no more packet). | |
818 | */ | |
819 | int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, | |
820 | uint8_t **bufptr, int len) | |
821 | { | |
822 | int rv; | |
823 | if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0) | |
824 | return -1; | |
825 | rv = rtp_parse_one_packet(s, pkt, bufptr, len); | |
826 | s->prev_ret = rv; | |
827 | while (rv == AVERROR(EAGAIN) && has_next_packet(s)) | |
828 | rv = rtp_parse_queued_packet(s, pkt); | |
829 | return rv ? rv : has_next_packet(s); | |
830 | } | |
831 | ||
832 | void ff_rtp_parse_close(RTPDemuxContext *s) | |
833 | { | |
834 | ff_rtp_reset_packet_queue(s); | |
835 | ff_srtp_free(&s->srtp); | |
836 | av_free(s); | |
837 | } | |
838 | ||
839 | int ff_parse_fmtp(AVFormatContext *s, | |
840 | AVStream *stream, PayloadContext *data, const char *p, | |
841 | int (*parse_fmtp)(AVFormatContext *s, | |
842 | AVStream *stream, | |
843 | PayloadContext *data, | |
844 | char *attr, char *value)) | |
845 | { | |
846 | char attr[256]; | |
847 | char *value; | |
848 | int res; | |
849 | int value_size = strlen(p) + 1; | |
850 | ||
851 | if (!(value = av_malloc(value_size))) { | |
852 | av_log(NULL, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n"); | |
853 | return AVERROR(ENOMEM); | |
854 | } | |
855 | ||
856 | // remove protocol identifier | |
857 | while (*p && *p == ' ') | |
858 | p++; // strip spaces | |
859 | while (*p && *p != ' ') | |
860 | p++; // eat protocol identifier | |
861 | while (*p && *p == ' ') | |
862 | p++; // strip trailing spaces | |
863 | ||
864 | while (ff_rtsp_next_attr_and_value(&p, | |
865 | attr, sizeof(attr), | |
866 | value, value_size)) { | |
867 | res = parse_fmtp(s, stream, data, attr, value); | |
868 | if (res < 0 && res != AVERROR_PATCHWELCOME) { | |
869 | av_free(value); | |
870 | return res; | |
871 | } | |
872 | } | |
873 | av_free(value); | |
874 | return 0; | |
875 | } | |
876 | ||
877 | int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx) | |
878 | { | |
879 | int ret; | |
880 | av_init_packet(pkt); | |
881 | ||
882 | pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data); | |
883 | pkt->stream_index = stream_idx; | |
884 | *dyn_buf = NULL; | |
885 | if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) { | |
886 | av_freep(&pkt->data); | |
887 | return ret; | |
888 | } | |
889 | return pkt->size; | |
890 | } |