FFmpeg coverage

Line	Branch	Exec	Source
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/intreadwrite.h"
25			#include "libavutil/mem.h"
26			#include "libavutil/time.h"
27
28			#include "libavcodec/bytestream.h"
29
30			#include "avformat.h"
31			#include "network.h"
32			#include "srtp.h"
33			#include "url.h"
34			#include "rtpdec.h"
35			#include "rtpdec_formats.h"
36			#include "internal.h"
37
38			#define MIN_FEEDBACK_INTERVAL 200000 /* 200 ms in us */
39
40			static const RTPDynamicProtocolHandler l24_dynamic_handler = {
41			.enc_name = "L24",
42			.codec_type = AVMEDIA_TYPE_AUDIO,
43			.codec_id = AV_CODEC_ID_PCM_S24BE,
44			};
45
46			static const RTPDynamicProtocolHandler gsm_dynamic_handler = {
47			.enc_name = "GSM",
48			.codec_type = AVMEDIA_TYPE_AUDIO,
49			.codec_id = AV_CODEC_ID_GSM,
50			};
51
52			static const RTPDynamicProtocolHandler realmedia_mp3_dynamic_handler = {
53			.enc_name = "X-MP3-draft-00",
54			.codec_type = AVMEDIA_TYPE_AUDIO,
55			.codec_id = AV_CODEC_ID_MP3ADU,
56			};
57
58			static const RTPDynamicProtocolHandler speex_dynamic_handler = {
59			.enc_name = "speex",
60			.codec_type = AVMEDIA_TYPE_AUDIO,
61			.codec_id = AV_CODEC_ID_SPEEX,
62			};
63
64			static const RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
65			.enc_name = "t140",
66			.codec_type = AVMEDIA_TYPE_SUBTITLE,
67			.codec_id = AV_CODEC_ID_TEXT,
68			};
69
70			extern const RTPDynamicProtocolHandler ff_rdt_video_handler;
71			extern const RTPDynamicProtocolHandler ff_rdt_audio_handler;
72			extern const RTPDynamicProtocolHandler ff_rdt_live_video_handler;
73			extern const RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
74
75			static const RTPDynamicProtocolHandler *const rtp_dynamic_protocol_handler_list[] = {
76			/* rtp */
77			&ff_ac3_dynamic_handler,
78			&ff_amr_nb_dynamic_handler,
79			&ff_amr_wb_dynamic_handler,
80			&ff_av1_dynamic_handler,
81			&ff_dv_dynamic_handler,
82			&ff_g726_16_dynamic_handler,
83			&ff_g726_24_dynamic_handler,
84			&ff_g726_32_dynamic_handler,
85			&ff_g726_40_dynamic_handler,
86			&ff_g726le_16_dynamic_handler,
87			&ff_g726le_24_dynamic_handler,
88			&ff_g726le_32_dynamic_handler,
89			&ff_g726le_40_dynamic_handler,
90			&ff_h261_dynamic_handler,
91			&ff_h263_1998_dynamic_handler,
92			&ff_h263_2000_dynamic_handler,
93			&ff_h263_rfc2190_dynamic_handler,
94			&ff_h264_dynamic_handler,
95			&ff_hevc_dynamic_handler,
96			&ff_ilbc_dynamic_handler,
97			&ff_jpeg_dynamic_handler,
98			&ff_mp4a_latm_dynamic_handler,
99			&ff_mp4v_es_dynamic_handler,
100			&ff_mpeg_audio_dynamic_handler,
101			&ff_mpeg_audio_robust_dynamic_handler,
102			&ff_mpeg_video_dynamic_handler,
103			&ff_mpeg4_generic_dynamic_handler,
104			&ff_mpegts_dynamic_handler,
105			&ff_ms_rtp_asf_pfa_handler,
106			&ff_ms_rtp_asf_pfv_handler,
107			&ff_qcelp_dynamic_handler,
108			&ff_qdm2_dynamic_handler,
109			&ff_qt_rtp_aud_handler,
110			&ff_qt_rtp_vid_handler,
111			&ff_quicktime_rtp_aud_handler,
112			&ff_quicktime_rtp_vid_handler,
113			&ff_rfc4175_rtp_handler,
114			&ff_svq3_dynamic_handler,
115			&ff_theora_dynamic_handler,
116			&ff_vc2hq_dynamic_handler,
117			&ff_vorbis_dynamic_handler,
118			&ff_vp8_dynamic_handler,
119			&ff_vp9_dynamic_handler,
120			&gsm_dynamic_handler,
121			&l24_dynamic_handler,
122			&ff_opus_dynamic_handler,
123			&realmedia_mp3_dynamic_handler,
124			&speex_dynamic_handler,
125			&t140_dynamic_handler,
126			/* rdt */
127			&ff_rdt_video_handler,
128			&ff_rdt_audio_handler,
129			&ff_rdt_live_video_handler,
130			&ff_rdt_live_audio_handler,
131			NULL,
132			};
133
134			/**
135			* Iterate over all registered rtp dynamic protocol handlers.
136			*
137			* @param opaque a pointer where libavformat will store the iteration state.
138			* Must point to NULL to start the iteration.
139			*
140			* @return the next registered rtp dynamic protocol handler
141			* or NULL when the iteration is finished
142			*/
143		✗	static const RTPDynamicProtocolHandler *rtp_handler_iterate(void **opaque)
144			{
145		✗	uintptr_t i = (uintptr_t)*opaque;
146		✗	const RTPDynamicProtocolHandler *r = rtp_dynamic_protocol_handler_list[i];
147
148		✗	if (r)
149		✗	*opaque = (void*)(i + 1);
150
151		✗	return r;
152			}
153
154		✗	const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
155			enum AVMediaType codec_type)
156			{
157		✗	void *i = 0;
158			const RTPDynamicProtocolHandler *handler;
159		✗	while (handler = rtp_handler_iterate(&i)) {
160		✗	if (handler->enc_name &&
161		✗	!av_strcasecmp(name, handler->enc_name) &&
162		✗	codec_type == handler->codec_type)
163		✗	return handler;
164			}
165		✗	return NULL;
166			}
167
168		✗	const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
169			enum AVMediaType codec_type)
170			{
171		✗	void *i = 0;
172			const RTPDynamicProtocolHandler *handler;
173		✗	while (handler = rtp_handler_iterate(&i)) {
174		✗	if (handler->static_payload_id && handler->static_payload_id == id &&
175		✗	codec_type == handler->codec_type)
176		✗	return handler;
177			}
178		✗	return NULL;
179			}
180
181		✗	static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
182			int len)
183			{
184			int payload_len;
185		✗	while (len >= 4) {
186		✗	payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
187
188		✗	switch (buf[1]) {
189		✗	case RTCP_SR:
190		✗	if (payload_len < 28) {
191		✗	av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
192		✗	return AVERROR_INVALIDDATA;
193			}
194
195		✗	s->last_sr.ssrc = AV_RB32(buf + 4);
196		✗	s->last_sr.ntp_timestamp = AV_RB64(buf + 8);
197		✗	s->last_sr.rtp_timestamp = AV_RB32(buf + 16);
198		✗	s->last_sr.sender_nb_packets = AV_RB32(buf + 20);
199		✗	s->last_sr.sender_nb_bytes = AV_RB32(buf + 24);
200
201		✗	s->pending_sr = 1;
202		✗	s->last_rtcp_reception_time = av_gettime_relative();
203
204		✗	if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
205		✗	s->first_rtcp_ntp_time = s->last_sr.ntp_timestamp;
206		✗	if (!s->base_timestamp)
207		✗	s->base_timestamp = s->last_sr.rtp_timestamp;
208		✗	s->rtcp_ts_offset = (int32_t)(s->last_sr.rtp_timestamp - s->base_timestamp);
209			}
210
211		✗	break;
212		✗	case RTCP_BYE:
213		✗	return -RTCP_BYE;
214			}
215
216		✗	buf += payload_len;
217		✗	len -= payload_len;
218			}
219		✗	return -1;
220			}
221
222			#define RTP_SEQ_MOD (1 << 16)
223
224		✗	static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
225			{
226		✗	memset(s, 0, sizeof(RTPStatistics));
227		✗	s->max_seq = base_sequence;
228		✗	s->probation = 1;
229		✗	}
230
231			/*
232			* Called whenever there is a large jump in sequence numbers,
233			* or when they get out of probation...
234			*/
235		✗	static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
236			{
237		✗	s->max_seq = seq;
238		✗	s->cycles = 0;
239		✗	s->base_seq = seq - 1;
240		✗	s->bad_seq = RTP_SEQ_MOD + 1;
241		✗	s->received = 0;
242		✗	s->expected_prior = 0;
243		✗	s->received_prior = 0;
244		✗	s->jitter = 0;
245		✗	s->transit = 0;
246		✗	}
247
248			/* Returns 1 if we should handle this packet. */
249		✗	static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
250			{
251		✗	uint16_t udelta = seq - s->max_seq;
252		✗	const int MAX_DROPOUT = 3000;
253		✗	const int MAX_MISORDER = 100;
254		✗	const int MIN_SEQUENTIAL = 2;
255
256			/* source not valid until MIN_SEQUENTIAL packets with sequence
257			* seq. numbers have been received */
258		✗	if (s->probation) {
259		✗	if (seq == s->max_seq + 1) {
260		✗	s->probation--;
261		✗	s->max_seq = seq;
262		✗	if (s->probation == 0) {
263		✗	rtp_init_sequence(s, seq);
264		✗	s->received++;
265		✗	return 1;
266			}
267			} else {
268		✗	s->probation = MIN_SEQUENTIAL - 1;
269		✗	s->max_seq = seq;
270			}
271		✗	} else if (udelta < MAX_DROPOUT) {
272			// in order, with permissible gap
273		✗	if (seq < s->max_seq) {
274			// sequence number wrapped; count another 64k cycles
275		✗	s->cycles += RTP_SEQ_MOD;
276			}
277		✗	s->max_seq = seq;
278		✗	} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
279			// sequence made a large jump...
280		✗	if (seq == s->bad_seq) {
281			/* two sequential packets -- assume that the other side
282			* restarted without telling us; just resync. */
283		✗	rtp_init_sequence(s, seq);
284			} else {
285		✗	s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
286		✗	return 0;
287			}
288			} else {
289			// duplicate or reordered packet...
290			}
291		✗	s->received++;
292		✗	return 1;
293			}
294
295		✗	static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
296			uint32_t arrival_timestamp)
297			{
298			// Most of this is pretty straight from RFC 3550 appendix A.8
299		✗	uint32_t transit = arrival_timestamp - sent_timestamp;
300		✗	uint32_t prev_transit = s->transit;
301		✗	int32_t d = transit - prev_transit;
302			// Doing the FFABS() call directly on the "transit - prev_transit"
303			// expression doesn't work, since it's an unsigned expression. Doing the
304			// transit calculation in unsigned is desired though, since it most
305			// probably will need to wrap around.
306		✗	d = FFABS(d);
307		✗	s->transit = transit;
308		✗	if (!prev_transit)
309		✗	return;
310		✗	s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
311			}
312
313		✗	int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
314			AVIOContext *avio, int count)
315			{
316			AVIOContext *pb;
317			uint8_t *buf;
318			int len;
319			int rtcp_bytes;
320		✗	RTPStatistics *stats = &s->statistics;
321			uint32_t lost;
322			uint32_t extended_max;
323			uint32_t expected_interval;
324			uint32_t received_interval;
325			int32_t lost_interval;
326			uint32_t expected;
327			uint32_t fraction;
328
329		✗	if ((!fd && !avio) || (count < 1))
330		✗	return -1;
331
332			/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
333			/* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
334		✗	s->octet_count += count;
335		✗	rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
336			RTCP_TX_RATIO_DEN;
337		✗	rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
338		✗	if (rtcp_bytes < 28)
339		✗	return -1;
340		✗	s->last_octet_count = s->octet_count;
341
342		✗	if (!fd)
343		✗	pb = avio;
344		✗	else if (avio_open_dyn_buf(&pb) < 0)
345		✗	return -1;
346
347			// Receiver Report
348		✗	avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
349		✗	avio_w8(pb, RTCP_RR);
350		✗	avio_wb16(pb, 7); /* length in words - 1 */
351			// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
352		✗	avio_wb32(pb, s->ssrc + 1);
353		✗	avio_wb32(pb, s->ssrc); // server SSRC
354			// some placeholders we should really fill...
355			// RFC 1889/p64
356		✗	extended_max = stats->cycles + stats->max_seq;
357		✗	expected = extended_max - stats->base_seq;
358		✗	lost = expected - stats->received;
359		✗	lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
360		✗	expected_interval = expected - stats->expected_prior;
361		✗	stats->expected_prior = expected;
362		✗	received_interval = stats->received - stats->received_prior;
363		✗	stats->received_prior = stats->received;
364		✗	lost_interval = expected_interval - received_interval;
365		✗	if (expected_interval == 0 || lost_interval <= 0)
366		✗	fraction = 0;
367			else
368		✗	fraction = (lost_interval << 8) / expected_interval;
369
370		✗	fraction = (fraction << 24) | lost;
371
372		✗	avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
373		✗	avio_wb32(pb, extended_max); /* max sequence received */
374		✗	avio_wb32(pb, stats->jitter >> 4); /* jitter */
375
376		✗	if (s->last_sr.ntp_timestamp == AV_NOPTS_VALUE) {
377		✗	avio_wb32(pb, 0); /* last SR timestamp */
378		✗	avio_wb32(pb, 0); /* delay since last SR */
379			} else {
380		✗	uint32_t middle_32_bits = s->last_sr.ntp_timestamp >> 16; // this is valid, right? do we need to handle 64 bit values special?
381		✗	uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
382			65536, AV_TIME_BASE);
383
384		✗	avio_wb32(pb, middle_32_bits); /* last SR timestamp */
385		✗	avio_wb32(pb, delay_since_last); /* delay since last SR */
386			}
387
388			// CNAME
389		✗	avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
390		✗	avio_w8(pb, RTCP_SDES);
391		✗	len = strlen(s->hostname);
392		✗	avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
393		✗	avio_wb32(pb, s->ssrc + 1);
394		✗	avio_w8(pb, 0x01);
395		✗	avio_w8(pb, len);
396		✗	avio_write(pb, s->hostname, len);
397		✗	avio_w8(pb, 0); /* END */
398			// padding
399		✗	for (len = (7 + len) % 4; len % 4; len++)
400		✗	avio_w8(pb, 0);
401
402		✗	avio_flush(pb);
403		✗	if (!fd)
404		✗	return 0;
405		✗	len = avio_close_dyn_buf(pb, &buf);
406		✗	if ((len > 0) && buf) {
407			int av_unused result;
408		✗	av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
409		✗	result = ffurl_write(fd, buf, len);
410		✗	av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
411		✗	av_free(buf);
412			}
413		✗	return 0;
414			}
415
416		✗	void ff_rtp_send_punch_packets(URLContext *rtp_handle)
417			{
418		✗	uint8_t buf[RTP_MIN_PACKET_LENGTH], *ptr = buf;
419
420			/* Send a small RTP packet */
421
422		✗	bytestream_put_byte(&ptr, (RTP_VERSION << 6));
423		✗	bytestream_put_byte(&ptr, 0); /* Payload type */
424		✗	bytestream_put_be16(&ptr, 0); /* Seq */
425		✗	bytestream_put_be32(&ptr, 0); /* Timestamp */
426		✗	bytestream_put_be32(&ptr, 0); /* SSRC */
427
428		✗	ffurl_write(rtp_handle, buf, ptr - buf);
429
430			/* Send a minimal RTCP RR */
431		✗	ptr = buf;
432		✗	bytestream_put_byte(&ptr, (RTP_VERSION << 6));
433		✗	bytestream_put_byte(&ptr, RTCP_RR); /* receiver report */
434		✗	bytestream_put_be16(&ptr, 1); /* length in words - 1 */
435		✗	bytestream_put_be32(&ptr, 0); /* our own SSRC */
436
437		✗	ffurl_write(rtp_handle, buf, ptr - buf);
438		✗	}
439
440		✗	static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
441			uint16_t *missing_mask)
442			{
443			int i;
444		✗	uint16_t next_seq = s->seq + 1;
445		✗	RTPPacket *pkt = s->queue;
446
447		✗	if (!pkt || pkt->seq == next_seq)
448		✗	return 0;
449
450		✗	*missing_mask = 0;
451		✗	for (i = 1; i <= 16; i++) {
452		✗	uint16_t missing_seq = next_seq + i;
453		✗	while (pkt) {
454		✗	int16_t diff = pkt->seq - missing_seq;
455		✗	if (diff >= 0)
456		✗	break;
457		✗	pkt = pkt->next;
458			}
459		✗	if (!pkt)
460		✗	break;
461		✗	if (pkt->seq == missing_seq)
462		✗	continue;
463		✗	*missing_mask |= 1 << (i - 1);
464			}
465
466		✗	*first_missing = next_seq;
467		✗	return 1;
468			}
469
470		✗	int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
471			AVIOContext *avio)
472			{
473			int len, need_keyframe, missing_packets;
474			AVIOContext *pb;
475			uint8_t *buf;
476			int64_t now;
477		✗	uint16_t first_missing = 0, missing_mask = 0;
478
479		✗	if (!fd && !avio)
480		✗	return -1;
481
482		✗	need_keyframe = s->handler && s->handler->need_keyframe &&
483		✗	s->handler->need_keyframe(s->dynamic_protocol_context);
484		✗	missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
485
486		✗	if (!need_keyframe && !missing_packets)
487		✗	return 0;
488
489			/* Send new feedback if enough time has elapsed since the last
490			* feedback packet. */
491
492		✗	now = av_gettime_relative();
493		✗	if (s->last_feedback_time &&
494		✗	(now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
495		✗	return 0;
496		✗	s->last_feedback_time = now;
497
498		✗	if (!fd)
499		✗	pb = avio;
500		✗	else if (avio_open_dyn_buf(&pb) < 0)
501		✗	return -1;
502
503		✗	if (need_keyframe) {
504		✗	avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
505		✗	avio_w8(pb, RTCP_PSFB);
506		✗	avio_wb16(pb, 2); /* length in words - 1 */
507			// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
508		✗	avio_wb32(pb, s->ssrc + 1);
509		✗	avio_wb32(pb, s->ssrc); // server SSRC
510			}
511
512		✗	if (missing_packets) {
513		✗	avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
514		✗	avio_w8(pb, RTCP_RTPFB);
515		✗	avio_wb16(pb, 3); /* length in words - 1 */
516		✗	avio_wb32(pb, s->ssrc + 1);
517		✗	avio_wb32(pb, s->ssrc); // server SSRC
518
519		✗	avio_wb16(pb, first_missing);
520		✗	avio_wb16(pb, missing_mask);
521			}
522
523		✗	avio_flush(pb);
524		✗	if (!fd)
525		✗	return 0;
526		✗	len = avio_close_dyn_buf(pb, &buf);
527		✗	if (len > 0 && buf) {
528		✗	ffurl_write(fd, buf, len);
529		✗	av_free(buf);
530			}
531		✗	return 0;
532			}
533
534			/**
535			* open a new RTP parse context for stream 'st'. 'st' can be NULL for
536			* MPEG-2 TS streams.
537			*/
538		✗	RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
539			int payload_type, int queue_size)
540			{
541			RTPDemuxContext *s;
542
543		✗	s = av_mallocz(sizeof(RTPDemuxContext));
544		✗	if (!s)
545		✗	return NULL;
546		✗	s->payload_type = payload_type;
547		✗	s->last_sr.ntp_timestamp = AV_NOPTS_VALUE;
548		✗	s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
549		✗	s->ic = s1;
550		✗	s->st = st;
551		✗	s->queue_size = queue_size;
552
553		✗	av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
554			s->queue_size);
555
556		✗	rtp_init_statistics(&s->statistics, 0);
557		✗	if (st) {
558		✗	switch (st->codecpar->codec_id) {
559		✗	case AV_CODEC_ID_ADPCM_G722:
560			/* According to RFC 3551, the stream clock rate is 8000
561			* even if the sample rate is 16000. */
562		✗	if (st->codecpar->sample_rate == 8000)
563		✗	st->codecpar->sample_rate = 16000;
564		✗	break;
565		✗	case AV_CODEC_ID_PCM_MULAW: {
566		✗	AVCodecParameters *par = st->codecpar;
567		✗	par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id);
568		✗	par->block_align = par->ch_layout.nb_channels * par->bits_per_coded_sample / 8;
569		✗	par->bit_rate = par->block_align * 8LL * par->sample_rate;
570		✗	break;
571			}
572		✗	default:
573		✗	break;
574			}
575			}
576			// needed to send back RTCP RR in RTSP sessions
577		✗	gethostname(s->hostname, sizeof(s->hostname));
578		✗	return s;
579			}
580
581		✗	void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
582			const RTPDynamicProtocolHandler *handler)
583			{
584		✗	s->dynamic_protocol_context = ctx;
585		✗	s->handler = handler;
586		✗	}
587
588		✗	void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
589			const char *params)
590			{
591		✗	if (!ff_srtp_set_crypto(&s->srtp, suite, params))
592		✗	s->srtp_enabled = 1;
593		✗	}
594
595		✗	static int rtp_set_prft(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) {
596			int64_t rtcp_time, delta_time;
597			int32_t delta_timestamp;
598
599			AVProducerReferenceTime *prft =
600		✗	(AVProducerReferenceTime *) av_packet_new_side_data(
601			pkt, AV_PKT_DATA_PRFT, sizeof(AVProducerReferenceTime));
602		✗	if (!prft)
603		✗	return AVERROR(ENOMEM);
604
605		✗	rtcp_time = ff_parse_ntp_time(s->last_sr.ntp_timestamp) - NTP_OFFSET_US;
606			/* Cast to int32_t to handle timestamp wraparound correctly */
607		✗	delta_timestamp = (int32_t)(timestamp - s->last_sr.rtp_timestamp);
608		✗	delta_time = av_rescale_q(delta_timestamp, s->st->time_base, AV_TIME_BASE_Q);
609
610		✗	prft->wallclock = rtcp_time + delta_time;
611		✗	prft->flags = 24;
612		✗	return 0;
613			}
614
615		✗	static int rtp_add_sr_sidedata(RTPDemuxContext *s, AVPacket *pkt) {
616			AVRTCPSenderReport *sr =
617		✗	(AVRTCPSenderReport *) av_packet_new_side_data(
618			pkt, AV_PKT_DATA_RTCP_SR, sizeof(AVRTCPSenderReport));
619		✗	if (!sr)
620		✗	return AVERROR(ENOMEM);
621
622		✗	memcpy(sr, &s->last_sr, sizeof(AVRTCPSenderReport));
623		✗	s->pending_sr = 0;
624		✗	return 0;
625			}
626
627			/**
628			* This was the second switch in rtp_parse packet.
629			* Normalizes time, if required, sets stream_index, etc.
630			*/
631		✗	static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
632			{
633		✗	if (s->pending_sr) {
634		✗	int ret = rtp_add_sr_sidedata(s, pkt);
635		✗	if (ret < 0)
636		✗	av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to add SR sidedata\n");
637			}
638
639		✗	if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
640		✗	return; /* Timestamp already set by depacketizer */
641		✗	if (timestamp == RTP_NOTS_VALUE)
642		✗	return;
643
644		✗	if (s->last_sr.ntp_timestamp != AV_NOPTS_VALUE) {
645		✗	if (rtp_set_prft(s, pkt, timestamp) < 0) {
646		✗	av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to set prft");
647			}
648			}
649
650		✗	if (s->last_sr.ntp_timestamp != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
651			int64_t addend;
652			int32_t delta_timestamp;
653
654			/* compute pts from timestamp with received ntp_time */
655			/* Cast to int32_t to handle timestamp wraparound correctly */
656		✗	delta_timestamp = (int32_t)(timestamp - s->last_sr.rtp_timestamp);
657			/* convert to the PTS timebase */
658		✗	addend = av_rescale(s->last_sr.ntp_timestamp - s->first_rtcp_ntp_time,
659		✗	s->st->time_base.den,
660		✗	(uint64_t) s->st->time_base.num << 32);
661		✗	pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
662			delta_timestamp;
663		✗	return;
664			}
665
666		✗	if (!s->base_timestamp)
667		✗	s->base_timestamp = timestamp;
668			/* assume that the difference is INT32_MIN < x < INT32_MAX,
669			* but allow the first timestamp to exceed INT32_MAX */
670		✗	if (!s->timestamp)
671		✗	s->unwrapped_timestamp += timestamp;
672			else
673		✗	s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
674		✗	s->timestamp = timestamp;
675		✗	pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
676		✗	s->base_timestamp;
677			}
678
679		✗	static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
680			const uint8_t *buf, int len)
681			{
682			unsigned int ssrc;
683		✗	int payload_type, seq, flags = 0;
684			int ext, csrc;
685			AVStream *st;
686			uint32_t timestamp;
687		✗	int rv = 0;
688
689		✗	csrc = buf[0] & 0x0f;
690		✗	ext = buf[0] & 0x10;
691		✗	payload_type = buf[1] & 0x7f;
692		✗	if (buf[1] & 0x80)
693		✗	flags |= RTP_FLAG_MARKER;
694		✗	seq = AV_RB16(buf + 2);
695		✗	timestamp = AV_RB32(buf + 4);
696		✗	ssrc = AV_RB32(buf + 8);
697			/* store the ssrc in the RTPDemuxContext */
698		✗	s->ssrc = ssrc;
699
700			/* NOTE: we can handle only one payload type */
701		✗	if (s->payload_type != payload_type)
702		✗	return -1;
703
704		✗	st = s->st;
705			// only do something with this if all the rtp checks pass...
706		✗	if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
707		✗	av_log(s->ic, AV_LOG_ERROR,
708			"RTP: PT=%02x: bad cseq %04x expected=%04x\n",
709		✗	payload_type, seq, ((s->seq + 1) & 0xffff));
710		✗	return -1;
711			}
712
713		✗	if (buf[0] & 0x20) {
714		✗	int padding = buf[len - 1];
715		✗	if (len >= 12 + padding)
716		✗	len -= padding;
717			}
718
719		✗	s->seq = seq;
720		✗	len -= 12;
721		✗	buf += 12;
722
723		✗	len -= 4 * csrc;
724		✗	buf += 4 * csrc;
725		✗	if (len < 0)
726		✗	return AVERROR_INVALIDDATA;
727
728			/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
729		✗	if (ext) {
730		✗	if (len < 4)
731		✗	return -1;
732			/* calculate the header extension length (stored as number
733			* of 32-bit words) */
734		✗	ext = (AV_RB16(buf + 2) + 1) << 2;
735
736		✗	if (len < ext)
737		✗	return -1;
738			// skip past RTP header extension
739		✗	len -= ext;
740		✗	buf += ext;
741			}
742
743		✗	if (s->handler && s->handler->parse_packet) {
744		✗	rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
745			s->st, pkt, &timestamp, buf, len, seq,
746			flags);
747		✗	} else if (st) {
748		✗	if ((rv = av_new_packet(pkt, len)) < 0)
749		✗	return rv;
750		✗	memcpy(pkt->data, buf, len);
751		✗	pkt->stream_index = st->index;
752			} else {
753		✗	return AVERROR(EINVAL);
754			}
755
756			// now perform timestamp things....
757		✗	finalize_packet(s, pkt, timestamp);
758
759		✗	return rv;
760			}
761
762		✗	void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
763			{
764		✗	while (s->queue) {
765		✗	RTPPacket *next = s->queue->next;
766		✗	av_freep(&s->queue->buf);
767		✗	av_freep(&s->queue);
768		✗	s->queue = next;
769			}
770		✗	s->seq = 0;
771		✗	s->queue_len = 0;
772		✗	s->prev_ret = 0;
773		✗	}
774
775		✗	static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
776			{
777		✗	uint16_t seq = AV_RB16(buf + 2);
778		✗	RTPPacket **cur = &s->queue, *packet;
779
780			/* Find the correct place in the queue to insert the packet */
781		✗	while (*cur) {
782		✗	int16_t diff = seq - (*cur)->seq;
783		✗	if (diff < 0)
784		✗	break;
785		✗	cur = &(*cur)->next;
786			}
787
788		✗	packet = av_mallocz(sizeof(*packet));
789		✗	if (!packet)
790		✗	return AVERROR(ENOMEM);
791		✗	packet->recvtime = av_gettime_relative();
792		✗	packet->seq = seq;
793		✗	packet->len = len;
794		✗	packet->buf = buf;
795		✗	packet->next = *cur;
796		✗	*cur = packet;
797		✗	s->queue_len++;
798
799		✗	return 0;
800			}
801
802		✗	static int has_next_packet(RTPDemuxContext *s)
803			{
804		✗	return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
805			}
806
807		✗	int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
808			{
809		✗	return s->queue ? s->queue->recvtime : 0;
810			}
811
812		✗	static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
813			{
814			int rv;
815			RTPPacket *next;
816
817		✗	if (s->queue_len <= 0)
818		✗	return -1;
819
820		✗	if (!has_next_packet(s)) {
821		✗	int pkt_missed = s->queue->seq - s->seq - 1;
822
823		✗	if (pkt_missed < 0)
824		✗	pkt_missed += UINT16_MAX;
825		✗	av_log(s->ic, AV_LOG_WARNING,
826			"RTP: missed %d packets\n", pkt_missed);
827			}
828
829			/* Parse the first packet in the queue, and dequeue it */
830		✗	rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
831		✗	next = s->queue->next;
832		✗	av_freep(&s->queue->buf);
833		✗	av_freep(&s->queue);
834		✗	s->queue = next;
835		✗	s->queue_len--;
836		✗	return rv;
837			}
838
839		✗	static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
840			uint8_t **bufptr, int len)
841			{
842		✗	uint8_t *buf = bufptr ? *bufptr : NULL;
843		✗	int flags = 0;
844			uint32_t timestamp;
845		✗	int rv = 0;
846
847		✗	if (!buf) {
848			/* If parsing of the previous packet actually returned 0 or an error,
849			* there's nothing more to be parsed from that packet, but we may have
850			* indicated that we can return the next enqueued packet. */
851		✗	if (s->prev_ret <= 0)
852		✗	return rtp_parse_queued_packet(s, pkt);
853			/* return the next packets, if any */
854		✗	if (s->handler && s->handler->parse_packet) {
855			/* timestamp should be overwritten by parse_packet, if not,
856			* the packet is left with pts == AV_NOPTS_VALUE */
857		✗	timestamp = RTP_NOTS_VALUE;
858		✗	rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
859			s->st, pkt, &timestamp, NULL, 0, 0,
860			flags);
861		✗	finalize_packet(s, pkt, timestamp);
862		✗	return rv;
863			}
864			}
865
866		✗	if (len < 12)
867		✗	return -1;
868
869		✗	if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
870		✗	return -1;
871		✗	if (RTP_PT_IS_RTCP(buf[1])) {
872		✗	return rtcp_parse_packet(s, buf, len);
873			}
874
875		✗	if (s->st) {
876		✗	int64_t received = av_gettime_relative();
877		✗	uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
878		✗	s->st->time_base);
879		✗	timestamp = AV_RB32(buf + 4);
880			// Calculate the jitter immediately, before queueing the packet
881			// into the reordering queue.
882		✗	rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
883			}
884
885		✗	if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
886			/* First packet, or no reordering */
887		✗	return rtp_parse_packet_internal(s, pkt, buf, len);
888			} else {
889		✗	uint16_t seq = AV_RB16(buf + 2);
890		✗	int16_t diff = seq - s->seq;
891		✗	if (diff < 0) {
892			/* Packet older than the previously emitted one, drop */
893		✗	av_log(s->ic, AV_LOG_WARNING,
894			"RTP: dropping old packet received too late\n");
895		✗	return -1;
896		✗	} else if (diff <= 1) {
897			/* Correct packet */
898		✗	rv = rtp_parse_packet_internal(s, pkt, buf, len);
899		✗	return rv;
900			} else {
901			/* Still missing some packet, enqueue this one. */
902		✗	rv = enqueue_packet(s, buf, len);
903		✗	if (rv < 0)
904		✗	return rv;
905		✗	*bufptr = NULL;
906			/* Return the first enqueued packet if the queue is full,
907			* even if we're missing something */
908		✗	if (s->queue_len >= s->queue_size) {
909		✗	av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
910		✗	return rtp_parse_queued_packet(s, pkt);
911			}
912		✗	return -1;
913			}
914			}
915			}
916
917			/**
918			* Parse an RTP or RTCP packet directly sent as a buffer.
919			* @param s RTP parse context.
920			* @param pkt returned packet
921			* @param bufptr pointer to the input buffer or NULL to read the next packets
922			* @param len buffer len
923			* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
924			* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
925			*/
926		✗	int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
927			uint8_t **bufptr, int len)
928			{
929			int rv;
930		✗	if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
931		✗	return -1;
932		✗	rv = rtp_parse_one_packet(s, pkt, bufptr, len);
933		✗	s->prev_ret = rv;
934		✗	while (rv < 0 && has_next_packet(s))
935		✗	rv = rtp_parse_queued_packet(s, pkt);
936		✗	return rv ? rv : has_next_packet(s);
937			}
938
939		✗	void ff_rtp_parse_close(RTPDemuxContext *s)
940			{
941		✗	ff_rtp_reset_packet_queue(s);
942		✗	ff_srtp_free(&s->srtp);
943		✗	av_free(s);
944		✗	}
945
946		✗	int ff_parse_fmtp(AVFormatContext *s,
947			AVStream *stream, PayloadContext *data, const char *p,
948			int (*parse_fmtp)(AVFormatContext *s,
949			AVStream *stream,
950			PayloadContext *data,
951			const char *attr, const char *value))
952			{
953			char attr[256];
954			char *value;
955			int res;
956		✗	int value_size = strlen(p) + 1;
957
958		✗	if (!(value = av_malloc(value_size))) {
959		✗	av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
960		✗	return AVERROR(ENOMEM);
961			}
962
963			// remove protocol identifier
964		✗	while (*p && *p == ' ')
965		✗	p++; // strip spaces
966		✗	while (*p && *p != ' ')
967		✗	p++; // eat protocol identifier
968		✗	while (*p && *p == ' ')
969		✗	p++; // strip trailing spaces
970
971		✗	while (ff_rtsp_next_attr_and_value(&p,
972			attr, sizeof(attr),
973			value, value_size)) {
974		✗	res = parse_fmtp(s, stream, data, attr, value);
975		✗	if (res < 0 && res != AVERROR_PATCHWELCOME) {
976		✗	av_free(value);
977		✗	return res;
978			}
979			}
980		✗	av_free(value);
981		✗	return 0;
982			}
983
984		✗	int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
985			{
986			int ret;
987		✗	av_packet_unref(pkt);
988
989		✗	pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
990		✗	pkt->stream_index = stream_idx;
991		✗	*dyn_buf = NULL;
992		✗	if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
993		✗	av_freep(&pkt->data);
994		✗	return ret;
995			}
996		✗	return pkt->size;
997			}
998