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