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 = av_zero_extend(av_clip_intp2(expected - stats->received, 23), 24);
359		✗	expected_interval = expected - stats->expected_prior;
360		✗	stats->expected_prior = expected;
361		✗	received_interval = stats->received - stats->received_prior;
362		✗	stats->received_prior = stats->received;
363		✗	lost_interval = expected_interval - received_interval;
364		✗	if (expected_interval == 0 || lost_interval <= 0)
365		✗	fraction = 0;
366			else
367		✗	fraction = (lost_interval << 8) / expected_interval;
368
369		✗	fraction = (fraction << 24) | lost;
370
371		✗	avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
372		✗	avio_wb32(pb, extended_max); /* max sequence received */
373		✗	avio_wb32(pb, stats->jitter >> 4); /* jitter */
374
375		✗	if (s->last_sr.ntp_timestamp == AV_NOPTS_VALUE) {
376		✗	avio_wb32(pb, 0); /* last SR timestamp */
377		✗	avio_wb32(pb, 0); /* delay since last SR */
378			} else {
379		✗	uint32_t middle_32_bits = s->last_sr.ntp_timestamp >> 16; // this is valid, right? do we need to handle 64 bit values special?
380		✗	uint32_t delay_since_last = av_rescale(av_gettime_relative() - s->last_rtcp_reception_time,
381			65536, AV_TIME_BASE);
382
383		✗	avio_wb32(pb, middle_32_bits); /* last SR timestamp */
384		✗	avio_wb32(pb, delay_since_last); /* delay since last SR */
385			}
386
387			// CNAME
388		✗	avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
389		✗	avio_w8(pb, RTCP_SDES);
390		✗	len = strlen(s->hostname);
391		✗	avio_wb16(pb, (7 + len + 3) / 4); /* length in words - 1 */
392		✗	avio_wb32(pb, s->ssrc + 1);
393		✗	avio_w8(pb, 0x01);
394		✗	avio_w8(pb, len);
395		✗	avio_write(pb, s->hostname, len);
396		✗	avio_w8(pb, 0); /* END */
397			// padding
398		✗	for (len = (7 + len) % 4; len % 4; len++)
399		✗	avio_w8(pb, 0);
400
401		✗	avio_flush(pb);
402		✗	if (!fd)
403		✗	return 0;
404		✗	len = avio_close_dyn_buf(pb, &buf);
405		✗	if ((len > 0) && buf) {
406			av_unused int result;
407		✗	av_log(s->ic, AV_LOG_TRACE, "sending %d bytes of RR\n", len);
408		✗	result = ffurl_write(fd, buf, len);
409		✗	av_log(s->ic, AV_LOG_TRACE, "result from ffurl_write: %d\n", result);
410		✗	av_free(buf);
411			}
412		✗	return 0;
413			}
414
415		✗	void ff_rtp_send_punch_packets(URLContext *rtp_handle)
416			{
417		✗	uint8_t buf[RTP_MIN_PACKET_LENGTH], *ptr = buf;
418
419			/* Send a small RTP packet */
420
421		✗	bytestream_put_byte(&ptr, (RTP_VERSION << 6));
422		✗	bytestream_put_byte(&ptr, 0); /* Payload type */
423		✗	bytestream_put_be16(&ptr, 0); /* Seq */
424		✗	bytestream_put_be32(&ptr, 0); /* Timestamp */
425		✗	bytestream_put_be32(&ptr, 0); /* SSRC */
426
427		✗	ffurl_write(rtp_handle, buf, ptr - buf);
428
429			/* Send a minimal RTCP RR */
430		✗	ptr = buf;
431		✗	bytestream_put_byte(&ptr, (RTP_VERSION << 6));
432		✗	bytestream_put_byte(&ptr, RTCP_RR); /* receiver report */
433		✗	bytestream_put_be16(&ptr, 1); /* length in words - 1 */
434		✗	bytestream_put_be32(&ptr, 0); /* our own SSRC */
435
436		✗	ffurl_write(rtp_handle, buf, ptr - buf);
437		✗	}
438
439		✗	static int find_missing_packets(RTPDemuxContext *s, uint16_t *first_missing,
440			uint16_t *missing_mask)
441			{
442			int i;
443		✗	uint16_t next_seq = s->seq + 1;
444		✗	RTPPacket *pkt = s->queue;
445
446		✗	if (!pkt || pkt->seq == next_seq)
447		✗	return 0;
448
449		✗	*missing_mask = 0;
450		✗	for (i = 1; i <= 16; i++) {
451		✗	uint16_t missing_seq = next_seq + i;
452		✗	while (pkt) {
453		✗	int16_t diff = pkt->seq - missing_seq;
454		✗	if (diff >= 0)
455		✗	break;
456		✗	pkt = pkt->next;
457			}
458		✗	if (!pkt)
459		✗	break;
460		✗	if (pkt->seq == missing_seq)
461		✗	continue;
462		✗	*missing_mask |= 1 << (i - 1);
463			}
464
465		✗	*first_missing = next_seq;
466		✗	return 1;
467			}
468
469		✗	int ff_rtp_send_rtcp_feedback(RTPDemuxContext *s, URLContext *fd,
470			AVIOContext *avio)
471			{
472			int len, need_keyframe, missing_packets;
473			AVIOContext *pb;
474			uint8_t *buf;
475			int64_t now;
476		✗	uint16_t first_missing = 0, missing_mask = 0;
477
478		✗	if (!fd && !avio)
479		✗	return -1;
480
481		✗	need_keyframe = s->handler && s->handler->need_keyframe &&
482		✗	s->handler->need_keyframe(s->dynamic_protocol_context);
483		✗	missing_packets = find_missing_packets(s, &first_missing, &missing_mask);
484
485		✗	if (!need_keyframe && !missing_packets)
486		✗	return 0;
487
488			/* Send new feedback if enough time has elapsed since the last
489			* feedback packet. */
490
491		✗	now = av_gettime_relative();
492		✗	if (s->last_feedback_time &&
493		✗	(now - s->last_feedback_time) < MIN_FEEDBACK_INTERVAL)
494		✗	return 0;
495		✗	s->last_feedback_time = now;
496
497		✗	if (!fd)
498		✗	pb = avio;
499		✗	else if (avio_open_dyn_buf(&pb) < 0)
500		✗	return -1;
501
502		✗	if (need_keyframe) {
503		✗	avio_w8(pb, (RTP_VERSION << 6) | 1); /* PLI */
504		✗	avio_w8(pb, RTCP_PSFB);
505		✗	avio_wb16(pb, 2); /* length in words - 1 */
506			// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
507		✗	avio_wb32(pb, s->ssrc + 1);
508		✗	avio_wb32(pb, s->ssrc); // server SSRC
509			}
510
511		✗	if (missing_packets) {
512		✗	avio_w8(pb, (RTP_VERSION << 6) | 1); /* NACK */
513		✗	avio_w8(pb, RTCP_RTPFB);
514		✗	avio_wb16(pb, 3); /* length in words - 1 */
515		✗	avio_wb32(pb, s->ssrc + 1);
516		✗	avio_wb32(pb, s->ssrc); // server SSRC
517
518		✗	avio_wb16(pb, first_missing);
519		✗	avio_wb16(pb, missing_mask);
520			}
521
522		✗	avio_flush(pb);
523		✗	if (!fd)
524		✗	return 0;
525		✗	len = avio_close_dyn_buf(pb, &buf);
526		✗	if (len > 0 && buf) {
527		✗	ffurl_write(fd, buf, len);
528		✗	av_free(buf);
529			}
530		✗	return 0;
531			}
532
533			/**
534			* open a new RTP parse context for stream 'st'. 'st' can be NULL for
535			* MPEG-2 TS streams.
536			*/
537		✗	RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
538			int payload_type, int queue_size)
539			{
540			RTPDemuxContext *s;
541
542		✗	s = av_mallocz(sizeof(RTPDemuxContext));
543		✗	if (!s)
544		✗	return NULL;
545		✗	s->payload_type = payload_type;
546		✗	s->last_sr.ntp_timestamp = AV_NOPTS_VALUE;
547		✗	s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
548		✗	s->ic = s1;
549		✗	s->st = st;
550		✗	s->queue_size = queue_size;
551
552		✗	av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
553			s->queue_size);
554
555		✗	rtp_init_statistics(&s->statistics, 0);
556		✗	if (st) {
557		✗	switch (st->codecpar->codec_id) {
558		✗	case AV_CODEC_ID_ADPCM_G722:
559			/* According to RFC 3551, the stream clock rate is 8000
560			* even if the sample rate is 16000. */
561		✗	if (st->codecpar->sample_rate == 8000)
562		✗	st->codecpar->sample_rate = 16000;
563		✗	break;
564		✗	case AV_CODEC_ID_PCM_MULAW: {
565		✗	AVCodecParameters *par = st->codecpar;
566		✗	par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id);
567		✗	par->block_align = par->ch_layout.nb_channels * par->bits_per_coded_sample / 8;
568		✗	par->bit_rate = par->block_align * 8LL * par->sample_rate;
569		✗	break;
570			}
571		✗	default:
572		✗	break;
573			}
574			}
575			// needed to send back RTCP RR in RTSP sessions
576		✗	gethostname(s->hostname, sizeof(s->hostname));
577		✗	return s;
578			}
579
580		✗	void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
581			const RTPDynamicProtocolHandler *handler)
582			{
583		✗	s->dynamic_protocol_context = ctx;
584		✗	s->handler = handler;
585		✗	}
586
587		✗	void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
588			const char *params)
589			{
590		✗	if (!ff_srtp_set_crypto(&s->srtp, suite, params))
591		✗	s->srtp_enabled = 1;
592		✗	}
593
594		✗	static int rtp_set_prft(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) {
595			int64_t rtcp_time, delta_time;
596			int32_t delta_timestamp;
597
598			AVProducerReferenceTime *prft =
599		✗	(AVProducerReferenceTime *) av_packet_new_side_data(
600			pkt, AV_PKT_DATA_PRFT, sizeof(AVProducerReferenceTime));
601		✗	if (!prft)
602		✗	return AVERROR(ENOMEM);
603
604		✗	rtcp_time = ff_parse_ntp_time(s->last_sr.ntp_timestamp) - NTP_OFFSET_US;
605			/* Cast to int32_t to handle timestamp wraparound correctly */
606		✗	delta_timestamp = (int32_t)(timestamp - s->last_sr.rtp_timestamp);
607		✗	delta_time = av_rescale_q(delta_timestamp, s->st->time_base, AV_TIME_BASE_Q);
608
609		✗	prft->wallclock = rtcp_time + delta_time;
610		✗	prft->flags = 24;
611		✗	return 0;
612			}
613
614		✗	static int rtp_add_sr_sidedata(RTPDemuxContext *s, AVPacket *pkt) {
615			AVRTCPSenderReport *sr =
616		✗	(AVRTCPSenderReport *) av_packet_new_side_data(
617			pkt, AV_PKT_DATA_RTCP_SR, sizeof(AVRTCPSenderReport));
618		✗	if (!sr)
619		✗	return AVERROR(ENOMEM);
620
621		✗	memcpy(sr, &s->last_sr, sizeof(AVRTCPSenderReport));
622		✗	s->pending_sr = 0;
623		✗	return 0;
624			}
625
626			/**
627			* This was the second switch in rtp_parse packet.
628			* Normalizes time, if required, sets stream_index, etc.
629			*/
630		✗	static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
631			{
632		✗	if (s->pending_sr) {
633		✗	int ret = rtp_add_sr_sidedata(s, pkt);
634		✗	if (ret < 0)
635		✗	av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to add SR sidedata\n");
636			}
637
638		✗	if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
639		✗	return; /* Timestamp already set by depacketizer */
640		✗	if (timestamp == RTP_NOTS_VALUE)
641		✗	return;
642
643		✗	if (s->last_sr.ntp_timestamp != AV_NOPTS_VALUE) {
644		✗	if (rtp_set_prft(s, pkt, timestamp) < 0) {
645		✗	av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to set prft");
646			}
647			}
648
649		✗	if (s->last_sr.ntp_timestamp != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
650			int64_t addend;
651			int32_t delta_timestamp;
652
653			/* compute pts from timestamp with received ntp_time */
654			/* Cast to int32_t to handle timestamp wraparound correctly */
655		✗	delta_timestamp = (int32_t)(timestamp - s->last_sr.rtp_timestamp);
656			/* convert to the PTS timebase */
657		✗	addend = av_rescale(s->last_sr.ntp_timestamp - s->first_rtcp_ntp_time,
658		✗	s->st->time_base.den,
659		✗	(uint64_t) s->st->time_base.num << 32);
660		✗	pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
661			delta_timestamp;
662		✗	return;
663			}
664
665		✗	if (!s->base_timestamp)
666		✗	s->base_timestamp = timestamp;
667			/* assume that the difference is INT32_MIN < x < INT32_MAX,
668			* but allow the first timestamp to exceed INT32_MAX */
669		✗	if (!s->timestamp)
670		✗	s->unwrapped_timestamp += timestamp;
671			else
672		✗	s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
673		✗	s->timestamp = timestamp;
674		✗	pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
675		✗	s->base_timestamp;
676			}
677
678		✗	static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
679			const uint8_t *buf, int len)
680			{
681			unsigned int ssrc;
682		✗	int payload_type, seq, flags = 0;
683			int ext, csrc;
684			AVStream *st;
685			uint32_t timestamp;
686		✗	int rv = 0;
687
688		✗	csrc = buf[0] & 0x0f;
689		✗	ext = buf[0] & 0x10;
690		✗	payload_type = buf[1] & 0x7f;
691		✗	if (buf[1] & 0x80)
692		✗	flags |= RTP_FLAG_MARKER;
693		✗	seq = AV_RB16(buf + 2);
694		✗	timestamp = AV_RB32(buf + 4);
695		✗	ssrc = AV_RB32(buf + 8);
696			/* store the ssrc in the RTPDemuxContext */
697		✗	s->ssrc = ssrc;
698
699			/* NOTE: we can handle only one payload type */
700		✗	if (s->payload_type != payload_type)
701		✗	return -1;
702
703		✗	st = s->st;
704			// only do something with this if all the rtp checks pass...
705		✗	if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
706		✗	av_log(s->ic, AV_LOG_ERROR,
707			"RTP: PT=%02x: bad cseq %04x expected=%04x\n",
708		✗	payload_type, seq, ((s->seq + 1) & 0xffff));
709		✗	return -1;
710			}
711
712		✗	if (buf[0] & 0x20) {
713		✗	int padding = buf[len - 1];
714		✗	if (len >= 12 + padding)
715		✗	len -= padding;
716			}
717
718		✗	s->seq = seq;
719		✗	len -= 12;
720		✗	buf += 12;
721
722		✗	len -= 4 * csrc;
723		✗	buf += 4 * csrc;
724		✗	if (len < 0)
725		✗	return AVERROR_INVALIDDATA;
726
727			/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
728		✗	if (ext) {
729		✗	if (len < 4)
730		✗	return -1;
731			/* calculate the header extension length (stored as number
732			* of 32-bit words) */
733		✗	ext = (AV_RB16(buf + 2) + 1) << 2;
734
735		✗	if (len < ext)
736		✗	return -1;
737			// skip past RTP header extension
738		✗	len -= ext;
739		✗	buf += ext;
740			}
741
742		✗	if (s->handler && s->handler->parse_packet) {
743		✗	rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
744			s->st, pkt, &timestamp, buf, len, seq,
745			flags);
746		✗	} else if (st) {
747		✗	if ((rv = av_new_packet(pkt, len)) < 0)
748		✗	return rv;
749		✗	memcpy(pkt->data, buf, len);
750		✗	pkt->stream_index = st->index;
751			} else {
752		✗	return AVERROR(EINVAL);
753			}
754
755			// now perform timestamp things....
756		✗	finalize_packet(s, pkt, timestamp);
757
758		✗	return rv;
759			}
760
761		✗	void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
762			{
763		✗	while (s->queue) {
764		✗	RTPPacket *next = s->queue->next;
765		✗	av_freep(&s->queue->buf);
766		✗	av_freep(&s->queue);
767		✗	s->queue = next;
768			}
769		✗	s->seq = 0;
770		✗	s->queue_len = 0;
771		✗	s->prev_ret = 0;
772		✗	}
773
774		✗	static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
775			{
776		✗	uint16_t seq = AV_RB16(buf + 2);
777		✗	RTPPacket **cur = &s->queue, *packet;
778
779			/* Find the correct place in the queue to insert the packet */
780		✗	while (*cur) {
781		✗	int16_t diff = seq - (*cur)->seq;
782		✗	if (diff < 0)
783		✗	break;
784		✗	cur = &(*cur)->next;
785			}
786
787		✗	packet = av_mallocz(sizeof(*packet));
788		✗	if (!packet)
789		✗	return AVERROR(ENOMEM);
790		✗	packet->recvtime = av_gettime_relative();
791		✗	packet->seq = seq;
792		✗	packet->len = len;
793		✗	packet->buf = buf;
794		✗	packet->next = *cur;
795		✗	*cur = packet;
796		✗	s->queue_len++;
797
798		✗	return 0;
799			}
800
801		✗	static int has_next_packet(RTPDemuxContext *s)
802			{
803		✗	return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
804			}
805
806		✗	int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
807			{
808		✗	return s->queue ? s->queue->recvtime : 0;
809			}
810
811		✗	static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
812			{
813			int rv;
814			RTPPacket *next;
815
816		✗	if (s->queue_len <= 0)
817		✗	return -1;
818
819		✗	if (!has_next_packet(s)) {
820		✗	int pkt_missed = s->queue->seq - s->seq - 1;
821
822		✗	if (pkt_missed < 0)
823		✗	pkt_missed += UINT16_MAX;
824		✗	av_log(s->ic, AV_LOG_WARNING,
825			"RTP: missed %d packets\n", pkt_missed);
826			}
827
828			/* Parse the first packet in the queue, and dequeue it */
829		✗	rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
830		✗	next = s->queue->next;
831		✗	av_freep(&s->queue->buf);
832		✗	av_freep(&s->queue);
833		✗	s->queue = next;
834		✗	s->queue_len--;
835		✗	return rv;
836			}
837
838		✗	static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
839			uint8_t **bufptr, int len)
840			{
841		✗	uint8_t *buf = bufptr ? *bufptr : NULL;
842		✗	int flags = 0;
843			uint32_t timestamp;
844		✗	int rv = 0;
845
846		✗	if (!buf) {
847			/* If parsing of the previous packet actually returned 0 or an error,
848			* there's nothing more to be parsed from that packet, but we may have
849			* indicated that we can return the next enqueued packet. */
850		✗	if (s->prev_ret <= 0)
851		✗	return rtp_parse_queued_packet(s, pkt);
852			/* return the next packets, if any */
853		✗	if (s->handler && s->handler->parse_packet) {
854			/* timestamp should be overwritten by parse_packet, if not,
855			* the packet is left with pts == AV_NOPTS_VALUE */
856		✗	timestamp = RTP_NOTS_VALUE;
857		✗	rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
858			s->st, pkt, &timestamp, NULL, 0, 0,
859			flags);
860		✗	finalize_packet(s, pkt, timestamp);
861		✗	return rv;
862			}
863			}
864
865		✗	if (len < 12)
866		✗	return -1;
867
868		✗	if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
869		✗	return -1;
870		✗	if (RTP_PT_IS_RTCP(buf[1])) {
871		✗	return rtcp_parse_packet(s, buf, len);
872			}
873
874		✗	if (s->st) {
875		✗	int64_t received = av_gettime_relative();
876		✗	uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
877		✗	s->st->time_base);
878		✗	timestamp = AV_RB32(buf + 4);
879			// Calculate the jitter immediately, before queueing the packet
880			// into the reordering queue.
881		✗	rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
882			}
883
884		✗	if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
885			/* First packet, or no reordering */
886		✗	return rtp_parse_packet_internal(s, pkt, buf, len);
887			} else {
888		✗	uint16_t seq = AV_RB16(buf + 2);
889		✗	int16_t diff = seq - s->seq;
890		✗	if (diff < 0) {
891			/* Packet older than the previously emitted one, drop */
892		✗	av_log(s->ic, AV_LOG_WARNING,
893			"RTP: dropping old packet received too late\n");
894		✗	return -1;
895		✗	} else if (diff <= 1) {
896			/* Correct packet */
897		✗	rv = rtp_parse_packet_internal(s, pkt, buf, len);
898		✗	return rv;
899			} else {
900			/* Still missing some packet, enqueue this one. */
901		✗	rv = enqueue_packet(s, buf, len);
902		✗	if (rv < 0)
903		✗	return rv;
904		✗	*bufptr = NULL;
905			/* Return the first enqueued packet if the queue is full,
906			* even if we're missing something */
907		✗	if (s->queue_len >= s->queue_size) {
908		✗	av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
909		✗	return rtp_parse_queued_packet(s, pkt);
910			}
911		✗	return -1;
912			}
913			}
914			}
915
916			/**
917			* Parse an RTP or RTCP packet directly sent as a buffer.
918			* @param s RTP parse context.
919			* @param pkt returned packet
920			* @param bufptr pointer to the input buffer or NULL to read the next packets
921			* @param len buffer len
922			* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
923			* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
924			*/
925		✗	int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
926			uint8_t **bufptr, int len)
927			{
928			int rv;
929		✗	if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
930		✗	return -1;
931		✗	rv = rtp_parse_one_packet(s, pkt, bufptr, len);
932		✗	s->prev_ret = rv;
933		✗	while (rv < 0 && has_next_packet(s))
934		✗	rv = rtp_parse_queued_packet(s, pkt);
935		✗	return rv ? rv : has_next_packet(s);
936			}
937
938		✗	void ff_rtp_parse_close(RTPDemuxContext *s)
939			{
940		✗	ff_rtp_reset_packet_queue(s);
941		✗	ff_srtp_free(&s->srtp);
942		✗	av_free(s);
943		✗	}
944
945		✗	int ff_parse_fmtp(AVFormatContext *s,
946			AVStream *stream, PayloadContext *data, const char *p,
947			int (*parse_fmtp)(AVFormatContext *s,
948			AVStream *stream,
949			PayloadContext *data,
950			const char *attr, const char *value))
951			{
952			char attr[256];
953			char *value;
954			int res;
955		✗	int value_size = strlen(p) + 1;
956
957		✗	if (!(value = av_malloc(value_size))) {
958		✗	av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
959		✗	return AVERROR(ENOMEM);
960			}
961
962			// remove protocol identifier
963		✗	while (*p && *p == ' ')
964		✗	p++; // strip spaces
965		✗	while (*p && *p != ' ')
966		✗	p++; // eat protocol identifier
967		✗	while (*p && *p == ' ')
968		✗	p++; // strip trailing spaces
969
970		✗	while (ff_rtsp_next_attr_and_value(&p,
971			attr, sizeof(attr),
972			value, value_size)) {
973		✗	res = parse_fmtp(s, stream, data, attr, value);
974		✗	if (res < 0 && res != AVERROR_PATCHWELCOME) {
975		✗	av_free(value);
976		✗	return res;
977			}
978			}
979		✗	av_free(value);
980		✗	return 0;
981			}
982
983		✗	int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
984			{
985			int ret;
986		✗	av_packet_unref(pkt);
987
988		✗	pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
989		✗	pkt->stream_index = stream_idx;
990		✗	*dyn_buf = NULL;
991		✗	if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
992		✗	av_freep(&pkt->data);
993		✗	return ret;
994			}
995		✗	return pkt->size;
996			}
997