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 opus_dynamic_handler = {
65			.enc_name = "opus",
66			.codec_type = AVMEDIA_TYPE_AUDIO,
67			.codec_id = AV_CODEC_ID_OPUS,
68			};
69
70			static const RTPDynamicProtocolHandler t140_dynamic_handler = { /* RFC 4103 */
71			.enc_name = "t140",
72			.codec_type = AVMEDIA_TYPE_SUBTITLE,
73			.codec_id = AV_CODEC_ID_TEXT,
74			};
75
76			extern const RTPDynamicProtocolHandler ff_rdt_video_handler;
77			extern const RTPDynamicProtocolHandler ff_rdt_audio_handler;
78			extern const RTPDynamicProtocolHandler ff_rdt_live_video_handler;
79			extern const RTPDynamicProtocolHandler ff_rdt_live_audio_handler;
80
81			static const RTPDynamicProtocolHandler *const rtp_dynamic_protocol_handler_list[] = {
82			/* rtp */
83			&ff_ac3_dynamic_handler,
84			&ff_amr_nb_dynamic_handler,
85			&ff_amr_wb_dynamic_handler,
86			&ff_av1_dynamic_handler,
87			&ff_dv_dynamic_handler,
88			&ff_g726_16_dynamic_handler,
89			&ff_g726_24_dynamic_handler,
90			&ff_g726_32_dynamic_handler,
91			&ff_g726_40_dynamic_handler,
92			&ff_g726le_16_dynamic_handler,
93			&ff_g726le_24_dynamic_handler,
94			&ff_g726le_32_dynamic_handler,
95			&ff_g726le_40_dynamic_handler,
96			&ff_h261_dynamic_handler,
97			&ff_h263_1998_dynamic_handler,
98			&ff_h263_2000_dynamic_handler,
99			&ff_h263_rfc2190_dynamic_handler,
100			&ff_h264_dynamic_handler,
101			&ff_hevc_dynamic_handler,
102			&ff_ilbc_dynamic_handler,
103			&ff_jpeg_dynamic_handler,
104			&ff_mp4a_latm_dynamic_handler,
105			&ff_mp4v_es_dynamic_handler,
106			&ff_mpeg_audio_dynamic_handler,
107			&ff_mpeg_audio_robust_dynamic_handler,
108			&ff_mpeg_video_dynamic_handler,
109			&ff_mpeg4_generic_dynamic_handler,
110			&ff_mpegts_dynamic_handler,
111			&ff_ms_rtp_asf_pfa_handler,
112			&ff_ms_rtp_asf_pfv_handler,
113			&ff_qcelp_dynamic_handler,
114			&ff_qdm2_dynamic_handler,
115			&ff_qt_rtp_aud_handler,
116			&ff_qt_rtp_vid_handler,
117			&ff_quicktime_rtp_aud_handler,
118			&ff_quicktime_rtp_vid_handler,
119			&ff_rfc4175_rtp_handler,
120			&ff_svq3_dynamic_handler,
121			&ff_theora_dynamic_handler,
122			&ff_vc2hq_dynamic_handler,
123			&ff_vorbis_dynamic_handler,
124			&ff_vp8_dynamic_handler,
125			&ff_vp9_dynamic_handler,
126			&gsm_dynamic_handler,
127			&l24_dynamic_handler,
128			&opus_dynamic_handler,
129			&realmedia_mp3_dynamic_handler,
130			&speex_dynamic_handler,
131			&t140_dynamic_handler,
132			/* rdt */
133			&ff_rdt_video_handler,
134			&ff_rdt_audio_handler,
135			&ff_rdt_live_video_handler,
136			&ff_rdt_live_audio_handler,
137			NULL,
138			};
139
140			/**
141			* Iterate over all registered rtp dynamic protocol handlers.
142			*
143			* @param opaque a pointer where libavformat will store the iteration state.
144			* Must point to NULL to start the iteration.
145			*
146			* @return the next registered rtp dynamic protocol handler
147			* or NULL when the iteration is finished
148			*/
149		✗	static const RTPDynamicProtocolHandler *rtp_handler_iterate(void **opaque)
150			{
151		✗	uintptr_t i = (uintptr_t)*opaque;
152		✗	const RTPDynamicProtocolHandler *r = rtp_dynamic_protocol_handler_list[i];
153
154		✗	if (r)
155		✗	*opaque = (void*)(i + 1);
156
157		✗	return r;
158			}
159
160		✗	const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
161			enum AVMediaType codec_type)
162			{
163		✗	void *i = 0;
164			const RTPDynamicProtocolHandler *handler;
165		✗	while (handler = rtp_handler_iterate(&i)) {
166		✗	if (handler->enc_name &&
167		✗	!av_strcasecmp(name, handler->enc_name) &&
168		✗	codec_type == handler->codec_type)
169		✗	return handler;
170			}
171		✗	return NULL;
172			}
173
174		✗	const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
175			enum AVMediaType codec_type)
176			{
177		✗	void *i = 0;
178			const RTPDynamicProtocolHandler *handler;
179		✗	while (handler = rtp_handler_iterate(&i)) {
180		✗	if (handler->static_payload_id && handler->static_payload_id == id &&
181		✗	codec_type == handler->codec_type)
182		✗	return handler;
183			}
184		✗	return NULL;
185			}
186
187		✗	static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
188			int len)
189			{
190			int payload_len;
191		✗	while (len >= 4) {
192		✗	payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
193
194		✗	switch (buf[1]) {
195		✗	case RTCP_SR:
196		✗	if (payload_len < 20) {
197		✗	av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
198		✗	return AVERROR_INVALIDDATA;
199			}
200
201		✗	s->last_rtcp_reception_time = av_gettime_relative();
202		✗	s->last_rtcp_ntp_time = AV_RB64(buf + 8);
203		✗	s->last_rtcp_timestamp = AV_RB32(buf + 16);
204		✗	if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
205		✗	s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
206		✗	if (!s->base_timestamp)
207		✗	s->base_timestamp = s->last_rtcp_timestamp;
208		✗	s->rtcp_ts_offset = (int32_t)(s->last_rtcp_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_rtcp_ntp_time == 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_rtcp_ntp_time >> 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		✗	static int opus_write_extradata(AVCodecParameters *codecpar)
535			{
536			uint8_t *bs;
537			int ret;
538
539			/* This function writes an extradata with a channel mapping family of 0.
540			* This mapping family only supports mono and stereo layouts. And RFC7587
541			* specifies that the number of channels in the SDP must be 2.
542			*/
543		✗	if (codecpar->ch_layout.nb_channels > 2) {
544		✗	return AVERROR_INVALIDDATA;
545			}
546
547		✗	ret = ff_alloc_extradata(codecpar, 19);
548		✗	if (ret < 0)
549		✗	return ret;
550
551		✗	bs = (uint8_t *)codecpar->extradata;
552
553			/* Opus magic */
554		✗	bytestream_put_buffer(&bs, "OpusHead", 8);
555			/* Version */
556		✗	bytestream_put_byte (&bs, 0x1);
557			/* Channel count */
558		✗	bytestream_put_byte (&bs, codecpar->ch_layout.nb_channels);
559			/* Pre skip */
560		✗	bytestream_put_le16 (&bs, 0);
561			/* Input sample rate */
562		✗	bytestream_put_le32 (&bs, 48000);
563			/* Output gain */
564		✗	bytestream_put_le16 (&bs, 0x0);
565			/* Mapping family */
566		✗	bytestream_put_byte (&bs, 0x0);
567
568		✗	return 0;
569			}
570
571			/**
572			* open a new RTP parse context for stream 'st'. 'st' can be NULL for
573			* MPEG-2 TS streams.
574			*/
575		✗	RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
576			int payload_type, int queue_size)
577			{
578			RTPDemuxContext *s;
579			int ret;
580
581		✗	s = av_mallocz(sizeof(RTPDemuxContext));
582		✗	if (!s)
583		✗	return NULL;
584		✗	s->payload_type = payload_type;
585		✗	s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
586		✗	s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
587		✗	s->ic = s1;
588		✗	s->st = st;
589		✗	s->queue_size = queue_size;
590
591		✗	av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
592			s->queue_size);
593
594		✗	rtp_init_statistics(&s->statistics, 0);
595		✗	if (st) {
596		✗	switch (st->codecpar->codec_id) {
597		✗	case AV_CODEC_ID_ADPCM_G722:
598			/* According to RFC 3551, the stream clock rate is 8000
599			* even if the sample rate is 16000. */
600		✗	if (st->codecpar->sample_rate == 8000)
601		✗	st->codecpar->sample_rate = 16000;
602		✗	break;
603		✗	case AV_CODEC_ID_OPUS:
604		✗	ret = opus_write_extradata(st->codecpar);
605		✗	if (ret < 0) {
606		✗	av_log(s1, AV_LOG_ERROR,
607			"Error creating opus extradata: %s\n",
608		✗	av_err2str(ret));
609		✗	av_free(s);
610		✗	return NULL;
611			}
612		✗	break;
613		✗	default:
614		✗	break;
615			}
616			}
617			// needed to send back RTCP RR in RTSP sessions
618		✗	gethostname(s->hostname, sizeof(s->hostname));
619		✗	return s;
620			}
621
622		✗	void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
623			const RTPDynamicProtocolHandler *handler)
624			{
625		✗	s->dynamic_protocol_context = ctx;
626		✗	s->handler = handler;
627		✗	}
628
629		✗	void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
630			const char *params)
631			{
632		✗	if (!ff_srtp_set_crypto(&s->srtp, suite, params))
633		✗	s->srtp_enabled = 1;
634		✗	}
635
636		✗	static int rtp_set_prft(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) {
637			int64_t rtcp_time, delta_timestamp, delta_time;
638
639			AVProducerReferenceTime *prft =
640		✗	(AVProducerReferenceTime *) av_packet_new_side_data(
641			pkt, AV_PKT_DATA_PRFT, sizeof(AVProducerReferenceTime));
642		✗	if (!prft)
643		✗	return AVERROR(ENOMEM);
644
645		✗	rtcp_time = ff_parse_ntp_time(s->last_rtcp_ntp_time) - NTP_OFFSET_US;
646		✗	delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp;
647		✗	delta_time = av_rescale_q(delta_timestamp, s->st->time_base, AV_TIME_BASE_Q);
648
649		✗	prft->wallclock = rtcp_time + delta_time;
650		✗	prft->flags = 24;
651		✗	return 0;
652			}
653
654			/**
655			* This was the second switch in rtp_parse packet.
656			* Normalizes time, if required, sets stream_index, etc.
657			*/
658		✗	static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
659			{
660		✗	if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
661		✗	return; /* Timestamp already set by depacketizer */
662		✗	if (timestamp == RTP_NOTS_VALUE)
663		✗	return;
664
665		✗	if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
666		✗	if (rtp_set_prft(s, pkt, timestamp) < 0) {
667		✗	av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to set prft");
668			}
669			}
670
671		✗	if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
672			int64_t addend;
673			int delta_timestamp;
674
675			/* compute pts from timestamp with received ntp_time */
676		✗	delta_timestamp = timestamp - s->last_rtcp_timestamp;
677			/* convert to the PTS timebase */
678		✗	addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
679		✗	s->st->time_base.den,
680		✗	(uint64_t) s->st->time_base.num << 32);
681		✗	pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
682			delta_timestamp;
683		✗	return;
684			}
685
686		✗	if (!s->base_timestamp)
687		✗	s->base_timestamp = timestamp;
688			/* assume that the difference is INT32_MIN < x < INT32_MAX,
689			* but allow the first timestamp to exceed INT32_MAX */
690		✗	if (!s->timestamp)
691		✗	s->unwrapped_timestamp += timestamp;
692			else
693		✗	s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
694		✗	s->timestamp = timestamp;
695		✗	pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
696		✗	s->base_timestamp;
697			}
698
699		✗	static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
700			const uint8_t *buf, int len)
701			{
702			unsigned int ssrc;
703		✗	int payload_type, seq, flags = 0;
704			int ext, csrc;
705			AVStream *st;
706			uint32_t timestamp;
707		✗	int rv = 0;
708
709		✗	csrc = buf[0] & 0x0f;
710		✗	ext = buf[0] & 0x10;
711		✗	payload_type = buf[1] & 0x7f;
712		✗	if (buf[1] & 0x80)
713		✗	flags |= RTP_FLAG_MARKER;
714		✗	seq = AV_RB16(buf + 2);
715		✗	timestamp = AV_RB32(buf + 4);
716		✗	ssrc = AV_RB32(buf + 8);
717			/* store the ssrc in the RTPDemuxContext */
718		✗	s->ssrc = ssrc;
719
720			/* NOTE: we can handle only one payload type */
721		✗	if (s->payload_type != payload_type)
722		✗	return -1;
723
724		✗	st = s->st;
725			// only do something with this if all the rtp checks pass...
726		✗	if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
727		✗	av_log(s->ic, AV_LOG_ERROR,
728			"RTP: PT=%02x: bad cseq %04x expected=%04x\n",
729		✗	payload_type, seq, ((s->seq + 1) & 0xffff));
730		✗	return -1;
731			}
732
733		✗	if (buf[0] & 0x20) {
734		✗	int padding = buf[len - 1];
735		✗	if (len >= 12 + padding)
736		✗	len -= padding;
737			}
738
739		✗	s->seq = seq;
740		✗	len -= 12;
741		✗	buf += 12;
742
743		✗	len -= 4 * csrc;
744		✗	buf += 4 * csrc;
745		✗	if (len < 0)
746		✗	return AVERROR_INVALIDDATA;
747
748			/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
749		✗	if (ext) {
750		✗	if (len < 4)
751		✗	return -1;
752			/* calculate the header extension length (stored as number
753			* of 32-bit words) */
754		✗	ext = (AV_RB16(buf + 2) + 1) << 2;
755
756		✗	if (len < ext)
757		✗	return -1;
758			// skip past RTP header extension
759		✗	len -= ext;
760		✗	buf += ext;
761			}
762
763		✗	if (s->handler && s->handler->parse_packet) {
764		✗	rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
765			s->st, pkt, &timestamp, buf, len, seq,
766			flags);
767		✗	} else if (st) {
768		✗	if ((rv = av_new_packet(pkt, len)) < 0)
769		✗	return rv;
770		✗	memcpy(pkt->data, buf, len);
771		✗	pkt->stream_index = st->index;
772			} else {
773		✗	return AVERROR(EINVAL);
774			}
775
776			// now perform timestamp things....
777		✗	finalize_packet(s, pkt, timestamp);
778
779		✗	return rv;
780			}
781
782		✗	void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
783			{
784		✗	while (s->queue) {
785		✗	RTPPacket *next = s->queue->next;
786		✗	av_freep(&s->queue->buf);
787		✗	av_freep(&s->queue);
788		✗	s->queue = next;
789			}
790		✗	s->seq = 0;
791		✗	s->queue_len = 0;
792		✗	s->prev_ret = 0;
793		✗	}
794
795		✗	static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
796			{
797		✗	uint16_t seq = AV_RB16(buf + 2);
798		✗	RTPPacket **cur = &s->queue, *packet;
799
800			/* Find the correct place in the queue to insert the packet */
801		✗	while (*cur) {
802		✗	int16_t diff = seq - (*cur)->seq;
803		✗	if (diff < 0)
804		✗	break;
805		✗	cur = &(*cur)->next;
806			}
807
808		✗	packet = av_mallocz(sizeof(*packet));
809		✗	if (!packet)
810		✗	return AVERROR(ENOMEM);
811		✗	packet->recvtime = av_gettime_relative();
812		✗	packet->seq = seq;
813		✗	packet->len = len;
814		✗	packet->buf = buf;
815		✗	packet->next = *cur;
816		✗	*cur = packet;
817		✗	s->queue_len++;
818
819		✗	return 0;
820			}
821
822		✗	static int has_next_packet(RTPDemuxContext *s)
823			{
824		✗	return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
825			}
826
827		✗	int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
828			{
829		✗	return s->queue ? s->queue->recvtime : 0;
830			}
831
832		✗	static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
833			{
834			int rv;
835			RTPPacket *next;
836
837		✗	if (s->queue_len <= 0)
838		✗	return -1;
839
840		✗	if (!has_next_packet(s)) {
841		✗	int pkt_missed = s->queue->seq - s->seq - 1;
842
843		✗	if (pkt_missed < 0)
844		✗	pkt_missed += UINT16_MAX;
845		✗	av_log(s->ic, AV_LOG_WARNING,
846			"RTP: missed %d packets\n", pkt_missed);
847			}
848
849			/* Parse the first packet in the queue, and dequeue it */
850		✗	rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
851		✗	next = s->queue->next;
852		✗	av_freep(&s->queue->buf);
853		✗	av_freep(&s->queue);
854		✗	s->queue = next;
855		✗	s->queue_len--;
856		✗	return rv;
857			}
858
859		✗	static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
860			uint8_t **bufptr, int len)
861			{
862		✗	uint8_t *buf = bufptr ? *bufptr : NULL;
863		✗	int flags = 0;
864			uint32_t timestamp;
865		✗	int rv = 0;
866
867		✗	if (!buf) {
868			/* If parsing of the previous packet actually returned 0 or an error,
869			* there's nothing more to be parsed from that packet, but we may have
870			* indicated that we can return the next enqueued packet. */
871		✗	if (s->prev_ret <= 0)
872		✗	return rtp_parse_queued_packet(s, pkt);
873			/* return the next packets, if any */
874		✗	if (s->handler && s->handler->parse_packet) {
875			/* timestamp should be overwritten by parse_packet, if not,
876			* the packet is left with pts == AV_NOPTS_VALUE */
877		✗	timestamp = RTP_NOTS_VALUE;
878		✗	rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
879			s->st, pkt, &timestamp, NULL, 0, 0,
880			flags);
881		✗	finalize_packet(s, pkt, timestamp);
882		✗	return rv;
883			}
884			}
885
886		✗	if (len < 12)
887		✗	return -1;
888
889		✗	if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
890		✗	return -1;
891		✗	if (RTP_PT_IS_RTCP(buf[1])) {
892		✗	return rtcp_parse_packet(s, buf, len);
893			}
894
895		✗	if (s->st) {
896		✗	int64_t received = av_gettime_relative();
897		✗	uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
898		✗	s->st->time_base);
899		✗	timestamp = AV_RB32(buf + 4);
900			// Calculate the jitter immediately, before queueing the packet
901			// into the reordering queue.
902		✗	rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
903			}
904
905		✗	if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
906			/* First packet, or no reordering */
907		✗	return rtp_parse_packet_internal(s, pkt, buf, len);
908			} else {
909		✗	uint16_t seq = AV_RB16(buf + 2);
910		✗	int16_t diff = seq - s->seq;
911		✗	if (diff < 0) {
912			/* Packet older than the previously emitted one, drop */
913		✗	av_log(s->ic, AV_LOG_WARNING,
914			"RTP: dropping old packet received too late\n");
915		✗	return -1;
916		✗	} else if (diff <= 1) {
917			/* Correct packet */
918		✗	rv = rtp_parse_packet_internal(s, pkt, buf, len);
919		✗	return rv;
920			} else {
921			/* Still missing some packet, enqueue this one. */
922		✗	rv = enqueue_packet(s, buf, len);
923		✗	if (rv < 0)
924		✗	return rv;
925		✗	*bufptr = NULL;
926			/* Return the first enqueued packet if the queue is full,
927			* even if we're missing something */
928		✗	if (s->queue_len >= s->queue_size) {
929		✗	av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
930		✗	return rtp_parse_queued_packet(s, pkt);
931			}
932		✗	return -1;
933			}
934			}
935			}
936
937			/**
938			* Parse an RTP or RTCP packet directly sent as a buffer.
939			* @param s RTP parse context.
940			* @param pkt returned packet
941			* @param bufptr pointer to the input buffer or NULL to read the next packets
942			* @param len buffer len
943			* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
944			* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
945			*/
946		✗	int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
947			uint8_t **bufptr, int len)
948			{
949			int rv;
950		✗	if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
951		✗	return -1;
952		✗	rv = rtp_parse_one_packet(s, pkt, bufptr, len);
953		✗	s->prev_ret = rv;
954		✗	while (rv < 0 && has_next_packet(s))
955		✗	rv = rtp_parse_queued_packet(s, pkt);
956		✗	return rv ? rv : has_next_packet(s);
957			}
958
959		✗	void ff_rtp_parse_close(RTPDemuxContext *s)
960			{
961		✗	ff_rtp_reset_packet_queue(s);
962		✗	ff_srtp_free(&s->srtp);
963		✗	av_free(s);
964		✗	}
965
966		✗	int ff_parse_fmtp(AVFormatContext *s,
967			AVStream *stream, PayloadContext *data, const char *p,
968			int (*parse_fmtp)(AVFormatContext *s,
969			AVStream *stream,
970			PayloadContext *data,
971			const char *attr, const char *value))
972			{
973			char attr[256];
974			char *value;
975			int res;
976		✗	int value_size = strlen(p) + 1;
977
978		✗	if (!(value = av_malloc(value_size))) {
979		✗	av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
980		✗	return AVERROR(ENOMEM);
981			}
982
983			// remove protocol identifier
984		✗	while (*p && *p == ' ')
985		✗	p++; // strip spaces
986		✗	while (*p && *p != ' ')
987		✗	p++; // eat protocol identifier
988		✗	while (*p && *p == ' ')
989		✗	p++; // strip trailing spaces
990
991		✗	while (ff_rtsp_next_attr_and_value(&p,
992			attr, sizeof(attr),
993			value, value_size)) {
994		✗	res = parse_fmtp(s, stream, data, attr, value);
995		✗	if (res < 0 && res != AVERROR_PATCHWELCOME) {
996		✗	av_free(value);
997		✗	return res;
998			}
999			}
1000		✗	av_free(value);
1001		✗	return 0;
1002			}
1003
1004		✗	int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
1005			{
1006			int ret;
1007		✗	av_packet_unref(pkt);
1008
1009		✗	pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
1010		✗	pkt->stream_index = stream_idx;
1011		✗	*dyn_buf = NULL;
1012		✗	if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
1013		✗	av_freep(&pkt->data);
1014		✗	return ret;
1015			}
1016		✗	return pkt->size;
1017			}
1018