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_dv_dynamic_handler,
87			&ff_g726_16_dynamic_handler,
88			&ff_g726_24_dynamic_handler,
89			&ff_g726_32_dynamic_handler,
90			&ff_g726_40_dynamic_handler,
91			&ff_g726le_16_dynamic_handler,
92			&ff_g726le_24_dynamic_handler,
93			&ff_g726le_32_dynamic_handler,
94			&ff_g726le_40_dynamic_handler,
95			&ff_h261_dynamic_handler,
96			&ff_h263_1998_dynamic_handler,
97			&ff_h263_2000_dynamic_handler,
98			&ff_h263_rfc2190_dynamic_handler,
99			&ff_h264_dynamic_handler,
100			&ff_hevc_dynamic_handler,
101			&ff_ilbc_dynamic_handler,
102			&ff_jpeg_dynamic_handler,
103			&ff_mp4a_latm_dynamic_handler,
104			&ff_mp4v_es_dynamic_handler,
105			&ff_mpeg_audio_dynamic_handler,
106			&ff_mpeg_audio_robust_dynamic_handler,
107			&ff_mpeg_video_dynamic_handler,
108			&ff_mpeg4_generic_dynamic_handler,
109			&ff_mpegts_dynamic_handler,
110			&ff_ms_rtp_asf_pfa_handler,
111			&ff_ms_rtp_asf_pfv_handler,
112			&ff_qcelp_dynamic_handler,
113			&ff_qdm2_dynamic_handler,
114			&ff_qt_rtp_aud_handler,
115			&ff_qt_rtp_vid_handler,
116			&ff_quicktime_rtp_aud_handler,
117			&ff_quicktime_rtp_vid_handler,
118			&ff_rfc4175_rtp_handler,
119			&ff_svq3_dynamic_handler,
120			&ff_theora_dynamic_handler,
121			&ff_vc2hq_dynamic_handler,
122			&ff_vorbis_dynamic_handler,
123			&ff_vp8_dynamic_handler,
124			&ff_vp9_dynamic_handler,
125			&gsm_dynamic_handler,
126			&l24_dynamic_handler,
127			&opus_dynamic_handler,
128			&realmedia_mp3_dynamic_handler,
129			&speex_dynamic_handler,
130			&t140_dynamic_handler,
131			/* rdt */
132			&ff_rdt_video_handler,
133			&ff_rdt_audio_handler,
134			&ff_rdt_live_video_handler,
135			&ff_rdt_live_audio_handler,
136			NULL,
137			};
138
139			/**
140			* Iterate over all registered rtp dynamic protocol handlers.
141			*
142			* @param opaque a pointer where libavformat will store the iteration state.
143			* Must point to NULL to start the iteration.
144			*
145			* @return the next registered rtp dynamic protocol handler
146			* or NULL when the iteration is finished
147			*/
148		✗	static const RTPDynamicProtocolHandler *rtp_handler_iterate(void **opaque)
149			{
150		✗	uintptr_t i = (uintptr_t)*opaque;
151		✗	const RTPDynamicProtocolHandler *r = rtp_dynamic_protocol_handler_list[i];
152
153		✗	if (r)
154		✗	*opaque = (void*)(i + 1);
155
156		✗	return r;
157			}
158
159		✗	const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
160			enum AVMediaType codec_type)
161			{
162		✗	void *i = 0;
163			const RTPDynamicProtocolHandler *handler;
164		✗	while (handler = rtp_handler_iterate(&i)) {
165		✗	if (handler->enc_name &&
166		✗	!av_strcasecmp(name, handler->enc_name) &&
167		✗	codec_type == handler->codec_type)
168		✗	return handler;
169			}
170		✗	return NULL;
171			}
172
173		✗	const RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
174			enum AVMediaType codec_type)
175			{
176		✗	void *i = 0;
177			const RTPDynamicProtocolHandler *handler;
178		✗	while (handler = rtp_handler_iterate(&i)) {
179		✗	if (handler->static_payload_id && handler->static_payload_id == id &&
180		✗	codec_type == handler->codec_type)
181		✗	return handler;
182			}
183		✗	return NULL;
184			}
185
186		✗	static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf,
187			int len)
188			{
189			int payload_len;
190		✗	while (len >= 4) {
191		✗	payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
192
193		✗	switch (buf[1]) {
194		✗	case RTCP_SR:
195		✗	if (payload_len < 20) {
196		✗	av_log(s->ic, AV_LOG_ERROR, "Invalid RTCP SR packet length\n");
197		✗	return AVERROR_INVALIDDATA;
198			}
199
200		✗	s->last_rtcp_reception_time = av_gettime_relative();
201		✗	s->last_rtcp_ntp_time = AV_RB64(buf + 8);
202		✗	s->last_rtcp_timestamp = AV_RB32(buf + 16);
203		✗	if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
204		✗	s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
205		✗	if (!s->base_timestamp)
206		✗	s->base_timestamp = s->last_rtcp_timestamp;
207		✗	s->rtcp_ts_offset = (int32_t)(s->last_rtcp_timestamp - s->base_timestamp);
208			}
209
210		✗	break;
211		✗	case RTCP_BYE:
212		✗	return -RTCP_BYE;
213			}
214
215		✗	buf += payload_len;
216		✗	len -= payload_len;
217			}
218		✗	return -1;
219			}
220
221			#define RTP_SEQ_MOD (1 << 16)
222
223		✗	static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence)
224			{
225		✗	memset(s, 0, sizeof(RTPStatistics));
226		✗	s->max_seq = base_sequence;
227		✗	s->probation = 1;
228		✗	}
229
230			/*
231			* Called whenever there is a large jump in sequence numbers,
232			* or when they get out of probation...
233			*/
234		✗	static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
235			{
236		✗	s->max_seq = seq;
237		✗	s->cycles = 0;
238		✗	s->base_seq = seq - 1;
239		✗	s->bad_seq = RTP_SEQ_MOD + 1;
240		✗	s->received = 0;
241		✗	s->expected_prior = 0;
242		✗	s->received_prior = 0;
243		✗	s->jitter = 0;
244		✗	s->transit = 0;
245		✗	}
246
247			/* Returns 1 if we should handle this packet. */
248		✗	static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
249			{
250		✗	uint16_t udelta = seq - s->max_seq;
251		✗	const int MAX_DROPOUT = 3000;
252		✗	const int MAX_MISORDER = 100;
253		✗	const int MIN_SEQUENTIAL = 2;
254
255			/* source not valid until MIN_SEQUENTIAL packets with sequence
256			* seq. numbers have been received */
257		✗	if (s->probation) {
258		✗	if (seq == s->max_seq + 1) {
259		✗	s->probation--;
260		✗	s->max_seq = seq;
261		✗	if (s->probation == 0) {
262		✗	rtp_init_sequence(s, seq);
263		✗	s->received++;
264		✗	return 1;
265			}
266			} else {
267		✗	s->probation = MIN_SEQUENTIAL - 1;
268		✗	s->max_seq = seq;
269			}
270		✗	} else if (udelta < MAX_DROPOUT) {
271			// in order, with permissible gap
272		✗	if (seq < s->max_seq) {
273			// sequence number wrapped; count another 64k cycles
274		✗	s->cycles += RTP_SEQ_MOD;
275			}
276		✗	s->max_seq = seq;
277		✗	} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
278			// sequence made a large jump...
279		✗	if (seq == s->bad_seq) {
280			/* two sequential packets -- assume that the other side
281			* restarted without telling us; just resync. */
282		✗	rtp_init_sequence(s, seq);
283			} else {
284		✗	s->bad_seq = (seq + 1) & (RTP_SEQ_MOD - 1);
285		✗	return 0;
286			}
287			} else {
288			// duplicate or reordered packet...
289			}
290		✗	s->received++;
291		✗	return 1;
292			}
293
294		✗	static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp,
295			uint32_t arrival_timestamp)
296			{
297			// Most of this is pretty straight from RFC 3550 appendix A.8
298		✗	uint32_t transit = arrival_timestamp - sent_timestamp;
299		✗	uint32_t prev_transit = s->transit;
300		✗	int32_t d = transit - prev_transit;
301			// Doing the FFABS() call directly on the "transit - prev_transit"
302			// expression doesn't work, since it's an unsigned expression. Doing the
303			// transit calculation in unsigned is desired though, since it most
304			// probably will need to wrap around.
305		✗	d = FFABS(d);
306		✗	s->transit = transit;
307		✗	if (!prev_transit)
308		✗	return;
309		✗	s->jitter += d - (int32_t) ((s->jitter + 8) >> 4);
310			}
311
312		✗	int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, URLContext *fd,
313			AVIOContext *avio, int count)
314			{
315			AVIOContext *pb;
316			uint8_t *buf;
317			int len;
318			int rtcp_bytes;
319		✗	RTPStatistics *stats = &s->statistics;
320			uint32_t lost;
321			uint32_t extended_max;
322			uint32_t expected_interval;
323			uint32_t received_interval;
324			int32_t lost_interval;
325			uint32_t expected;
326			uint32_t fraction;
327
328		✗	if ((!fd && !avio) || (count < 1))
329		✗	return -1;
330
331			/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
332			/* XXX: MPEG pts hardcoded. RTCP send every 0.5 seconds */
333		✗	s->octet_count += count;
334		✗	rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
335			RTCP_TX_RATIO_DEN;
336		✗	rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
337		✗	if (rtcp_bytes < 28)
338		✗	return -1;
339		✗	s->last_octet_count = s->octet_count;
340
341		✗	if (!fd)
342		✗	pb = avio;
343		✗	else if (avio_open_dyn_buf(&pb) < 0)
344		✗	return -1;
345
346			// Receiver Report
347		✗	avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
348		✗	avio_w8(pb, RTCP_RR);
349		✗	avio_wb16(pb, 7); /* length in words - 1 */
350			// our own SSRC: we use the server's SSRC + 1 to avoid conflicts
351		✗	avio_wb32(pb, s->ssrc + 1);
352		✗	avio_wb32(pb, s->ssrc); // server SSRC
353			// some placeholders we should really fill...
354			// RFC 1889/p64
355		✗	extended_max = stats->cycles + stats->max_seq;
356		✗	expected = extended_max - stats->base_seq;
357		✗	lost = expected - stats->received;
358		✗	lost = FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
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_rtcp_ntp_time == 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_rtcp_ntp_time >> 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			int av_unused 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		✗	static int opus_write_extradata(AVCodecParameters *codecpar)
534			{
535			uint8_t *bs;
536			int ret;
537
538			/* This function writes an extradata with a channel mapping family of 0.
539			* This mapping family only supports mono and stereo layouts. And RFC7587
540			* specifies that the number of channels in the SDP must be 2.
541			*/
542		✗	if (codecpar->ch_layout.nb_channels > 2) {
543		✗	return AVERROR_INVALIDDATA;
544			}
545
546		✗	ret = ff_alloc_extradata(codecpar, 19);
547		✗	if (ret < 0)
548		✗	return ret;
549
550		✗	bs = (uint8_t *)codecpar->extradata;
551
552			/* Opus magic */
553		✗	bytestream_put_buffer(&bs, "OpusHead", 8);
554			/* Version */
555		✗	bytestream_put_byte (&bs, 0x1);
556			/* Channel count */
557		✗	bytestream_put_byte (&bs, codecpar->ch_layout.nb_channels);
558			/* Pre skip */
559		✗	bytestream_put_le16 (&bs, 0);
560			/* Input sample rate */
561		✗	bytestream_put_le32 (&bs, 48000);
562			/* Output gain */
563		✗	bytestream_put_le16 (&bs, 0x0);
564			/* Mapping family */
565		✗	bytestream_put_byte (&bs, 0x0);
566
567		✗	return 0;
568			}
569
570			/**
571			* open a new RTP parse context for stream 'st'. 'st' can be NULL for
572			* MPEG-2 TS streams.
573			*/
574		✗	RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st,
575			int payload_type, int queue_size)
576			{
577			RTPDemuxContext *s;
578			int ret;
579
580		✗	s = av_mallocz(sizeof(RTPDemuxContext));
581		✗	if (!s)
582		✗	return NULL;
583		✗	s->payload_type = payload_type;
584		✗	s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
585		✗	s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
586		✗	s->ic = s1;
587		✗	s->st = st;
588		✗	s->queue_size = queue_size;
589
590		✗	av_log(s->ic, AV_LOG_VERBOSE, "setting jitter buffer size to %d\n",
591			s->queue_size);
592
593		✗	rtp_init_statistics(&s->statistics, 0);
594		✗	if (st) {
595		✗	switch (st->codecpar->codec_id) {
596		✗	case AV_CODEC_ID_ADPCM_G722:
597			/* According to RFC 3551, the stream clock rate is 8000
598			* even if the sample rate is 16000. */
599		✗	if (st->codecpar->sample_rate == 8000)
600		✗	st->codecpar->sample_rate = 16000;
601		✗	break;
602		✗	case AV_CODEC_ID_OPUS:
603		✗	ret = opus_write_extradata(st->codecpar);
604		✗	if (ret < 0) {
605		✗	av_log(s1, AV_LOG_ERROR,
606			"Error creating opus extradata: %s\n",
607		✗	av_err2str(ret));
608		✗	av_free(s);
609		✗	return NULL;
610			}
611		✗	break;
612		✗	default:
613		✗	break;
614			}
615			}
616			// needed to send back RTCP RR in RTSP sessions
617		✗	gethostname(s->hostname, sizeof(s->hostname));
618		✗	return s;
619			}
620
621		✗	void ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
622			const RTPDynamicProtocolHandler *handler)
623			{
624		✗	s->dynamic_protocol_context = ctx;
625		✗	s->handler = handler;
626		✗	}
627
628		✗	void ff_rtp_parse_set_crypto(RTPDemuxContext *s, const char *suite,
629			const char *params)
630			{
631		✗	if (!ff_srtp_set_crypto(&s->srtp, suite, params))
632		✗	s->srtp_enabled = 1;
633		✗	}
634
635		✗	static int rtp_set_prft(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) {
636			int64_t rtcp_time, delta_timestamp, delta_time;
637
638			AVProducerReferenceTime *prft =
639		✗	(AVProducerReferenceTime *) av_packet_new_side_data(
640			pkt, AV_PKT_DATA_PRFT, sizeof(AVProducerReferenceTime));
641		✗	if (!prft)
642		✗	return AVERROR(ENOMEM);
643
644		✗	rtcp_time = ff_parse_ntp_time(s->last_rtcp_ntp_time) - NTP_OFFSET_US;
645		✗	delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp;
646		✗	delta_time = av_rescale_q(delta_timestamp, s->st->time_base, AV_TIME_BASE_Q);
647
648		✗	prft->wallclock = rtcp_time + delta_time;
649		✗	prft->flags = 24;
650		✗	return 0;
651			}
652
653			/**
654			* This was the second switch in rtp_parse packet.
655			* Normalizes time, if required, sets stream_index, etc.
656			*/
657		✗	static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
658			{
659		✗	if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
660		✗	return; /* Timestamp already set by depacketizer */
661		✗	if (timestamp == RTP_NOTS_VALUE)
662		✗	return;
663
664		✗	if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
665		✗	if (rtp_set_prft(s, pkt, timestamp) < 0) {
666		✗	av_log(s->ic, AV_LOG_WARNING, "rtpdec: failed to set prft");
667			}
668			}
669
670		✗	if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
671			int64_t addend;
672			int delta_timestamp;
673
674			/* compute pts from timestamp with received ntp_time */
675		✗	delta_timestamp = timestamp - s->last_rtcp_timestamp;
676			/* convert to the PTS timebase */
677		✗	addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time,
678		✗	s->st->time_base.den,
679		✗	(uint64_t) s->st->time_base.num << 32);
680		✗	pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
681			delta_timestamp;
682		✗	return;
683			}
684
685		✗	if (!s->base_timestamp)
686		✗	s->base_timestamp = timestamp;
687			/* assume that the difference is INT32_MIN < x < INT32_MAX,
688			* but allow the first timestamp to exceed INT32_MAX */
689		✗	if (!s->timestamp)
690		✗	s->unwrapped_timestamp += timestamp;
691			else
692		✗	s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
693		✗	s->timestamp = timestamp;
694		✗	pkt->pts = s->unwrapped_timestamp + s->range_start_offset -
695		✗	s->base_timestamp;
696			}
697
698		✗	static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
699			const uint8_t *buf, int len)
700			{
701			unsigned int ssrc;
702		✗	int payload_type, seq, flags = 0;
703			int ext, csrc;
704			AVStream *st;
705			uint32_t timestamp;
706		✗	int rv = 0;
707
708		✗	csrc = buf[0] & 0x0f;
709		✗	ext = buf[0] & 0x10;
710		✗	payload_type = buf[1] & 0x7f;
711		✗	if (buf[1] & 0x80)
712		✗	flags |= RTP_FLAG_MARKER;
713		✗	seq = AV_RB16(buf + 2);
714		✗	timestamp = AV_RB32(buf + 4);
715		✗	ssrc = AV_RB32(buf + 8);
716			/* store the ssrc in the RTPDemuxContext */
717		✗	s->ssrc = ssrc;
718
719			/* NOTE: we can handle only one payload type */
720		✗	if (s->payload_type != payload_type)
721		✗	return -1;
722
723		✗	st = s->st;
724			// only do something with this if all the rtp checks pass...
725		✗	if (!rtp_valid_packet_in_sequence(&s->statistics, seq)) {
726		✗	av_log(s->ic, AV_LOG_ERROR,
727			"RTP: PT=%02x: bad cseq %04x expected=%04x\n",
728		✗	payload_type, seq, ((s->seq + 1) & 0xffff));
729		✗	return -1;
730			}
731
732		✗	if (buf[0] & 0x20) {
733		✗	int padding = buf[len - 1];
734		✗	if (len >= 12 + padding)
735		✗	len -= padding;
736			}
737
738		✗	s->seq = seq;
739		✗	len -= 12;
740		✗	buf += 12;
741
742		✗	len -= 4 * csrc;
743		✗	buf += 4 * csrc;
744		✗	if (len < 0)
745		✗	return AVERROR_INVALIDDATA;
746
747			/* RFC 3550 Section 5.3.1 RTP Header Extension handling */
748		✗	if (ext) {
749		✗	if (len < 4)
750		✗	return -1;
751			/* calculate the header extension length (stored as number
752			* of 32-bit words) */
753		✗	ext = (AV_RB16(buf + 2) + 1) << 2;
754
755		✗	if (len < ext)
756		✗	return -1;
757			// skip past RTP header extension
758		✗	len -= ext;
759		✗	buf += ext;
760			}
761
762		✗	if (s->handler && s->handler->parse_packet) {
763		✗	rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
764			s->st, pkt, &timestamp, buf, len, seq,
765			flags);
766		✗	} else if (st) {
767		✗	if ((rv = av_new_packet(pkt, len)) < 0)
768		✗	return rv;
769		✗	memcpy(pkt->data, buf, len);
770		✗	pkt->stream_index = st->index;
771			} else {
772		✗	return AVERROR(EINVAL);
773			}
774
775			// now perform timestamp things....
776		✗	finalize_packet(s, pkt, timestamp);
777
778		✗	return rv;
779			}
780
781		✗	void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
782			{
783		✗	while (s->queue) {
784		✗	RTPPacket *next = s->queue->next;
785		✗	av_freep(&s->queue->buf);
786		✗	av_freep(&s->queue);
787		✗	s->queue = next;
788			}
789		✗	s->seq = 0;
790		✗	s->queue_len = 0;
791		✗	s->prev_ret = 0;
792		✗	}
793
794		✗	static int enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
795			{
796		✗	uint16_t seq = AV_RB16(buf + 2);
797		✗	RTPPacket **cur = &s->queue, *packet;
798
799			/* Find the correct place in the queue to insert the packet */
800		✗	while (*cur) {
801		✗	int16_t diff = seq - (*cur)->seq;
802		✗	if (diff < 0)
803		✗	break;
804		✗	cur = &(*cur)->next;
805			}
806
807		✗	packet = av_mallocz(sizeof(*packet));
808		✗	if (!packet)
809		✗	return AVERROR(ENOMEM);
810		✗	packet->recvtime = av_gettime_relative();
811		✗	packet->seq = seq;
812		✗	packet->len = len;
813		✗	packet->buf = buf;
814		✗	packet->next = *cur;
815		✗	*cur = packet;
816		✗	s->queue_len++;
817
818		✗	return 0;
819			}
820
821		✗	static int has_next_packet(RTPDemuxContext *s)
822			{
823		✗	return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
824			}
825
826		✗	int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
827			{
828		✗	return s->queue ? s->queue->recvtime : 0;
829			}
830
831		✗	static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
832			{
833			int rv;
834			RTPPacket *next;
835
836		✗	if (s->queue_len <= 0)
837		✗	return -1;
838
839		✗	if (!has_next_packet(s)) {
840		✗	int pkt_missed = s->queue->seq - s->seq - 1;
841
842		✗	if (pkt_missed < 0)
843		✗	pkt_missed += UINT16_MAX;
844		✗	av_log(s->ic, AV_LOG_WARNING,
845			"RTP: missed %d packets\n", pkt_missed);
846			}
847
848			/* Parse the first packet in the queue, and dequeue it */
849		✗	rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
850		✗	next = s->queue->next;
851		✗	av_freep(&s->queue->buf);
852		✗	av_freep(&s->queue);
853		✗	s->queue = next;
854		✗	s->queue_len--;
855		✗	return rv;
856			}
857
858		✗	static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
859			uint8_t **bufptr, int len)
860			{
861		✗	uint8_t *buf = bufptr ? *bufptr : NULL;
862		✗	int flags = 0;
863			uint32_t timestamp;
864		✗	int rv = 0;
865
866		✗	if (!buf) {
867			/* If parsing of the previous packet actually returned 0 or an error,
868			* there's nothing more to be parsed from that packet, but we may have
869			* indicated that we can return the next enqueued packet. */
870		✗	if (s->prev_ret <= 0)
871		✗	return rtp_parse_queued_packet(s, pkt);
872			/* return the next packets, if any */
873		✗	if (s->handler && s->handler->parse_packet) {
874			/* timestamp should be overwritten by parse_packet, if not,
875			* the packet is left with pts == AV_NOPTS_VALUE */
876		✗	timestamp = RTP_NOTS_VALUE;
877		✗	rv = s->handler->parse_packet(s->ic, s->dynamic_protocol_context,
878			s->st, pkt, &timestamp, NULL, 0, 0,
879			flags);
880		✗	finalize_packet(s, pkt, timestamp);
881		✗	return rv;
882			}
883			}
884
885		✗	if (len < 12)
886		✗	return -1;
887
888		✗	if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
889		✗	return -1;
890		✗	if (RTP_PT_IS_RTCP(buf[1])) {
891		✗	return rtcp_parse_packet(s, buf, len);
892			}
893
894		✗	if (s->st) {
895		✗	int64_t received = av_gettime_relative();
896		✗	uint32_t arrival_ts = av_rescale_q(received, AV_TIME_BASE_Q,
897		✗	s->st->time_base);
898		✗	timestamp = AV_RB32(buf + 4);
899			// Calculate the jitter immediately, before queueing the packet
900			// into the reordering queue.
901		✗	rtcp_update_jitter(&s->statistics, timestamp, arrival_ts);
902			}
903
904		✗	if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
905			/* First packet, or no reordering */
906		✗	return rtp_parse_packet_internal(s, pkt, buf, len);
907			} else {
908		✗	uint16_t seq = AV_RB16(buf + 2);
909		✗	int16_t diff = seq - s->seq;
910		✗	if (diff < 0) {
911			/* Packet older than the previously emitted one, drop */
912		✗	av_log(s->ic, AV_LOG_WARNING,
913			"RTP: dropping old packet received too late\n");
914		✗	return -1;
915		✗	} else if (diff <= 1) {
916			/* Correct packet */
917		✗	rv = rtp_parse_packet_internal(s, pkt, buf, len);
918		✗	return rv;
919			} else {
920			/* Still missing some packet, enqueue this one. */
921		✗	rv = enqueue_packet(s, buf, len);
922		✗	if (rv < 0)
923		✗	return rv;
924		✗	*bufptr = NULL;
925			/* Return the first enqueued packet if the queue is full,
926			* even if we're missing something */
927		✗	if (s->queue_len >= s->queue_size) {
928		✗	av_log(s->ic, AV_LOG_WARNING, "jitter buffer full\n");
929		✗	return rtp_parse_queued_packet(s, pkt);
930			}
931		✗	return -1;
932			}
933			}
934			}
935
936			/**
937			* Parse an RTP or RTCP packet directly sent as a buffer.
938			* @param s RTP parse context.
939			* @param pkt returned packet
940			* @param bufptr pointer to the input buffer or NULL to read the next packets
941			* @param len buffer len
942			* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
943			* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
944			*/
945		✗	int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
946			uint8_t **bufptr, int len)
947			{
948			int rv;
949		✗	if (s->srtp_enabled && bufptr && ff_srtp_decrypt(&s->srtp, *bufptr, &len) < 0)
950		✗	return -1;
951		✗	rv = rtp_parse_one_packet(s, pkt, bufptr, len);
952		✗	s->prev_ret = rv;
953		✗	while (rv < 0 && has_next_packet(s))
954		✗	rv = rtp_parse_queued_packet(s, pkt);
955		✗	return rv ? rv : has_next_packet(s);
956			}
957
958		✗	void ff_rtp_parse_close(RTPDemuxContext *s)
959			{
960		✗	ff_rtp_reset_packet_queue(s);
961		✗	ff_srtp_free(&s->srtp);
962		✗	av_free(s);
963		✗	}
964
965		✗	int ff_parse_fmtp(AVFormatContext *s,
966			AVStream *stream, PayloadContext *data, const char *p,
967			int (*parse_fmtp)(AVFormatContext *s,
968			AVStream *stream,
969			PayloadContext *data,
970			const char *attr, const char *value))
971			{
972			char attr[256];
973			char *value;
974			int res;
975		✗	int value_size = strlen(p) + 1;
976
977		✗	if (!(value = av_malloc(value_size))) {
978		✗	av_log(s, AV_LOG_ERROR, "Failed to allocate data for FMTP.\n");
979		✗	return AVERROR(ENOMEM);
980			}
981
982			// remove protocol identifier
983		✗	while (*p && *p == ' ')
984		✗	p++; // strip spaces
985		✗	while (*p && *p != ' ')
986		✗	p++; // eat protocol identifier
987		✗	while (*p && *p == ' ')
988		✗	p++; // strip trailing spaces
989
990		✗	while (ff_rtsp_next_attr_and_value(&p,
991			attr, sizeof(attr),
992			value, value_size)) {
993		✗	res = parse_fmtp(s, stream, data, attr, value);
994		✗	if (res < 0 && res != AVERROR_PATCHWELCOME) {
995		✗	av_free(value);
996		✗	return res;
997			}
998			}
999		✗	av_free(value);
1000		✗	return 0;
1001			}
1002
1003		✗	int ff_rtp_finalize_packet(AVPacket *pkt, AVIOContext **dyn_buf, int stream_idx)
1004			{
1005			int ret;
1006		✗	av_packet_unref(pkt);
1007
1008		✗	pkt->size = avio_close_dyn_buf(*dyn_buf, &pkt->data);
1009		✗	pkt->stream_index = stream_idx;
1010		✗	*dyn_buf = NULL;
1011		✗	if ((ret = av_packet_from_data(pkt, pkt->data, pkt->size)) < 0) {
1012		✗	av_freep(&pkt->data);
1013		✗	return ret;
1014			}
1015		✗	return pkt->size;
1016			}
1017