lavc: add AV prefix to codec ids.
[libav.git] / libavformat / rtpdec.c
CommitLineData
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1/*
2 * RTP input format
406792e7 3 * Copyright (c) 2002 Fabrice Bellard
8eb793c4 4 *
2912e87a 5 * This file is part of Libav.
8eb793c4 6 *
2912e87a 7 * Libav is free software; you can redistribute it and/or
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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 *
2912e87a 12 * Libav is distributed in the hope that it will be useful,
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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
2912e87a 18 * License along with Libav; if not, write to the Free Software
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
245976da 21
0ebcdf5c 22#include "libavutil/mathematics.h"
bb3244de 23#include "libavutil/avstring.h"
c4ef6a3e 24#include "libavutil/time.h"
9106a698 25#include "libavcodec/get_bits.h"
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26#include "avformat.h"
27#include "mpegts.h"
925e908b 28#include "url.h"
8eb793c4 29
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30#include "network.h"
31
302879cb 32#include "rtpdec.h"
965a3ddb 33#include "rtpdec_formats.h"
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34
35//#define DEBUG
36
37/* TODO: - add RTCP statistics reporting (should be optional).
38
39 - add support for h263/mpeg4 packetized output : IDEA: send a
40 buffer to 'rtp_write_packet' contains all the packets for ONE
41 frame. Each packet should have a four byte header containing
42 the length in big endian format (same trick as
403ee835 43 'ffio_open_dyn_packet_buf')
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44*/
45
69ad22c7 46static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
2eeefe20
MS
47 .enc_name = "X-MP3-draft-00",
48 .codec_type = AVMEDIA_TYPE_AUDIO,
49 .codec_id = CODEC_ID_MP3ADU,
50};
51
8eb793c4 52/* statistics functions */
119cc033 53static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
8eb793c4 54
0369d2b0 55void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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56{
57 handler->next= RTPFirstDynamicPayloadHandler;
58 RTPFirstDynamicPayloadHandler= handler;
59}
60
61void av_register_rtp_dynamic_payload_handlers(void)
62{
9b3788ef
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63 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
64 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
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RB
65 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
66 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
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RB
67 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
08bddfcd 69 ff_register_dynamic_payload_handler(&ff_h263_rfc2190_dynamic_handler);
0369d2b0 70 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
89c39605 71 ff_register_dynamic_payload_handler(&ff_ilbc_dynamic_handler);
e6327fba 72 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
887af2aa 73 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
a59096e4 74 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
4449df6b 75 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
1ddc176e 76 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
51291e60 77 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
35014efc 78 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
2eeefe20 79 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
e9fce261
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80
81 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
82 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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83
84 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
85 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
86 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
87 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
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MS
88
89 ff_register_dynamic_payload_handler(&ff_g726_16_dynamic_handler);
90 ff_register_dynamic_payload_handler(&ff_g726_24_dynamic_handler);
91 ff_register_dynamic_payload_handler(&ff_g726_32_dynamic_handler);
92 ff_register_dynamic_payload_handler(&ff_g726_40_dynamic_handler);
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93}
94
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MS
95RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
96 enum AVMediaType codec_type)
97{
98 RTPDynamicProtocolHandler *handler;
99 for (handler = RTPFirstDynamicPayloadHandler;
100 handler; handler = handler->next)
bb3244de 101 if (!av_strcasecmp(name, handler->enc_name) &&
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MS
102 codec_type == handler->codec_type)
103 return handler;
104 return NULL;
105}
106
107RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
108 enum AVMediaType codec_type)
109{
110 RTPDynamicProtocolHandler *handler;
111 for (handler = RTPFirstDynamicPayloadHandler;
112 handler; handler = handler->next)
113 if (handler->static_payload_id && handler->static_payload_id == id &&
114 codec_type == handler->codec_type)
115 return handler;
116 return NULL;
117}
118
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119static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
120{
ff328c02 121 int payload_len;
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122 while (len >= 4) {
123 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
124
ff328c02
JA
125 switch (buf[1]) {
126 case RTCP_SR:
07b77fe3 127 if (payload_len < 20) {
ff328c02
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128 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
129 return AVERROR_INVALIDDATA;
130 }
ff328c02 131
682d28a9 132 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
682d28a9 133 s->last_rtcp_timestamp = AV_RB32(buf + 16);
3a1cdcc7
MS
134 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
135 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
136 if (!s->base_timestamp)
137 s->base_timestamp = s->last_rtcp_timestamp;
138 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
139 }
ff328c02 140
ff328c02 141 break;
b20359f5
JA
142 case RTCP_BYE:
143 return -RTCP_BYE;
ff328c02 144 }
07b77fe3
JB
145
146 buf += payload_len;
147 len -= payload_len;
ff328c02 148 }
b20359f5 149 return -1;
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150}
151
152#define RTP_SEQ_MOD (1<<16)
153
154/**
155* called on parse open packet
156*/
157static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
158{
159 memset(s, 0, sizeof(RTPStatistics));
160 s->max_seq= base_sequence;
161 s->probation= 1;
162}
163
164/**
165* called whenever there is a large jump in sequence numbers, or when they get out of probation...
166*/
167static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
168{
169 s->max_seq= seq;
170 s->cycles= 0;
171 s->base_seq= seq -1;
172 s->bad_seq= RTP_SEQ_MOD + 1;
173 s->received= 0;
174 s->expected_prior= 0;
175 s->received_prior= 0;
176 s->jitter= 0;
177 s->transit= 0;
178}
179
180/**
181* returns 1 if we should handle this packet.
182*/
183static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
184{
185 uint16_t udelta= seq - s->max_seq;
186 const int MAX_DROPOUT= 3000;
187 const int MAX_MISORDER = 100;
188 const int MIN_SEQUENTIAL = 2;
189
190 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
191 if(s->probation)
192 {
193 if(seq==s->max_seq + 1) {
194 s->probation--;
195 s->max_seq= seq;
196 if(s->probation==0) {
197 rtp_init_sequence(s, seq);
198 s->received++;
199 return 1;
200 }
201 } else {
202 s->probation= MIN_SEQUENTIAL - 1;
203 s->max_seq = seq;
204 }
205 } else if (udelta < MAX_DROPOUT) {
206 // in order, with permissible gap
207 if(seq < s->max_seq) {
208 //sequence number wrapped; count antother 64k cycles
209 s->cycles += RTP_SEQ_MOD;
210 }
211 s->max_seq= seq;
212 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
213 // sequence made a large jump...
214 if(seq==s->bad_seq) {
215 // two sequential packets-- assume that the other side restarted without telling us; just resync.
216 rtp_init_sequence(s, seq);
217 } else {
218 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
219 return 0;
220 }
221 } else {
222 // duplicate or reordered packet...
223 }
224 s->received++;
225 return 1;
226}
227
bfc6db44 228int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
8eb793c4 229{
ae628ec1 230 AVIOContext *pb;
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231 uint8_t *buf;
232 int len;
233 int rtcp_bytes;
234 RTPStatistics *stats= &s->statistics;
235 uint32_t lost;
236 uint32_t extended_max;
237 uint32_t expected_interval;
238 uint32_t received_interval;
239 uint32_t lost_interval;
240 uint32_t expected;
241 uint32_t fraction;
242 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
243
244 if (!s->rtp_ctx || (count < 1))
245 return -1;
246
247 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
248 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
249 s->octet_count += count;
250 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
251 RTCP_TX_RATIO_DEN;
252 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
253 if (rtcp_bytes < 28)
254 return -1;
255 s->last_octet_count = s->octet_count;
256
b92c5452 257 if (avio_open_dyn_buf(&pb) < 0)
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258 return -1;
259
260 // Receiver Report
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261 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
262 avio_w8(pb, RTCP_RR);
263 avio_wb16(pb, 7); /* length in words - 1 */
952139a3 264 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
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265 avio_wb32(pb, s->ssrc + 1);
266 avio_wb32(pb, s->ssrc); // server SSRC
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267 // some placeholders we should really fill...
268 // RFC 1889/p64
269 extended_max= stats->cycles + stats->max_seq;
270 expected= extended_max - stats->base_seq + 1;
271 lost= expected - stats->received;
272 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
273 expected_interval= expected - stats->expected_prior;
274 stats->expected_prior= expected;
275 received_interval= stats->received - stats->received_prior;
276 stats->received_prior= stats->received;
277 lost_interval= expected_interval - received_interval;
278 if (expected_interval==0 || lost_interval<=0) fraction= 0;
279 else fraction = (lost_interval<<8)/expected_interval;
280
281 fraction= (fraction<<24) | lost;
282
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283 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
284 avio_wb32(pb, extended_max); /* max sequence received */
285 avio_wb32(pb, stats->jitter>>4); /* jitter */
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286
287 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
288 {
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289 avio_wb32(pb, 0); /* last SR timestamp */
290 avio_wb32(pb, 0); /* delay since last SR */
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291 } else {
292 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
293 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
294
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295 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
296 avio_wb32(pb, delay_since_last); /* delay since last SR */
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297 }
298
299 // CNAME
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300 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
301 avio_w8(pb, RTCP_SDES);
8eb793c4 302 len = strlen(s->hostname);
77eb5504 303 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
ad7beb2c 304 avio_wb32(pb, s->ssrc + 1);
77eb5504
AK
305 avio_w8(pb, 0x01);
306 avio_w8(pb, len);
307 avio_write(pb, s->hostname, len);
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308 // padding
309 for (len = (6 + len) % 4; len % 4; len++) {
77eb5504 310 avio_w8(pb, 0);
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311 }
312
b7f2fdde 313 avio_flush(pb);
6dc7d80d 314 len = avio_close_dyn_buf(pb, &buf);
8eb793c4 315 if ((len > 0) && buf) {
5e1166b3 316 int av_unused result;
dfd2a005 317 av_dlog(s->ic, "sending %d bytes of RR\n", len);
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AK
318 result= ffurl_write(s->rtp_ctx, buf, len);
319 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
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320 av_free(buf);
321 }
322 return 0;
323}
324
bfc6db44 325void ff_rtp_send_punch_packets(URLContext* rtp_handle)
9c8fa20d 326{
ae628ec1 327 AVIOContext *pb;
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MS
328 uint8_t *buf;
329 int len;
330
331 /* Send a small RTP packet */
b92c5452 332 if (avio_open_dyn_buf(&pb) < 0)
9c8fa20d
MS
333 return;
334
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335 avio_w8(pb, (RTP_VERSION << 6));
336 avio_w8(pb, 0); /* Payload type */
337 avio_wb16(pb, 0); /* Seq */
338 avio_wb32(pb, 0); /* Timestamp */
339 avio_wb32(pb, 0); /* SSRC */
9c8fa20d 340
b7f2fdde 341 avio_flush(pb);
6dc7d80d 342 len = avio_close_dyn_buf(pb, &buf);
9c8fa20d 343 if ((len > 0) && buf)
925e908b 344 ffurl_write(rtp_handle, buf, len);
9c8fa20d
MS
345 av_free(buf);
346
347 /* Send a minimal RTCP RR */
b92c5452 348 if (avio_open_dyn_buf(&pb) < 0)
9c8fa20d
MS
349 return;
350
77eb5504
AK
351 avio_w8(pb, (RTP_VERSION << 6));
352 avio_w8(pb, RTCP_RR); /* receiver report */
353 avio_wb16(pb, 1); /* length in words - 1 */
354 avio_wb32(pb, 0); /* our own SSRC */
9c8fa20d 355
b7f2fdde 356 avio_flush(pb);
6dc7d80d 357 len = avio_close_dyn_buf(pb, &buf);
9c8fa20d 358 if ((len > 0) && buf)
925e908b 359 ffurl_write(rtp_handle, buf, len);
9c8fa20d
MS
360 av_free(buf);
361}
362
363
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364/**
365 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
366 * MPEG2TS streams to indicate that they should be demuxed inside the
367 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
8eb793c4 368 */
bfc6db44 369RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
8eb793c4
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370{
371 RTPDemuxContext *s;
372
373 s = av_mallocz(sizeof(RTPDemuxContext));
374 if (!s)
375 return NULL;
376 s->payload_type = payload_type;
377 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
2cab6b48 378 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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379 s->ic = s1;
380 s->st = st;
58ee0991 381 s->queue_size = queue_size;
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382 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
383 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 384 s->ts = ff_mpegts_parse_open(s->ic);
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LA
385 if (s->ts == NULL) {
386 av_free(s);
387 return NULL;
388 }
45600148 389 } else if (st) {
8eb793c4
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390 switch(st->codec->codec_id) {
391 case CODEC_ID_MPEG1VIDEO:
392 case CODEC_ID_MPEG2VIDEO:
393 case CODEC_ID_MP2:
394 case CODEC_ID_MP3:
395 case CODEC_ID_MPEG4:
45aa9080 396 case CODEC_ID_H263:
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LA
397 case CODEC_ID_H264:
398 st->need_parsing = AVSTREAM_PARSE_FULL;
399 break;
5602a464
JR
400 case CODEC_ID_VORBIS:
401 st->need_parsing = AVSTREAM_PARSE_HEADERS;
402 break;
0048a2a8 403 case CODEC_ID_ADPCM_G722:
0048a2a8
MS
404 /* According to RFC 3551, the stream clock rate is 8000
405 * even if the sample rate is 16000. */
406 if (st->codec->sample_rate == 8000)
407 st->codec->sample_rate = 16000;
408 break;
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409 default:
410 break;
411 }
412 }
413 // needed to send back RTCP RR in RTSP sessions
414 s->rtp_ctx = rtpc;
415 gethostname(s->hostname, sizeof(s->hostname));
416 return s;
417}
418
99a1d191 419void
bfc6db44
MS
420ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
421 RTPDynamicProtocolHandler *handler)
99a1d191
RB
422{
423 s->dynamic_protocol_context = ctx;
424 s->parse_packet = handler->parse_packet;
425}
426
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LA
427/**
428 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
429 */
430static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
431{
79d482b1
MS
432 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
433 return; /* Timestamp already set by depacketizer */
b8a1b880
JB
434 if (timestamp == RTP_NOTS_VALUE)
435 return;
436
525c5b08 437 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && s->ic->nb_streams > 1) {
fba7815d
LA
438 int64_t addend;
439 int delta_timestamp;
440
441 /* compute pts from timestamp with received ntp_time */
442 delta_timestamp = timestamp - s->last_rtcp_timestamp;
443 /* convert to the PTS timebase */
2cab6b48 444 addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32);
3a1cdcc7
MS
445 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
446 delta_timestamp;
447 return;
fba7815d 448 }
b8a1b880 449
3a1cdcc7
MS
450 if (!s->base_timestamp)
451 s->base_timestamp = timestamp;
12348ca2
JB
452 /* assume that the difference is INT32_MIN < x < INT32_MAX, but allow the first timestamp to exceed INT32_MAX */
453 if (!s->timestamp)
454 s->unwrapped_timestamp += timestamp;
455 else
456 s->unwrapped_timestamp += (int32_t)(timestamp - s->timestamp);
457 s->timestamp = timestamp;
458 pkt->pts = s->unwrapped_timestamp + s->range_start_offset - s->base_timestamp;
8eb793c4
LA
459}
460
02607418
MS
461static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
462 const uint8_t *buf, int len)
8eb793c4
LA
463{
464 unsigned int ssrc, h;
f841a0fc 465 int payload_type, seq, ret, flags = 0;
9446b4bb 466 int ext;
8eb793c4
LA
467 AVStream *st;
468 uint32_t timestamp;
469 int rv= 0;
470
9446b4bb 471 ext = buf[0] & 0x10;
8eb793c4 472 payload_type = buf[1] & 0x7f;
144ae29d
RB
473 if (buf[1] & 0x80)
474 flags |= RTP_FLAG_MARKER;
8eb793c4
LA
475 seq = AV_RB16(buf + 2);
476 timestamp = AV_RB32(buf + 4);
477 ssrc = AV_RB32(buf + 8);
478 /* store the ssrc in the RTPDemuxContext */
479 s->ssrc = ssrc;
480
481 /* NOTE: we can handle only one payload type */
482 if (s->payload_type != payload_type)
483 return -1;
484
485 st = s->st;
486 // only do something with this if all the rtp checks pass...
487 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
488 {
489 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
490 payload_type, seq, ((s->seq + 1) & 0xffff));
491 return -1;
492 }
493
4838cdab
MS
494 if (buf[0] & 0x20) {
495 int padding = buf[len - 1];
496 if (len >= 12 + padding)
497 len -= padding;
498 }
499
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LA
500 s->seq = seq;
501 len -= 12;
502 buf += 12;
503
9446b4bb
RS
504 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
505 if (ext) {
506 if (len < 4)
507 return -1;
508 /* calculate the header extension length (stored as number
509 * of 32-bit words) */
510 ext = (AV_RB16(buf + 2) + 1) << 2;
511
512 if (len < ext)
513 return -1;
514 // skip past RTP header extension
515 len -= ext;
516 buf += ext;
517 }
518
8eb793c4
LA
519 if (!st) {
520 /* specific MPEG2TS demux support */
9125806e 521 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
946df059
MS
522 /* The only error that can be returned from ff_mpegts_parse_packet
523 * is "no more data to return from the provided buffer", so return
524 * AVERROR(EAGAIN) for all errors */
4ffff367 525 if (ret < 0)
946df059 526 return AVERROR(EAGAIN);
8eb793c4
LA
527 if (ret < len) {
528 s->read_buf_size = len - ret;
529 memcpy(s->buf, buf + ret, s->read_buf_size);
530 s->read_buf_index = 0;
531 return 1;
532 }
f3e71942 533 return 0;
b4e3330c 534 } else if (s->parse_packet) {
1a45a9f4 535 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 536 s->st, pkt, &timestamp, buf, len, flags);
8eb793c4
LA
537 } else {
538 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
539 switch(st->codec->codec_id) {
540 case CODEC_ID_MP2:
76faff6e 541 case CODEC_ID_MP3:
8eb793c4
LA
542 /* better than nothing: skip mpeg audio RTP header */
543 if (len <= 4)
544 return -1;
545 h = AV_RB32(buf);
546 len -= 4;
547 buf += 4;
548 av_new_packet(pkt, len);
549 memcpy(pkt->data, buf, len);
550 break;
551 case CODEC_ID_MPEG1VIDEO:
552 case CODEC_ID_MPEG2VIDEO:
553 /* better than nothing: skip mpeg video RTP header */
554 if (len <= 4)
555 return -1;
556 h = AV_RB32(buf);
557 buf += 4;
558 len -= 4;
559 if (h & (1 << 26)) {
560 /* mpeg2 */
561 if (len <= 4)
562 return -1;
563 buf += 4;
564 len -= 4;
565 }
566 av_new_packet(pkt, len);
567 memcpy(pkt->data, buf, len);
568 break;
8eb793c4 569 default:
f739b36d
RB
570 av_new_packet(pkt, len);
571 memcpy(pkt->data, buf, len);
8eb793c4
LA
572 break;
573 }
eafb17d1
RB
574
575 pkt->stream_index = st->index;
f3e71942 576 }
8eb793c4 577
95f03cf3
RB
578 // now perform timestamp things....
579 finalize_packet(s, pkt, timestamp);
f3e71942 580
8eb793c4
LA
581 return rv;
582}
583
58ee0991
MS
584void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
585{
586 while (s->queue) {
587 RTPPacket *next = s->queue->next;
588 av_free(s->queue->buf);
589 av_free(s->queue);
590 s->queue = next;
591 }
592 s->seq = 0;
593 s->queue_len = 0;
594 s->prev_ret = 0;
595}
596
597static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
598{
599 uint16_t seq = AV_RB16(buf + 2);
600 RTPPacket *cur = s->queue, *prev = NULL, *packet;
601
602 /* Find the correct place in the queue to insert the packet */
603 while (cur) {
604 int16_t diff = seq - cur->seq;
605 if (diff < 0)
606 break;
607 prev = cur;
608 cur = cur->next;
609 }
610
611 packet = av_mallocz(sizeof(*packet));
612 if (!packet)
613 return;
614 packet->recvtime = av_gettime();
615 packet->seq = seq;
616 packet->len = len;
617 packet->buf = buf;
618 packet->next = cur;
619 if (prev)
620 prev->next = packet;
621 else
622 s->queue = packet;
623 s->queue_len++;
624}
625
626static int has_next_packet(RTPDemuxContext *s)
627{
ddcf8411 628 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
58ee0991
MS
629}
630
631int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
632{
633 return s->queue ? s->queue->recvtime : 0;
634}
635
636static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
637{
638 int rv;
639 RTPPacket *next;
640
641 if (s->queue_len <= 0)
642 return -1;
643
644 if (!has_next_packet(s))
645 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
646 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
647
648 /* Parse the first packet in the queue, and dequeue it */
649 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
650 next = s->queue->next;
651 av_free(s->queue->buf);
652 av_free(s->queue);
653 s->queue = next;
654 s->queue_len--;
4ffff367 655 return rv;
58ee0991
MS
656}
657
4ffff367 658static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
02607418
MS
659 uint8_t **bufptr, int len)
660{
661 uint8_t* buf = bufptr ? *bufptr : NULL;
662 int ret, flags = 0;
663 uint32_t timestamp;
664 int rv= 0;
665
666 if (!buf) {
f6e138b4
MS
667 /* If parsing of the previous packet actually returned 0 or an error,
668 * there's nothing more to be parsed from that packet, but we may have
58ee0991 669 * indicated that we can return the next enqueued packet. */
f6e138b4 670 if (s->prev_ret <= 0)
58ee0991 671 return rtp_parse_queued_packet(s, pkt);
02607418
MS
672 /* return the next packets, if any */
673 if(s->st && s->parse_packet) {
674 /* timestamp should be overwritten by parse_packet, if not,
675 * the packet is left with pts == AV_NOPTS_VALUE */
676 timestamp = RTP_NOTS_VALUE;
677 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
678 s->st, pkt, &timestamp, NULL, 0, flags);
679 finalize_packet(s, pkt, timestamp);
4ffff367 680 return rv;
02607418
MS
681 } else {
682 // TODO: Move to a dynamic packet handler (like above)
4ffff367 683 if (s->read_buf_index >= s->read_buf_size)
91ec7aea 684 return AVERROR(EAGAIN);
02607418
MS
685 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
686 s->read_buf_size - s->read_buf_index);
4ffff367 687 if (ret < 0)
946df059 688 return AVERROR(EAGAIN);
02607418
MS
689 s->read_buf_index += ret;
690 if (s->read_buf_index < s->read_buf_size)
691 return 1;
4ffff367
MS
692 else
693 return 0;
02607418
MS
694 }
695 }
696
697 if (len < 12)
698 return -1;
699
700 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
701 return -1;
298a587f 702 if (RTP_PT_IS_RTCP(buf[1])) {
02607418
MS
703 return rtcp_parse_packet(s, buf, len);
704 }
705
65cdee9c 706 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
58ee0991
MS
707 /* First packet, or no reordering */
708 return rtp_parse_packet_internal(s, pkt, buf, len);
709 } else {
710 uint16_t seq = AV_RB16(buf + 2);
711 int16_t diff = seq - s->seq;
712 if (diff < 0) {
713 /* Packet older than the previously emitted one, drop */
714 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
715 "RTP: dropping old packet received too late\n");
716 return -1;
717 } else if (diff <= 1) {
718 /* Correct packet */
719 rv = rtp_parse_packet_internal(s, pkt, buf, len);
4ffff367 720 return rv;
58ee0991
MS
721 } else {
722 /* Still missing some packet, enqueue this one. */
723 enqueue_packet(s, buf, len);
724 *bufptr = NULL;
725 /* Return the first enqueued packet if the queue is full,
726 * even if we're missing something */
727 if (s->queue_len >= s->queue_size)
728 return rtp_parse_queued_packet(s, pkt);
729 return -1;
730 }
731 }
02607418
MS
732}
733
4ffff367
MS
734/**
735 * Parse an RTP or RTCP packet directly sent as a buffer.
736 * @param s RTP parse context.
737 * @param pkt returned packet
738 * @param bufptr pointer to the input buffer or NULL to read the next packets
739 * @param len buffer len
740 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
741 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
742 */
bfc6db44
MS
743int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
744 uint8_t **bufptr, int len)
4ffff367
MS
745{
746 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
747 s->prev_ret = rv;
d678a6fd
MS
748 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
749 rv = rtp_parse_queued_packet(s, pkt);
4ffff367
MS
750 return rv ? rv : has_next_packet(s);
751}
752
bfc6db44 753void ff_rtp_parse_close(RTPDemuxContext *s)
8eb793c4 754{
58ee0991 755 ff_rtp_reset_packet_queue(s);
8eb793c4 756 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 757 ff_mpegts_parse_close(s->ts);
8eb793c4
LA
758 }
759 av_free(s);
760}
016bc031
JA
761
762int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
763 int (*parse_fmtp)(AVStream *stream,
764 PayloadContext *data,
765 char *attr, char *value))
766{
767 char attr[256];
824535e3 768 char *value;
016bc031 769 int res;
824535e3
JA
770 int value_size = strlen(p) + 1;
771
772 if (!(value = av_malloc(value_size))) {
773 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
774 return AVERROR(ENOMEM);
775 }
016bc031
JA
776
777 // remove protocol identifier
778 while (*p && *p == ' ') p++; // strip spaces
779 while (*p && *p != ' ') p++; // eat protocol identifier
780 while (*p && *p == ' ') p++; // strip trailing spaces
781
782 while (ff_rtsp_next_attr_and_value(&p,
783 attr, sizeof(attr),
824535e3 784 value, value_size)) {
016bc031
JA
785
786 res = parse_fmtp(stream, data, attr, value);
824535e3
JA
787 if (res < 0 && res != AVERROR_PATCHWELCOME) {
788 av_free(value);
016bc031 789 return res;
824535e3 790 }
016bc031 791 }
824535e3 792 av_free(value);
016bc031
JA
793 return 0;
794}