avio: make url_read_complete() internal.
[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
7246177d 22/* needed for gethostname() */
d0feff2a 23#define _XOPEN_SOURCE 600
7246177d 24
9106a698 25#include "libavcodec/get_bits.h"
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26#include "avformat.h"
27#include "mpegts.h"
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28
29#include <unistd.h>
1e515c42 30#include <strings.h>
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31#include "network.h"
32
302879cb 33#include "rtpdec.h"
965a3ddb 34#include "rtpdec_formats.h"
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35
36//#define DEBUG
37
38/* TODO: - add RTCP statistics reporting (should be optional).
39
40 - add support for h263/mpeg4 packetized output : IDEA: send a
41 buffer to 'rtp_write_packet' contains all the packets for ONE
42 frame. Each packet should have a four byte header containing
43 the length in big endian format (same trick as
403ee835 44 'ffio_open_dyn_packet_buf')
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45*/
46
69ad22c7 47static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
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48 .enc_name = "X-MP3-draft-00",
49 .codec_type = AVMEDIA_TYPE_AUDIO,
50 .codec_id = CODEC_ID_MP3ADU,
51};
52
8eb793c4 53/* statistics functions */
119cc033 54static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
8eb793c4 55
0369d2b0 56void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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57{
58 handler->next= RTPFirstDynamicPayloadHandler;
59 RTPFirstDynamicPayloadHandler= handler;
60}
61
62void av_register_rtp_dynamic_payload_handlers(void)
63{
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64 ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler);
65 ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler);
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66 ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler);
67 ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler);
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68 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
69 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
0369d2b0 70 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
e6327fba 71 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
887af2aa 72 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
a59096e4 73 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
4449df6b 74 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
1ddc176e 75 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
51291e60 76 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
35014efc 77 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
2eeefe20 78 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
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79
80 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
81 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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82
83 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
84 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
85 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
86 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
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87}
88
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89RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
90 enum AVMediaType codec_type)
91{
92 RTPDynamicProtocolHandler *handler;
93 for (handler = RTPFirstDynamicPayloadHandler;
94 handler; handler = handler->next)
95 if (!strcasecmp(name, handler->enc_name) &&
96 codec_type == handler->codec_type)
97 return handler;
98 return NULL;
99}
100
101RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
102 enum AVMediaType codec_type)
103{
104 RTPDynamicProtocolHandler *handler;
105 for (handler = RTPFirstDynamicPayloadHandler;
106 handler; handler = handler->next)
107 if (handler->static_payload_id && handler->static_payload_id == id &&
108 codec_type == handler->codec_type)
109 return handler;
110 return NULL;
111}
112
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113static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
114{
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115 int payload_len;
116 while (len >= 2) {
117 switch (buf[1]) {
118 case RTCP_SR:
119 if (len < 16) {
120 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
121 return AVERROR_INVALIDDATA;
122 }
123 payload_len = (AV_RB16(buf + 2) + 1) * 4;
124
682d28a9 125 s->last_rtcp_ntp_time = AV_RB64(buf + 8);
682d28a9 126 s->last_rtcp_timestamp = AV_RB32(buf + 16);
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127 if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) {
128 s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
129 if (!s->base_timestamp)
130 s->base_timestamp = s->last_rtcp_timestamp;
131 s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
132 }
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133
134 buf += payload_len;
135 len -= payload_len;
136 break;
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137 case RTCP_BYE:
138 return -RTCP_BYE;
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139 default:
140 return -1;
141 }
142 }
b20359f5 143 return -1;
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144}
145
146#define RTP_SEQ_MOD (1<<16)
147
148/**
149* called on parse open packet
150*/
151static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
152{
153 memset(s, 0, sizeof(RTPStatistics));
154 s->max_seq= base_sequence;
155 s->probation= 1;
156}
157
158/**
159* called whenever there is a large jump in sequence numbers, or when they get out of probation...
160*/
161static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
162{
163 s->max_seq= seq;
164 s->cycles= 0;
165 s->base_seq= seq -1;
166 s->bad_seq= RTP_SEQ_MOD + 1;
167 s->received= 0;
168 s->expected_prior= 0;
169 s->received_prior= 0;
170 s->jitter= 0;
171 s->transit= 0;
172}
173
174/**
175* returns 1 if we should handle this packet.
176*/
177static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
178{
179 uint16_t udelta= seq - s->max_seq;
180 const int MAX_DROPOUT= 3000;
181 const int MAX_MISORDER = 100;
182 const int MIN_SEQUENTIAL = 2;
183
184 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
185 if(s->probation)
186 {
187 if(seq==s->max_seq + 1) {
188 s->probation--;
189 s->max_seq= seq;
190 if(s->probation==0) {
191 rtp_init_sequence(s, seq);
192 s->received++;
193 return 1;
194 }
195 } else {
196 s->probation= MIN_SEQUENTIAL - 1;
197 s->max_seq = seq;
198 }
199 } else if (udelta < MAX_DROPOUT) {
200 // in order, with permissible gap
201 if(seq < s->max_seq) {
202 //sequence number wrapped; count antother 64k cycles
203 s->cycles += RTP_SEQ_MOD;
204 }
205 s->max_seq= seq;
206 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
207 // sequence made a large jump...
208 if(seq==s->bad_seq) {
209 // two sequential packets-- assume that the other side restarted without telling us; just resync.
210 rtp_init_sequence(s, seq);
211 } else {
212 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
213 return 0;
214 }
215 } else {
216 // duplicate or reordered packet...
217 }
218 s->received++;
219 return 1;
220}
221
222#if 0
223/**
224* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
225* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
226* never change. I left this in in case someone else can see a way. (rdm)
227*/
228static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
229{
230 uint32_t transit= arrival_timestamp - sent_timestamp;
231 int d;
232 s->transit= transit;
233 d= FFABS(transit - s->transit);
234 s->jitter += d - ((s->jitter + 8)>>4);
235}
236#endif
237
238int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
239{
ae628ec1 240 AVIOContext *pb;
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241 uint8_t *buf;
242 int len;
243 int rtcp_bytes;
244 RTPStatistics *stats= &s->statistics;
245 uint32_t lost;
246 uint32_t extended_max;
247 uint32_t expected_interval;
248 uint32_t received_interval;
249 uint32_t lost_interval;
250 uint32_t expected;
251 uint32_t fraction;
252 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
253
254 if (!s->rtp_ctx || (count < 1))
255 return -1;
256
257 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
258 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
259 s->octet_count += count;
260 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
261 RTCP_TX_RATIO_DEN;
262 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
263 if (rtcp_bytes < 28)
264 return -1;
265 s->last_octet_count = s->octet_count;
266
b92c5452 267 if (avio_open_dyn_buf(&pb) < 0)
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268 return -1;
269
270 // Receiver Report
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271 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
272 avio_w8(pb, RTCP_RR);
273 avio_wb16(pb, 7); /* length in words - 1 */
952139a3 274 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
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275 avio_wb32(pb, s->ssrc + 1);
276 avio_wb32(pb, s->ssrc); // server SSRC
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277 // some placeholders we should really fill...
278 // RFC 1889/p64
279 extended_max= stats->cycles + stats->max_seq;
280 expected= extended_max - stats->base_seq + 1;
281 lost= expected - stats->received;
282 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
283 expected_interval= expected - stats->expected_prior;
284 stats->expected_prior= expected;
285 received_interval= stats->received - stats->received_prior;
286 stats->received_prior= stats->received;
287 lost_interval= expected_interval - received_interval;
288 if (expected_interval==0 || lost_interval<=0) fraction= 0;
289 else fraction = (lost_interval<<8)/expected_interval;
290
291 fraction= (fraction<<24) | lost;
292
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293 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
294 avio_wb32(pb, extended_max); /* max sequence received */
295 avio_wb32(pb, stats->jitter>>4); /* jitter */
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296
297 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
298 {
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299 avio_wb32(pb, 0); /* last SR timestamp */
300 avio_wb32(pb, 0); /* delay since last SR */
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301 } else {
302 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
303 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
304
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305 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
306 avio_wb32(pb, delay_since_last); /* delay since last SR */
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307 }
308
309 // CNAME
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310 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
311 avio_w8(pb, RTCP_SDES);
8eb793c4 312 len = strlen(s->hostname);
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313 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
314 avio_wb32(pb, s->ssrc);
315 avio_w8(pb, 0x01);
316 avio_w8(pb, len);
317 avio_write(pb, s->hostname, len);
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318 // padding
319 for (len = (6 + len) % 4; len % 4; len++) {
77eb5504 320 avio_w8(pb, 0);
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321 }
322
b7f2fdde 323 avio_flush(pb);
6dc7d80d 324 len = avio_close_dyn_buf(pb, &buf);
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325 if ((len > 0) && buf) {
326 int result;
dfd2a005 327 av_dlog(s->ic, "sending %d bytes of RR\n", len);
8eb793c4 328 result= url_write(s->rtp_ctx, buf, len);
dfd2a005 329 av_dlog(s->ic, "result from url_write: %d\n", result);
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330 av_free(buf);
331 }
332 return 0;
333}
334
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335void rtp_send_punch_packets(URLContext* rtp_handle)
336{
ae628ec1 337 AVIOContext *pb;
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338 uint8_t *buf;
339 int len;
340
341 /* Send a small RTP packet */
b92c5452 342 if (avio_open_dyn_buf(&pb) < 0)
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343 return;
344
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345 avio_w8(pb, (RTP_VERSION << 6));
346 avio_w8(pb, 0); /* Payload type */
347 avio_wb16(pb, 0); /* Seq */
348 avio_wb32(pb, 0); /* Timestamp */
349 avio_wb32(pb, 0); /* SSRC */
9c8fa20d 350
b7f2fdde 351 avio_flush(pb);
6dc7d80d 352 len = avio_close_dyn_buf(pb, &buf);
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353 if ((len > 0) && buf)
354 url_write(rtp_handle, buf, len);
355 av_free(buf);
356
357 /* Send a minimal RTCP RR */
b92c5452 358 if (avio_open_dyn_buf(&pb) < 0)
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359 return;
360
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361 avio_w8(pb, (RTP_VERSION << 6));
362 avio_w8(pb, RTCP_RR); /* receiver report */
363 avio_wb16(pb, 1); /* length in words - 1 */
364 avio_wb32(pb, 0); /* our own SSRC */
9c8fa20d 365
b7f2fdde 366 avio_flush(pb);
6dc7d80d 367 len = avio_close_dyn_buf(pb, &buf);
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368 if ((len > 0) && buf)
369 url_write(rtp_handle, buf, len);
370 av_free(buf);
371}
372
373
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374/**
375 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
376 * MPEG2TS streams to indicate that they should be demuxed inside the
377 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
8eb793c4 378 */
58ee0991 379RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
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380{
381 RTPDemuxContext *s;
382
383 s = av_mallocz(sizeof(RTPDemuxContext));
384 if (!s)
385 return NULL;
386 s->payload_type = payload_type;
387 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
2cab6b48 388 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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389 s->ic = s1;
390 s->st = st;
58ee0991 391 s->queue_size = queue_size;
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392 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
393 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 394 s->ts = ff_mpegts_parse_open(s->ic);
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395 if (s->ts == NULL) {
396 av_free(s);
397 return NULL;
398 }
399 } else {
400 switch(st->codec->codec_id) {
401 case CODEC_ID_MPEG1VIDEO:
402 case CODEC_ID_MPEG2VIDEO:
403 case CODEC_ID_MP2:
404 case CODEC_ID_MP3:
405 case CODEC_ID_MPEG4:
45aa9080 406 case CODEC_ID_H263:
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407 case CODEC_ID_H264:
408 st->need_parsing = AVSTREAM_PARSE_FULL;
409 break;
0048a2a8 410 case CODEC_ID_ADPCM_G722:
0048a2a8
MS
411 /* According to RFC 3551, the stream clock rate is 8000
412 * even if the sample rate is 16000. */
413 if (st->codec->sample_rate == 8000)
414 st->codec->sample_rate = 16000;
415 break;
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416 default:
417 break;
418 }
419 }
420 // needed to send back RTCP RR in RTSP sessions
421 s->rtp_ctx = rtpc;
422 gethostname(s->hostname, sizeof(s->hostname));
423 return s;
424}
425
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426void
427rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
428 RTPDynamicProtocolHandler *handler)
429{
430 s->dynamic_protocol_context = ctx;
431 s->parse_packet = handler->parse_packet;
432}
433
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434/**
435 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
436 */
437static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
438{
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439 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
440 return; /* Timestamp already set by depacketizer */
d74c6145 441 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
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442 int64_t addend;
443 int delta_timestamp;
444
445 /* compute pts from timestamp with received ntp_time */
446 delta_timestamp = timestamp - s->last_rtcp_timestamp;
447 /* convert to the PTS timebase */
2cab6b48 448 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);
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449 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
450 delta_timestamp;
451 return;
fba7815d 452 }
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453 if (timestamp == RTP_NOTS_VALUE)
454 return;
455 if (!s->base_timestamp)
456 s->base_timestamp = timestamp;
457 pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
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458}
459
02607418
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460static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
461 const uint8_t *buf, int len)
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462{
463 unsigned int ssrc, h;
f841a0fc 464 int payload_type, seq, ret, flags = 0;
9446b4bb 465 int ext;
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466 AVStream *st;
467 uint32_t timestamp;
468 int rv= 0;
469
9446b4bb 470 ext = buf[0] & 0x10;
8eb793c4 471 payload_type = buf[1] & 0x7f;
144ae29d
RB
472 if (buf[1] & 0x80)
473 flags |= RTP_FLAG_MARKER;
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474 seq = AV_RB16(buf + 2);
475 timestamp = AV_RB32(buf + 4);
476 ssrc = AV_RB32(buf + 8);
477 /* store the ssrc in the RTPDemuxContext */
478 s->ssrc = ssrc;
479
480 /* NOTE: we can handle only one payload type */
481 if (s->payload_type != payload_type)
482 return -1;
483
484 st = s->st;
485 // only do something with this if all the rtp checks pass...
486 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
487 {
488 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
489 payload_type, seq, ((s->seq + 1) & 0xffff));
490 return -1;
491 }
492
4838cdab
MS
493 if (buf[0] & 0x20) {
494 int padding = buf[len - 1];
495 if (len >= 12 + padding)
496 len -= padding;
497 }
498
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499 s->seq = seq;
500 len -= 12;
501 buf += 12;
502
9446b4bb
RS
503 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
504 if (ext) {
505 if (len < 4)
506 return -1;
507 /* calculate the header extension length (stored as number
508 * of 32-bit words) */
509 ext = (AV_RB16(buf + 2) + 1) << 2;
510
511 if (len < ext)
512 return -1;
513 // skip past RTP header extension
514 len -= ext;
515 buf += ext;
516 }
517
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518 if (!st) {
519 /* specific MPEG2TS demux support */
9125806e 520 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
946df059
MS
521 /* The only error that can be returned from ff_mpegts_parse_packet
522 * is "no more data to return from the provided buffer", so return
523 * AVERROR(EAGAIN) for all errors */
4ffff367 524 if (ret < 0)
946df059 525 return AVERROR(EAGAIN);
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526 if (ret < len) {
527 s->read_buf_size = len - ret;
528 memcpy(s->buf, buf + ret, s->read_buf_size);
529 s->read_buf_index = 0;
530 return 1;
531 }
f3e71942 532 return 0;
b4e3330c 533 } else if (s->parse_packet) {
1a45a9f4 534 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 535 s->st, pkt, &timestamp, buf, len, flags);
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536 } else {
537 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
538 switch(st->codec->codec_id) {
539 case CODEC_ID_MP2:
76faff6e 540 case CODEC_ID_MP3:
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541 /* better than nothing: skip mpeg audio RTP header */
542 if (len <= 4)
543 return -1;
544 h = AV_RB32(buf);
545 len -= 4;
546 buf += 4;
547 av_new_packet(pkt, len);
548 memcpy(pkt->data, buf, len);
549 break;
550 case CODEC_ID_MPEG1VIDEO:
551 case CODEC_ID_MPEG2VIDEO:
552 /* better than nothing: skip mpeg video RTP header */
553 if (len <= 4)
554 return -1;
555 h = AV_RB32(buf);
556 buf += 4;
557 len -= 4;
558 if (h & (1 << 26)) {
559 /* mpeg2 */
560 if (len <= 4)
561 return -1;
562 buf += 4;
563 len -= 4;
564 }
565 av_new_packet(pkt, len);
566 memcpy(pkt->data, buf, len);
567 break;
8eb793c4 568 default:
f739b36d
RB
569 av_new_packet(pkt, len);
570 memcpy(pkt->data, buf, len);
8eb793c4
LA
571 break;
572 }
eafb17d1
RB
573
574 pkt->stream_index = st->index;
f3e71942 575 }
8eb793c4 576
95f03cf3
RB
577 // now perform timestamp things....
578 finalize_packet(s, pkt, timestamp);
f3e71942 579
8eb793c4
LA
580 return rv;
581}
582
58ee0991
MS
583void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
584{
585 while (s->queue) {
586 RTPPacket *next = s->queue->next;
587 av_free(s->queue->buf);
588 av_free(s->queue);
589 s->queue = next;
590 }
591 s->seq = 0;
592 s->queue_len = 0;
593 s->prev_ret = 0;
594}
595
596static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
597{
598 uint16_t seq = AV_RB16(buf + 2);
599 RTPPacket *cur = s->queue, *prev = NULL, *packet;
600
601 /* Find the correct place in the queue to insert the packet */
602 while (cur) {
603 int16_t diff = seq - cur->seq;
604 if (diff < 0)
605 break;
606 prev = cur;
607 cur = cur->next;
608 }
609
610 packet = av_mallocz(sizeof(*packet));
611 if (!packet)
612 return;
613 packet->recvtime = av_gettime();
614 packet->seq = seq;
615 packet->len = len;
616 packet->buf = buf;
617 packet->next = cur;
618 if (prev)
619 prev->next = packet;
620 else
621 s->queue = packet;
622 s->queue_len++;
623}
624
625static int has_next_packet(RTPDemuxContext *s)
626{
ddcf8411 627 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
58ee0991
MS
628}
629
630int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
631{
632 return s->queue ? s->queue->recvtime : 0;
633}
634
635static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
636{
637 int rv;
638 RTPPacket *next;
639
640 if (s->queue_len <= 0)
641 return -1;
642
643 if (!has_next_packet(s))
644 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
645 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
646
647 /* Parse the first packet in the queue, and dequeue it */
648 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
649 next = s->queue->next;
650 av_free(s->queue->buf);
651 av_free(s->queue);
652 s->queue = next;
653 s->queue_len--;
4ffff367 654 return rv;
58ee0991
MS
655}
656
4ffff367 657static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
02607418
MS
658 uint8_t **bufptr, int len)
659{
660 uint8_t* buf = bufptr ? *bufptr : NULL;
661 int ret, flags = 0;
662 uint32_t timestamp;
663 int rv= 0;
664
665 if (!buf) {
f6e138b4
MS
666 /* If parsing of the previous packet actually returned 0 or an error,
667 * there's nothing more to be parsed from that packet, but we may have
58ee0991 668 * indicated that we can return the next enqueued packet. */
f6e138b4 669 if (s->prev_ret <= 0)
58ee0991 670 return rtp_parse_queued_packet(s, pkt);
02607418
MS
671 /* return the next packets, if any */
672 if(s->st && s->parse_packet) {
673 /* timestamp should be overwritten by parse_packet, if not,
674 * the packet is left with pts == AV_NOPTS_VALUE */
675 timestamp = RTP_NOTS_VALUE;
676 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
677 s->st, pkt, &timestamp, NULL, 0, flags);
678 finalize_packet(s, pkt, timestamp);
4ffff367 679 return rv;
02607418
MS
680 } else {
681 // TODO: Move to a dynamic packet handler (like above)
4ffff367 682 if (s->read_buf_index >= s->read_buf_size)
91ec7aea 683 return AVERROR(EAGAIN);
02607418
MS
684 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
685 s->read_buf_size - s->read_buf_index);
4ffff367 686 if (ret < 0)
946df059 687 return AVERROR(EAGAIN);
02607418
MS
688 s->read_buf_index += ret;
689 if (s->read_buf_index < s->read_buf_size)
690 return 1;
4ffff367
MS
691 else
692 return 0;
02607418
MS
693 }
694 }
695
696 if (len < 12)
697 return -1;
698
699 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
700 return -1;
701 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
702 return rtcp_parse_packet(s, buf, len);
703 }
704
65cdee9c 705 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
58ee0991
MS
706 /* First packet, or no reordering */
707 return rtp_parse_packet_internal(s, pkt, buf, len);
708 } else {
709 uint16_t seq = AV_RB16(buf + 2);
710 int16_t diff = seq - s->seq;
711 if (diff < 0) {
712 /* Packet older than the previously emitted one, drop */
713 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
714 "RTP: dropping old packet received too late\n");
715 return -1;
716 } else if (diff <= 1) {
717 /* Correct packet */
718 rv = rtp_parse_packet_internal(s, pkt, buf, len);
4ffff367 719 return rv;
58ee0991
MS
720 } else {
721 /* Still missing some packet, enqueue this one. */
722 enqueue_packet(s, buf, len);
723 *bufptr = NULL;
724 /* Return the first enqueued packet if the queue is full,
725 * even if we're missing something */
726 if (s->queue_len >= s->queue_size)
727 return rtp_parse_queued_packet(s, pkt);
728 return -1;
729 }
730 }
02607418
MS
731}
732
4ffff367
MS
733/**
734 * Parse an RTP or RTCP packet directly sent as a buffer.
735 * @param s RTP parse context.
736 * @param pkt returned packet
737 * @param bufptr pointer to the input buffer or NULL to read the next packets
738 * @param len buffer len
739 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
740 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
741 */
742int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
743 uint8_t **bufptr, int len)
744{
745 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
746 s->prev_ret = rv;
d678a6fd
MS
747 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
748 rv = rtp_parse_queued_packet(s, pkt);
4ffff367
MS
749 return rv ? rv : has_next_packet(s);
750}
751
8eb793c4
LA
752void rtp_parse_close(RTPDemuxContext *s)
753{
58ee0991 754 ff_rtp_reset_packet_queue(s);
8eb793c4 755 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 756 ff_mpegts_parse_close(s->ts);
8eb793c4
LA
757 }
758 av_free(s);
759}
016bc031
JA
760
761int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
762 int (*parse_fmtp)(AVStream *stream,
763 PayloadContext *data,
764 char *attr, char *value))
765{
766 char attr[256];
824535e3 767 char *value;
016bc031 768 int res;
824535e3
JA
769 int value_size = strlen(p) + 1;
770
771 if (!(value = av_malloc(value_size))) {
772 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
773 return AVERROR(ENOMEM);
774 }
016bc031
JA
775
776 // remove protocol identifier
777 while (*p && *p == ' ') p++; // strip spaces
778 while (*p && *p != ' ') p++; // eat protocol identifier
779 while (*p && *p == ' ') p++; // strip trailing spaces
780
781 while (ff_rtsp_next_attr_and_value(&p,
782 attr, sizeof(attr),
824535e3 783 value, value_size)) {
016bc031
JA
784
785 res = parse_fmtp(stream, data, attr, value);
824535e3
JA
786 if (res < 0 && res != AVERROR_PATCHWELCOME) {
787 av_free(value);
016bc031 788 return res;
824535e3 789 }
016bc031 790 }
824535e3 791 av_free(value);
016bc031
JA
792 return 0;
793}