h264: change unsupported bit depth message to error level
[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"
9106a698 23#include "libavcodec/get_bits.h"
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24#include "avformat.h"
25#include "mpegts.h"
925e908b 26#include "url.h"
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27
28#include <unistd.h>
1e515c42 29#include <strings.h>
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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 = {
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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{
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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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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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67 ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler);
68 ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler);
0369d2b0 69 ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler);
e6327fba 70 ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler);
887af2aa 71 ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler);
a59096e4 72 ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler);
4449df6b 73 ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler);
1ddc176e 74 ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler);
51291e60 75 ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler);
35014efc 76 ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
2eeefe20 77 ff_register_dynamic_payload_handler(&ff_realmedia_mp3_dynamic_handler);
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78
79 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
80 ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);
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81
82 ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler);
83 ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler);
84 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler);
85 ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler);
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86}
87
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88RTPDynamicProtocolHandler *ff_rtp_handler_find_by_name(const char *name,
89 enum AVMediaType codec_type)
90{
91 RTPDynamicProtocolHandler *handler;
92 for (handler = RTPFirstDynamicPayloadHandler;
93 handler; handler = handler->next)
94 if (!strcasecmp(name, handler->enc_name) &&
95 codec_type == handler->codec_type)
96 return handler;
97 return NULL;
98}
99
100RTPDynamicProtocolHandler *ff_rtp_handler_find_by_id(int id,
101 enum AVMediaType codec_type)
102{
103 RTPDynamicProtocolHandler *handler;
104 for (handler = RTPFirstDynamicPayloadHandler;
105 handler; handler = handler->next)
106 if (handler->static_payload_id && handler->static_payload_id == id &&
107 codec_type == handler->codec_type)
108 return handler;
109 return NULL;
110}
111
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112static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
113{
ff328c02 114 int payload_len;
07b77fe3
JB
115 while (len >= 4) {
116 payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
117
ff328c02
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118 switch (buf[1]) {
119 case RTCP_SR:
07b77fe3 120 if (payload_len < 20) {
ff328c02
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121 av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
122 return AVERROR_INVALIDDATA;
123 }
ff328c02 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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MS
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 }
ff328c02 133
ff328c02 134 break;
b20359f5
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135 case RTCP_BYE:
136 return -RTCP_BYE;
ff328c02 137 }
07b77fe3
JB
138
139 buf += payload_len;
140 len -= payload_len;
ff328c02 141 }
b20359f5 142 return -1;
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143}
144
145#define RTP_SEQ_MOD (1<<16)
146
147/**
148* called on parse open packet
149*/
150static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
151{
152 memset(s, 0, sizeof(RTPStatistics));
153 s->max_seq= base_sequence;
154 s->probation= 1;
155}
156
157/**
158* called whenever there is a large jump in sequence numbers, or when they get out of probation...
159*/
160static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
161{
162 s->max_seq= seq;
163 s->cycles= 0;
164 s->base_seq= seq -1;
165 s->bad_seq= RTP_SEQ_MOD + 1;
166 s->received= 0;
167 s->expected_prior= 0;
168 s->received_prior= 0;
169 s->jitter= 0;
170 s->transit= 0;
171}
172
173/**
174* returns 1 if we should handle this packet.
175*/
176static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
177{
178 uint16_t udelta= seq - s->max_seq;
179 const int MAX_DROPOUT= 3000;
180 const int MAX_MISORDER = 100;
181 const int MIN_SEQUENTIAL = 2;
182
183 /* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
184 if(s->probation)
185 {
186 if(seq==s->max_seq + 1) {
187 s->probation--;
188 s->max_seq= seq;
189 if(s->probation==0) {
190 rtp_init_sequence(s, seq);
191 s->received++;
192 return 1;
193 }
194 } else {
195 s->probation= MIN_SEQUENTIAL - 1;
196 s->max_seq = seq;
197 }
198 } else if (udelta < MAX_DROPOUT) {
199 // in order, with permissible gap
200 if(seq < s->max_seq) {
201 //sequence number wrapped; count antother 64k cycles
202 s->cycles += RTP_SEQ_MOD;
203 }
204 s->max_seq= seq;
205 } else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
206 // sequence made a large jump...
207 if(seq==s->bad_seq) {
208 // two sequential packets-- assume that the other side restarted without telling us; just resync.
209 rtp_init_sequence(s, seq);
210 } else {
211 s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
212 return 0;
213 }
214 } else {
215 // duplicate or reordered packet...
216 }
217 s->received++;
218 return 1;
219}
220
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221int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
222{
ae628ec1 223 AVIOContext *pb;
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224 uint8_t *buf;
225 int len;
226 int rtcp_bytes;
227 RTPStatistics *stats= &s->statistics;
228 uint32_t lost;
229 uint32_t extended_max;
230 uint32_t expected_interval;
231 uint32_t received_interval;
232 uint32_t lost_interval;
233 uint32_t expected;
234 uint32_t fraction;
235 uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
236
237 if (!s->rtp_ctx || (count < 1))
238 return -1;
239
240 /* TODO: I think this is way too often; RFC 1889 has algorithm for this */
241 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
242 s->octet_count += count;
243 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
244 RTCP_TX_RATIO_DEN;
245 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
246 if (rtcp_bytes < 28)
247 return -1;
248 s->last_octet_count = s->octet_count;
249
b92c5452 250 if (avio_open_dyn_buf(&pb) < 0)
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251 return -1;
252
253 // Receiver Report
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254 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
255 avio_w8(pb, RTCP_RR);
256 avio_wb16(pb, 7); /* length in words - 1 */
952139a3 257 // our own SSRC: we use the server's SSRC + 1 to avoid conflicts
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258 avio_wb32(pb, s->ssrc + 1);
259 avio_wb32(pb, s->ssrc); // server SSRC
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260 // some placeholders we should really fill...
261 // RFC 1889/p64
262 extended_max= stats->cycles + stats->max_seq;
263 expected= extended_max - stats->base_seq + 1;
264 lost= expected - stats->received;
265 lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
266 expected_interval= expected - stats->expected_prior;
267 stats->expected_prior= expected;
268 received_interval= stats->received - stats->received_prior;
269 stats->received_prior= stats->received;
270 lost_interval= expected_interval - received_interval;
271 if (expected_interval==0 || lost_interval<=0) fraction= 0;
272 else fraction = (lost_interval<<8)/expected_interval;
273
274 fraction= (fraction<<24) | lost;
275
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276 avio_wb32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
277 avio_wb32(pb, extended_max); /* max sequence received */
278 avio_wb32(pb, stats->jitter>>4); /* jitter */
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279
280 if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
281 {
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282 avio_wb32(pb, 0); /* last SR timestamp */
283 avio_wb32(pb, 0); /* delay since last SR */
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284 } else {
285 uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
286 uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
287
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288 avio_wb32(pb, middle_32_bits); /* last SR timestamp */
289 avio_wb32(pb, delay_since_last); /* delay since last SR */
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290 }
291
292 // CNAME
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293 avio_w8(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
294 avio_w8(pb, RTCP_SDES);
8eb793c4 295 len = strlen(s->hostname);
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296 avio_wb16(pb, (6 + len + 3) / 4); /* length in words - 1 */
297 avio_wb32(pb, s->ssrc);
298 avio_w8(pb, 0x01);
299 avio_w8(pb, len);
300 avio_write(pb, s->hostname, len);
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301 // padding
302 for (len = (6 + len) % 4; len % 4; len++) {
77eb5504 303 avio_w8(pb, 0);
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304 }
305
b7f2fdde 306 avio_flush(pb);
6dc7d80d 307 len = avio_close_dyn_buf(pb, &buf);
8eb793c4 308 if ((len > 0) && buf) {
5e1166b3 309 int av_unused result;
dfd2a005 310 av_dlog(s->ic, "sending %d bytes of RR\n", len);
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AK
311 result= ffurl_write(s->rtp_ctx, buf, len);
312 av_dlog(s->ic, "result from ffurl_write: %d\n", result);
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313 av_free(buf);
314 }
315 return 0;
316}
317
9c8fa20d
MS
318void rtp_send_punch_packets(URLContext* rtp_handle)
319{
ae628ec1 320 AVIOContext *pb;
9c8fa20d
MS
321 uint8_t *buf;
322 int len;
323
324 /* Send a small RTP packet */
b92c5452 325 if (avio_open_dyn_buf(&pb) < 0)
9c8fa20d
MS
326 return;
327
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AK
328 avio_w8(pb, (RTP_VERSION << 6));
329 avio_w8(pb, 0); /* Payload type */
330 avio_wb16(pb, 0); /* Seq */
331 avio_wb32(pb, 0); /* Timestamp */
332 avio_wb32(pb, 0); /* SSRC */
9c8fa20d 333
b7f2fdde 334 avio_flush(pb);
6dc7d80d 335 len = avio_close_dyn_buf(pb, &buf);
9c8fa20d 336 if ((len > 0) && buf)
925e908b 337 ffurl_write(rtp_handle, buf, len);
9c8fa20d
MS
338 av_free(buf);
339
340 /* Send a minimal RTCP RR */
b92c5452 341 if (avio_open_dyn_buf(&pb) < 0)
9c8fa20d
MS
342 return;
343
77eb5504
AK
344 avio_w8(pb, (RTP_VERSION << 6));
345 avio_w8(pb, RTCP_RR); /* receiver report */
346 avio_wb16(pb, 1); /* length in words - 1 */
347 avio_wb32(pb, 0); /* our own SSRC */
9c8fa20d 348
b7f2fdde 349 avio_flush(pb);
6dc7d80d 350 len = avio_close_dyn_buf(pb, &buf);
9c8fa20d 351 if ((len > 0) && buf)
925e908b 352 ffurl_write(rtp_handle, buf, len);
9c8fa20d
MS
353 av_free(buf);
354}
355
356
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357/**
358 * open a new RTP parse context for stream 'st'. 'st' can be NULL for
359 * MPEG2TS streams to indicate that they should be demuxed inside the
360 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
8eb793c4 361 */
58ee0991 362RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
8eb793c4
LA
363{
364 RTPDemuxContext *s;
365
366 s = av_mallocz(sizeof(RTPDemuxContext));
367 if (!s)
368 return NULL;
369 s->payload_type = payload_type;
370 s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
2cab6b48 371 s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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372 s->ic = s1;
373 s->st = st;
58ee0991 374 s->queue_size = queue_size;
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375 rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
376 if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
9125806e 377 s->ts = ff_mpegts_parse_open(s->ic);
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378 if (s->ts == NULL) {
379 av_free(s);
380 return NULL;
381 }
382 } else {
383 switch(st->codec->codec_id) {
384 case CODEC_ID_MPEG1VIDEO:
385 case CODEC_ID_MPEG2VIDEO:
386 case CODEC_ID_MP2:
387 case CODEC_ID_MP3:
388 case CODEC_ID_MPEG4:
45aa9080 389 case CODEC_ID_H263:
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390 case CODEC_ID_H264:
391 st->need_parsing = AVSTREAM_PARSE_FULL;
392 break;
0048a2a8 393 case CODEC_ID_ADPCM_G722:
0048a2a8
MS
394 /* According to RFC 3551, the stream clock rate is 8000
395 * even if the sample rate is 16000. */
396 if (st->codec->sample_rate == 8000)
397 st->codec->sample_rate = 16000;
398 break;
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399 default:
400 break;
401 }
402 }
403 // needed to send back RTCP RR in RTSP sessions
404 s->rtp_ctx = rtpc;
405 gethostname(s->hostname, sizeof(s->hostname));
406 return s;
407}
408
99a1d191
RB
409void
410rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
411 RTPDynamicProtocolHandler *handler)
412{
413 s->dynamic_protocol_context = ctx;
414 s->parse_packet = handler->parse_packet;
415}
416
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417/**
418 * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
419 */
420static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
421{
79d482b1
MS
422 if (pkt->pts != AV_NOPTS_VALUE || pkt->dts != AV_NOPTS_VALUE)
423 return; /* Timestamp already set by depacketizer */
d74c6145 424 if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) {
fba7815d
LA
425 int64_t addend;
426 int delta_timestamp;
427
428 /* compute pts from timestamp with received ntp_time */
429 delta_timestamp = timestamp - s->last_rtcp_timestamp;
430 /* convert to the PTS timebase */
2cab6b48 431 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
432 pkt->pts = s->range_start_offset + s->rtcp_ts_offset + addend +
433 delta_timestamp;
434 return;
fba7815d 435 }
3a1cdcc7
MS
436 if (timestamp == RTP_NOTS_VALUE)
437 return;
438 if (!s->base_timestamp)
439 s->base_timestamp = timestamp;
440 pkt->pts = s->range_start_offset + timestamp - s->base_timestamp;
8eb793c4
LA
441}
442
02607418
MS
443static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
444 const uint8_t *buf, int len)
8eb793c4
LA
445{
446 unsigned int ssrc, h;
f841a0fc 447 int payload_type, seq, ret, flags = 0;
9446b4bb 448 int ext;
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LA
449 AVStream *st;
450 uint32_t timestamp;
451 int rv= 0;
452
9446b4bb 453 ext = buf[0] & 0x10;
8eb793c4 454 payload_type = buf[1] & 0x7f;
144ae29d
RB
455 if (buf[1] & 0x80)
456 flags |= RTP_FLAG_MARKER;
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LA
457 seq = AV_RB16(buf + 2);
458 timestamp = AV_RB32(buf + 4);
459 ssrc = AV_RB32(buf + 8);
460 /* store the ssrc in the RTPDemuxContext */
461 s->ssrc = ssrc;
462
463 /* NOTE: we can handle only one payload type */
464 if (s->payload_type != payload_type)
465 return -1;
466
467 st = s->st;
468 // only do something with this if all the rtp checks pass...
469 if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
470 {
471 av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
472 payload_type, seq, ((s->seq + 1) & 0xffff));
473 return -1;
474 }
475
4838cdab
MS
476 if (buf[0] & 0x20) {
477 int padding = buf[len - 1];
478 if (len >= 12 + padding)
479 len -= padding;
480 }
481
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LA
482 s->seq = seq;
483 len -= 12;
484 buf += 12;
485
9446b4bb
RS
486 /* RFC 3550 Section 5.3.1 RTP Header Extension handling */
487 if (ext) {
488 if (len < 4)
489 return -1;
490 /* calculate the header extension length (stored as number
491 * of 32-bit words) */
492 ext = (AV_RB16(buf + 2) + 1) << 2;
493
494 if (len < ext)
495 return -1;
496 // skip past RTP header extension
497 len -= ext;
498 buf += ext;
499 }
500
8eb793c4
LA
501 if (!st) {
502 /* specific MPEG2TS demux support */
9125806e 503 ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len);
946df059
MS
504 /* The only error that can be returned from ff_mpegts_parse_packet
505 * is "no more data to return from the provided buffer", so return
506 * AVERROR(EAGAIN) for all errors */
4ffff367 507 if (ret < 0)
946df059 508 return AVERROR(EAGAIN);
8eb793c4
LA
509 if (ret < len) {
510 s->read_buf_size = len - ret;
511 memcpy(s->buf, buf + ret, s->read_buf_size);
512 s->read_buf_index = 0;
513 return 1;
514 }
f3e71942 515 return 0;
b4e3330c 516 } else if (s->parse_packet) {
1a45a9f4 517 rv = s->parse_packet(s->ic, s->dynamic_protocol_context,
9b932b8a 518 s->st, pkt, &timestamp, buf, len, flags);
8eb793c4
LA
519 } else {
520 // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
521 switch(st->codec->codec_id) {
522 case CODEC_ID_MP2:
76faff6e 523 case CODEC_ID_MP3:
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LA
524 /* better than nothing: skip mpeg audio RTP header */
525 if (len <= 4)
526 return -1;
527 h = AV_RB32(buf);
528 len -= 4;
529 buf += 4;
530 av_new_packet(pkt, len);
531 memcpy(pkt->data, buf, len);
532 break;
533 case CODEC_ID_MPEG1VIDEO:
534 case CODEC_ID_MPEG2VIDEO:
535 /* better than nothing: skip mpeg video RTP header */
536 if (len <= 4)
537 return -1;
538 h = AV_RB32(buf);
539 buf += 4;
540 len -= 4;
541 if (h & (1 << 26)) {
542 /* mpeg2 */
543 if (len <= 4)
544 return -1;
545 buf += 4;
546 len -= 4;
547 }
548 av_new_packet(pkt, len);
549 memcpy(pkt->data, buf, len);
550 break;
8eb793c4 551 default:
f739b36d
RB
552 av_new_packet(pkt, len);
553 memcpy(pkt->data, buf, len);
8eb793c4
LA
554 break;
555 }
eafb17d1
RB
556
557 pkt->stream_index = st->index;
f3e71942 558 }
8eb793c4 559
95f03cf3
RB
560 // now perform timestamp things....
561 finalize_packet(s, pkt, timestamp);
f3e71942 562
8eb793c4
LA
563 return rv;
564}
565
58ee0991
MS
566void ff_rtp_reset_packet_queue(RTPDemuxContext *s)
567{
568 while (s->queue) {
569 RTPPacket *next = s->queue->next;
570 av_free(s->queue->buf);
571 av_free(s->queue);
572 s->queue = next;
573 }
574 s->seq = 0;
575 s->queue_len = 0;
576 s->prev_ret = 0;
577}
578
579static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len)
580{
581 uint16_t seq = AV_RB16(buf + 2);
582 RTPPacket *cur = s->queue, *prev = NULL, *packet;
583
584 /* Find the correct place in the queue to insert the packet */
585 while (cur) {
586 int16_t diff = seq - cur->seq;
587 if (diff < 0)
588 break;
589 prev = cur;
590 cur = cur->next;
591 }
592
593 packet = av_mallocz(sizeof(*packet));
594 if (!packet)
595 return;
596 packet->recvtime = av_gettime();
597 packet->seq = seq;
598 packet->len = len;
599 packet->buf = buf;
600 packet->next = cur;
601 if (prev)
602 prev->next = packet;
603 else
604 s->queue = packet;
605 s->queue_len++;
606}
607
608static int has_next_packet(RTPDemuxContext *s)
609{
ddcf8411 610 return s->queue && s->queue->seq == (uint16_t) (s->seq + 1);
58ee0991
MS
611}
612
613int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s)
614{
615 return s->queue ? s->queue->recvtime : 0;
616}
617
618static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt)
619{
620 int rv;
621 RTPPacket *next;
622
623 if (s->queue_len <= 0)
624 return -1;
625
626 if (!has_next_packet(s))
627 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
628 "RTP: missed %d packets\n", s->queue->seq - s->seq - 1);
629
630 /* Parse the first packet in the queue, and dequeue it */
631 rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len);
632 next = s->queue->next;
633 av_free(s->queue->buf);
634 av_free(s->queue);
635 s->queue = next;
636 s->queue_len--;
4ffff367 637 return rv;
58ee0991
MS
638}
639
4ffff367 640static int rtp_parse_one_packet(RTPDemuxContext *s, AVPacket *pkt,
02607418
MS
641 uint8_t **bufptr, int len)
642{
643 uint8_t* buf = bufptr ? *bufptr : NULL;
644 int ret, flags = 0;
645 uint32_t timestamp;
646 int rv= 0;
647
648 if (!buf) {
f6e138b4
MS
649 /* If parsing of the previous packet actually returned 0 or an error,
650 * there's nothing more to be parsed from that packet, but we may have
58ee0991 651 * indicated that we can return the next enqueued packet. */
f6e138b4 652 if (s->prev_ret <= 0)
58ee0991 653 return rtp_parse_queued_packet(s, pkt);
02607418
MS
654 /* return the next packets, if any */
655 if(s->st && s->parse_packet) {
656 /* timestamp should be overwritten by parse_packet, if not,
657 * the packet is left with pts == AV_NOPTS_VALUE */
658 timestamp = RTP_NOTS_VALUE;
659 rv= s->parse_packet(s->ic, s->dynamic_protocol_context,
660 s->st, pkt, &timestamp, NULL, 0, flags);
661 finalize_packet(s, pkt, timestamp);
4ffff367 662 return rv;
02607418
MS
663 } else {
664 // TODO: Move to a dynamic packet handler (like above)
4ffff367 665 if (s->read_buf_index >= s->read_buf_size)
91ec7aea 666 return AVERROR(EAGAIN);
02607418
MS
667 ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
668 s->read_buf_size - s->read_buf_index);
4ffff367 669 if (ret < 0)
946df059 670 return AVERROR(EAGAIN);
02607418
MS
671 s->read_buf_index += ret;
672 if (s->read_buf_index < s->read_buf_size)
673 return 1;
4ffff367
MS
674 else
675 return 0;
02607418
MS
676 }
677 }
678
679 if (len < 12)
680 return -1;
681
682 if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
683 return -1;
684 if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) {
685 return rtcp_parse_packet(s, buf, len);
686 }
687
65cdee9c 688 if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
58ee0991
MS
689 /* First packet, or no reordering */
690 return rtp_parse_packet_internal(s, pkt, buf, len);
691 } else {
692 uint16_t seq = AV_RB16(buf + 2);
693 int16_t diff = seq - s->seq;
694 if (diff < 0) {
695 /* Packet older than the previously emitted one, drop */
696 av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING,
697 "RTP: dropping old packet received too late\n");
698 return -1;
699 } else if (diff <= 1) {
700 /* Correct packet */
701 rv = rtp_parse_packet_internal(s, pkt, buf, len);
4ffff367 702 return rv;
58ee0991
MS
703 } else {
704 /* Still missing some packet, enqueue this one. */
705 enqueue_packet(s, buf, len);
706 *bufptr = NULL;
707 /* Return the first enqueued packet if the queue is full,
708 * even if we're missing something */
709 if (s->queue_len >= s->queue_size)
710 return rtp_parse_queued_packet(s, pkt);
711 return -1;
712 }
713 }
02607418
MS
714}
715
4ffff367
MS
716/**
717 * Parse an RTP or RTCP packet directly sent as a buffer.
718 * @param s RTP parse context.
719 * @param pkt returned packet
720 * @param bufptr pointer to the input buffer or NULL to read the next packets
721 * @param len buffer len
722 * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
723 * (use buf as NULL to read the next). -1 if no packet (error or no more packet).
724 */
725int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
726 uint8_t **bufptr, int len)
727{
728 int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
729 s->prev_ret = rv;
d678a6fd
MS
730 while (rv == AVERROR(EAGAIN) && has_next_packet(s))
731 rv = rtp_parse_queued_packet(s, pkt);
4ffff367
MS
732 return rv ? rv : has_next_packet(s);
733}
734
8eb793c4
LA
735void rtp_parse_close(RTPDemuxContext *s)
736{
58ee0991 737 ff_rtp_reset_packet_queue(s);
8eb793c4 738 if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
9125806e 739 ff_mpegts_parse_close(s->ts);
8eb793c4
LA
740 }
741 av_free(s);
742}
016bc031
JA
743
744int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p,
745 int (*parse_fmtp)(AVStream *stream,
746 PayloadContext *data,
747 char *attr, char *value))
748{
749 char attr[256];
824535e3 750 char *value;
016bc031 751 int res;
824535e3
JA
752 int value_size = strlen(p) + 1;
753
754 if (!(value = av_malloc(value_size))) {
755 av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP.");
756 return AVERROR(ENOMEM);
757 }
016bc031
JA
758
759 // remove protocol identifier
760 while (*p && *p == ' ') p++; // strip spaces
761 while (*p && *p != ' ') p++; // eat protocol identifier
762 while (*p && *p == ' ') p++; // strip trailing spaces
763
764 while (ff_rtsp_next_attr_and_value(&p,
765 attr, sizeof(attr),
824535e3 766 value, value_size)) {
016bc031
JA
767
768 res = parse_fmtp(stream, data, attr, value);
824535e3
JA
769 if (res < 0 && res != AVERROR_PATCHWELCOME) {
770 av_free(value);
016bc031 771 return res;
824535e3 772 }
016bc031 773 }
824535e3 774 av_free(value);
016bc031
JA
775 return 0;
776}