Mercurial > libavformat.hg
comparison rtp.c @ 1425:00d9393a126f libavformat
make ffmpeg able to send back a RTCP receiver report.
Patch by Thijs thijsvermeir A telenet P be
Original thread:
Date: Oct 27, 2006 12:58 PM
Subject: [Ffmpeg-devel] [PATCH proposal] RTCP receiver report
author | gpoirier |
---|---|
date | Fri, 27 Oct 2006 18:19:29 +0000 |
parents | 1c39ce5c6a5d |
children | f1614c754d5b |
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1424:1c39ce5c6a5d | 1425:00d9393a126f |
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257 s->last_rtcp_timestamp = decode_be32(buf + 16); | 257 s->last_rtcp_timestamp = decode_be32(buf + 16); |
258 return 0; | 258 return 0; |
259 } | 259 } |
260 | 260 |
261 /** | 261 /** |
262 * some rtp servers assume client is dead if they don't hear from them... | |
263 * so we send a Receiver Report to the provided ByteIO context | |
264 * (we don't have access to the rtcp handle from here) | |
265 */ | |
266 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count) | |
267 { | |
268 ByteIOContext pb; | |
269 uint8_t *buf; | |
270 int len; | |
271 int rtcp_bytes; | |
272 | |
273 if (!s->rtp_ctx || (count < 1)) | |
274 return -1; | |
275 | |
276 /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ | |
277 s->octet_count += count; | |
278 rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / | |
279 RTCP_TX_RATIO_DEN; | |
280 rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? | |
281 if (rtcp_bytes < 28) | |
282 return -1; | |
283 s->last_octet_count = s->octet_count; | |
284 | |
285 if (url_open_dyn_buf(&pb) < 0) | |
286 return -1; | |
287 | |
288 // Receiver Report | |
289 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ | |
290 put_byte(&pb, 201); | |
291 put_be16(&pb, 7); /* length in words - 1 */ | |
292 put_be32(&pb, s->ssrc); // our own SSRC | |
293 put_be32(&pb, s->ssrc); // XXX: should be the server's here! | |
294 // some placeholders we should really fill... | |
295 put_be32(&pb, ((0 << 24) | (0 & 0x0ffffff))); /* 0% lost, total 0 lost */ | |
296 put_be32(&pb, (0 << 16) | s->seq); | |
297 put_be32(&pb, 0x68); /* jitter */ | |
298 put_be32(&pb, -1); /* last SR timestamp */ | |
299 put_be32(&pb, 1); /* delay since last SR */ | |
300 | |
301 // CNAME | |
302 put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */ | |
303 put_byte(&pb, 202); | |
304 len = strlen(s->hostname); | |
305 put_be16(&pb, (6 + len + 3) / 4); /* length in words - 1 */ | |
306 put_be32(&pb, s->ssrc); | |
307 put_byte(&pb, 0x01); | |
308 put_byte(&pb, len); | |
309 put_buffer(&pb, s->hostname, len); | |
310 // padding | |
311 for (len = (6 + len) % 4; len % 4; len++) { | |
312 put_byte(&pb, 0); | |
313 } | |
314 | |
315 put_flush_packet(&pb); | |
316 len = url_close_dyn_buf(&pb, &buf); | |
317 if ((len > 0) && buf) { | |
318 #if defined(DEBUG) | |
319 printf("sending %d bytes of RR\n", len); | |
320 #endif | |
321 url_write(s->rtp_ctx, buf, len); | |
322 av_free(buf); | |
323 } | |
324 return 0; | |
325 } | |
326 | |
327 /** | |
262 * open a new RTP parse context for stream 'st'. 'st' can be NULL for | 328 * open a new RTP parse context for stream 'st'. 'st' can be NULL for |
263 * MPEG2TS streams to indicate that they should be demuxed inside the | 329 * MPEG2TS streams to indicate that they should be demuxed inside the |
264 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) | 330 * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) |
265 * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. | |
266 */ | 331 */ |
267 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data) | 332 RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data) |
268 { | 333 { |
269 RTPDemuxContext *s; | 334 RTPDemuxContext *s; |
270 | 335 |
271 s = av_mallocz(sizeof(RTPDemuxContext)); | 336 s = av_mallocz(sizeof(RTPDemuxContext)); |
272 if (!s) | 337 if (!s) |
297 break; | 362 break; |
298 default: | 363 default: |
299 break; | 364 break; |
300 } | 365 } |
301 } | 366 } |
367 // needed to send back RTCP RR in RTSP sessions | |
368 s->rtp_ctx = rtpc; | |
369 gethostname(s->hostname, sizeof(s->hostname)); | |
302 return s; | 370 return s; |
303 } | 371 } |
304 | 372 |
305 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) | 373 static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
306 { | 374 { |
397 } | 465 } |
398 payload_type = buf[1] & 0x7f; | 466 payload_type = buf[1] & 0x7f; |
399 seq = (buf[2] << 8) | buf[3]; | 467 seq = (buf[2] << 8) | buf[3]; |
400 timestamp = decode_be32(buf + 4); | 468 timestamp = decode_be32(buf + 4); |
401 ssrc = decode_be32(buf + 8); | 469 ssrc = decode_be32(buf + 8); |
470 /* store the ssrc in the RTPDemuxContext */ | |
471 s->ssrc = ssrc; | |
402 | 472 |
403 /* NOTE: we can handle only one payload type */ | 473 /* NOTE: we can handle only one payload type */ |
404 if (s->payload_type != payload_type) | 474 if (s->payload_type != payload_type) |
405 return -1; | 475 return -1; |
406 | 476 |