* RTP input format
* Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-/* needed for gethostname() */
-#define _XOPEN_SOURCE 600
-
+#include "libavutil/mathematics.h"
#include "libavcodec/get_bits.h"
#include "avformat.h"
#include "mpegts.h"
+#include "url.h"
#include <unistd.h>
#include <strings.h>
buffer to 'rtp_write_packet' contains all the packets for ONE
frame. Each packet should have a four byte header containing
the length in big endian format (same trick as
- 'url_open_dyn_packet_buf')
+ 'ffio_open_dyn_packet_buf')
*/
static RTPDynamicProtocolHandler ff_realmedia_mp3_dynamic_handler = {
static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
{
int payload_len;
- while (len >= 2) {
+ while (len >= 4) {
+ payload_len = FFMIN(len, (AV_RB16(buf + 2) + 1) * 4);
+
switch (buf[1]) {
case RTCP_SR:
- if (len < 16) {
+ if (payload_len < 20) {
av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n");
return AVERROR_INVALIDDATA;
}
- payload_len = (AV_RB16(buf + 2) + 1) * 4;
s->last_rtcp_ntp_time = AV_RB64(buf + 8);
s->last_rtcp_timestamp = AV_RB32(buf + 16);
s->rtcp_ts_offset = s->last_rtcp_timestamp - s->base_timestamp;
}
- buf += payload_len;
- len -= payload_len;
break;
case RTCP_BYE:
return -RTCP_BYE;
- default:
- return -1;
}
+
+ buf += payload_len;
+ len -= payload_len;
}
return -1;
}
return 1;
}
-#if 0
-/**
-* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
-* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
-* never change. I left this in in case someone else can see a way. (rdm)
-*/
-static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
-{
- uint32_t transit= arrival_timestamp - sent_timestamp;
- int d;
- s->transit= transit;
- d= FFABS(transit - s->transit);
- s->jitter += d - ((s->jitter + 8)>>4);
-}
-#endif
-
-int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
+int ff_rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
AVIOContext *pb;
uint8_t *buf;
return -1;
s->last_octet_count = s->octet_count;
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_open_dyn_buf(&pb) < 0)
return -1;
// Receiver Report
}
avio_flush(pb);
- len = url_close_dyn_buf(pb, &buf);
+ len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf) {
- int result;
+ int av_unused result;
av_dlog(s->ic, "sending %d bytes of RR\n", len);
- result= url_write(s->rtp_ctx, buf, len);
- av_dlog(s->ic, "result from url_write: %d\n", result);
+ result= ffurl_write(s->rtp_ctx, buf, len);
+ av_dlog(s->ic, "result from ffurl_write: %d\n", result);
av_free(buf);
}
return 0;
}
-void rtp_send_punch_packets(URLContext* rtp_handle)
+void ff_rtp_send_punch_packets(URLContext* rtp_handle)
{
AVIOContext *pb;
uint8_t *buf;
int len;
/* Send a small RTP packet */
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_open_dyn_buf(&pb) < 0)
return;
avio_w8(pb, (RTP_VERSION << 6));
avio_wb32(pb, 0); /* SSRC */
avio_flush(pb);
- len = url_close_dyn_buf(pb, &buf);
+ len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
- url_write(rtp_handle, buf, len);
+ ffurl_write(rtp_handle, buf, len);
av_free(buf);
/* Send a minimal RTCP RR */
- if (url_open_dyn_buf(&pb) < 0)
+ if (avio_open_dyn_buf(&pb) < 0)
return;
avio_w8(pb, (RTP_VERSION << 6));
avio_wb32(pb, 0); /* our own SSRC */
avio_flush(pb);
- len = url_close_dyn_buf(pb, &buf);
+ len = avio_close_dyn_buf(pb, &buf);
if ((len > 0) && buf)
- url_write(rtp_handle, buf, len);
+ ffurl_write(rtp_handle, buf, len);
av_free(buf);
}
* MPEG2TS streams to indicate that they should be demuxed inside the
* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
*/
-RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
+RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size)
{
RTPDemuxContext *s;
}
void
-rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
- RTPDynamicProtocolHandler *handler)
+ff_rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx,
+ RTPDynamicProtocolHandler *handler)
{
s->dynamic_protocol_context = ctx;
s->parse_packet = handler->parse_packet;
* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
*/
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
- uint8_t **bufptr, int len)
+int ff_rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+ uint8_t **bufptr, int len)
{
int rv = rtp_parse_one_packet(s, pkt, bufptr, len);
s->prev_ret = rv;
return rv ? rv : has_next_packet(s);
}
-void rtp_parse_close(RTPDemuxContext *s)
+void ff_rtp_parse_close(RTPDemuxContext *s)
{
ff_rtp_reset_packet_queue(s);
if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
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