* RTSP/SDP client
* Copyright (c) 2002 Fabrice Bellard
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
#define SDP_MAX_SIZE 16384
#define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
+#define DEFAULT_REORDERING_DELAY 100000
#define OFFSET(x) offsetof(RTSPState, x)
#define DEC AV_OPT_FLAG_DECODING_PARAM
av_freep(content_ptr);
/* If method is set, this is called from ff_rtsp_send_cmd,
* where a reply to exactly this request is awaited. For
- * callers from within packet reciving, we just want to
+ * callers from within packet receiving, we just want to
* return to the caller and go back to receiving packets. */
if (method)
goto start;
{
RTSPState *rt = s->priv_data;
HTTPAuthType cur_auth_type;
- int ret;
+ int ret, attempts = 0;
retry:
cur_auth_type = rt->auth_state.auth_type;
if ((ret = ff_rtsp_read_reply(s, reply, content_ptr, 0, method) ) < 0)
return ret;
+ attempts++;
- if (reply->status_code == 401 && cur_auth_type == HTTP_AUTH_NONE &&
- rt->auth_state.auth_type != HTTP_AUTH_NONE)
+ if (reply->status_code == 401 &&
+ (cur_auth_type == HTTP_AUTH_NONE || rt->auth_state.stale) &&
+ rt->auth_state.auth_type != HTTP_AUTH_NONE && attempts < 2)
goto retry;
if (reply->status_code > 400){
/* default timeout: 1 minute */
rt->timeout = 60;
- /* for each stream, make the setup request */
- /* XXX: we assume the same server is used for the control of each
- * RTSP stream */
-
/* Choose a random starting offset within the first half of the
* port range, to allow for a number of ports to try even if the offset
* happens to be at the end of the random range. */
&s->interrupt_callback, NULL))
goto rtp_opened;
}
-
av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
err = AVERROR(EIO);
goto fail;
break;
}
case RTSP_LOWER_TRANSPORT_UDP_MULTICAST: {
- char url[1024], namebuf[50];
+ char url[1024], namebuf[50], optbuf[20] = "";
struct sockaddr_storage addr;
int port, ttl;
port = rtsp_st->sdp_port;
ttl = rtsp_st->sdp_ttl;
}
+ if (ttl > 0)
+ snprintf(optbuf, sizeof(optbuf), "?ttl=%d", ttl);
getnameinfo((struct sockaddr*) &addr, sizeof(addr),
namebuf, sizeof(namebuf), NULL, 0, NI_NUMERICHOST);
ff_url_join(url, sizeof(url), "rtp", NULL, namebuf,
- port, "?ttl=%d", ttl);
+ port, "%s", optbuf);
if (ffurl_open(&rtsp_st->rtp_handle, url, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, NULL) < 0) {
err = AVERROR_INVALIDDATA;
if (!ff_network_init())
return AVERROR(EIO);
+ if (s->max_delay < 0) /* Not set by the caller */
+ s->max_delay = s->iformat ? DEFAULT_REORDERING_DELAY : 0;
+
rt->control_transport = RTSP_MODE_PLAIN;
if (rt->lower_transport_mask & (1 << RTSP_LOWER_TRANSPORT_HTTP)) {
rt->lower_transport_mask = 1 << RTSP_LOWER_TRANSPORT_TCP;
if (!ff_network_init())
return AVERROR(EIO);
+ if (s->max_delay < 0) /* Not set by the caller */
+ s->max_delay = DEFAULT_REORDERING_DELAY;
+
/* read the whole sdp file */
/* XXX: better loading */
content = av_malloc(SDP_MAX_SIZE);
.read_header = sdp_read_header,
.read_packet = ff_rtsp_fetch_packet,
.read_close = sdp_read_close,
- .priv_class = &sdp_demuxer_class
+ .priv_class = &sdp_demuxer_class,
};
#endif /* CONFIG_SDP_DEMUXER */
.read_header = rtp_read_header,
.read_packet = ff_rtsp_fetch_packet,
.read_close = sdp_read_close,
- .flags = AVFMT_NOFILE,
- .priv_class = &rtp_demuxer_class
+ .flags = AVFMT_NOFILE,
+ .priv_class = &rtp_demuxer_class,
};
#endif /* CONFIG_RTP_DEMUXER */