* RTSP demuxer
* 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
*/
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
+#include "libavutil/mathematics.h"
#include "avformat.h"
#include "internal.h"
#include "os_support.h"
#include "rtsp.h"
#include "rdt.h"
-
-//#define DEBUG
-//#define DEBUG_RTP_TCP
+#include "url.h"
static int rtsp_read_play(AVFormatContext *s)
{
rtpctx->last_rtcp_ntp_time = AV_NOPTS_VALUE;
rtpctx->first_rtcp_ntp_time = AV_NOPTS_VALUE;
rtpctx->base_timestamp = 0;
+ rtpctx->timestamp = 0;
+ rtpctx->unwrapped_timestamp = 0;
rtpctx->rtcp_ts_offset = 0;
}
}
cmd[0] = 0;
} else {
snprintf(cmd, sizeof(cmd),
- "Range: npt=%0.3f-\r\n",
- (double)rt->seek_timestamp / AV_TIME_BASE);
+ "Range: npt=%"PRId64".%03"PRId64"-\r\n",
+ rt->seek_timestamp / AV_TIME_BASE,
+ rt->seek_timestamp / (AV_TIME_BASE / 1000) % 1000);
}
ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return 0;
}
-static int rtsp_read_header(AVFormatContext *s,
- AVFormatParameters *ap)
+static int rtsp_read_header(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
int ret;
if (ret)
return ret;
- rt->real_setup_cache = av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
- if (!rt->real_setup_cache)
+ rt->real_setup_cache = !s->nb_streams ? NULL :
+ av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
+ if (!rt->real_setup_cache && s->nb_streams)
return AVERROR(ENOMEM);
rt->real_setup = rt->real_setup_cache + s->nb_streams;
- if (ap->initial_pause) {
+ if (rt->initial_pause) {
/* do not start immediately */
} else {
if (rtsp_read_play(s) < 0) {
int id, len, i, ret;
RTSPStream *rtsp_st;
-#ifdef DEBUG_RTP_TCP
av_dlog(s, "tcp_read_packet:\n");
-#endif
redo:
for (;;) {
RTSPMessageHeader reply;
if (rt->state != RTSP_STATE_STREAMING)
return 0;
}
- ret = url_read_complete(rt->rtsp_hd, buf, 3);
+ ret = ffurl_read_complete(rt->rtsp_hd, buf, 3);
if (ret != 3)
return -1;
id = buf[0];
len = AV_RB16(buf + 1);
-#ifdef DEBUG_RTP_TCP
av_dlog(s, "id=%d len=%d\n", id, len);
-#endif
if (len > buf_size || len < 12)
goto redo;
/* get the data */
- ret = url_read_complete(rt->rtsp_hd, buf, len);
+ ret = ffurl_read_complete(rt->rtsp_hd, buf, len);
if (ret != len)
return -1;
if (rt->transport == RTSP_TRANSPORT_RDT &&
rt->packets++;
/* send dummy request to keep TCP connection alive */
- if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2) {
- if (rt->server_type == RTSP_SERVER_WMS) {
+ if ((av_gettime() - rt->last_cmd_time) / 1000000 >= rt->timeout / 2 ||
+ rt->auth_state.stale) {
+ if (rt->server_type == RTSP_SERVER_WMS ||
+ (rt->server_type != RTSP_SERVER_REAL &&
+ rt->get_parameter_supported)) {
ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL);
} else {
ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL);
}
+ /* The stale flag should be reset when creating the auth response in
+ * ff_rtsp_send_cmd_async, but reset it here just in case we never
+ * called the auth code (if we didn't have any credentials set). */
+ rt->auth_state.stale = 0;
}
return 0;
{
RTSPState *rt = s->priv_data;
-#if 0
- /* NOTE: it is valid to flush the buffer here */
- if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP) {
- url_fclose(&rt->rtsp_gb);
- }
-#endif
ff_rtsp_send_cmd_async(s, "TEARDOWN", rt->control_uri, NULL);
ff_rtsp_close_streams(s);
return 0;
}
+static const AVClass rtsp_demuxer_class = {
+ .class_name = "RTSP demuxer",
+ .item_name = av_default_item_name,
+ .option = ff_rtsp_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
AVInputFormat ff_rtsp_demuxer = {
- "rtsp",
- NULL_IF_CONFIG_SMALL("RTSP input format"),
- sizeof(RTSPState),
- rtsp_probe,
- rtsp_read_header,
- rtsp_read_packet,
- rtsp_read_close,
- rtsp_read_seek,
- .flags = AVFMT_NOFILE,
- .read_play = rtsp_read_play,
- .read_pause = rtsp_read_pause,
+ .name = "rtsp",
+ .long_name = NULL_IF_CONFIG_SMALL("RTSP input format"),
+ .priv_data_size = sizeof(RTSPState),
+ .read_probe = rtsp_probe,
+ .read_header = rtsp_read_header,
+ .read_packet = rtsp_read_packet,
+ .read_close = rtsp_read_close,
+ .read_seek = rtsp_read_seek,
+ .flags = AVFMT_NOFILE,
+ .read_play = rtsp_read_play,
+ .read_pause = rtsp_read_pause,
+ .priv_class = &rtsp_demuxer_class,
};
projects / ffmpeg / blobdiff