Commit 5592bdd4 authored by Wim Taymans's avatar Wim Taymans
Browse files

gst/rtsp/gstrtspsrc.*: Fix method detection again.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Fix method detection again.
Keep track of when we must send a Range header.
Use segment values for Range, Speed and Scale headers.
Parse Speed and Scale headers to update the segment values.
parent 09a56877
2007-08-22 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Fix method detection again.
Keep track of when we must send a Range header.
Use segment values for Range, Speed and Scale headers.
Parse Speed and Scale headers to update the segment values.
2007-08-22 Stefan Kost <ensonic@users.sf.net>
patch by: Mark Nauwelaerts <manauw@skynet.be>
......
......@@ -1134,10 +1134,9 @@ gst_rtspsrc_do_seek (GstRTSPSrc * src, GstSegment * segment)
{
gboolean res;
/* PLAY from new position, we are flushing now */
src->position = ((gdouble) segment->last_stop) / GST_SECOND;
/* PLAY will add the range header now. */
src->state = GST_RTSP_STATE_SEEKING;
src->need_range = TRUE;
res = gst_rtspsrc_play (src);
......@@ -3139,9 +3138,8 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
gint indx = 0;
gint i;
/* reset supported methods, FIXME, extensions should be able to configure
* this. */
src->methods = GST_RTSP_PLAY | GST_RTSP_PAUSE;
/* reset supported methods */
src->methods = 0;
/* Try Allow Header first */
field = GST_RTSP_HDR_ALLOW;
......@@ -3181,11 +3179,14 @@ gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
if (src->methods == 0) {
/* neither Allow nor Public are required, assume the server supports
* DESCRIBE, SETUP, PLAY and PAUSE */
* at least DESCRIBE, SETUP, we always assume it supports PLAY and PAUSE as
* well. */
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
src->methods =
GST_RTSP_DESCRIBE | GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE;
src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
}
/* always assume PLAY and PAUSED, FIXME, extensions should be able to override
* this */
src->methods |= GST_RTSP_PLAY | GST_RTSP_PAUSE;
/* we need describe and setup */
if (!(src->methods & GST_RTSP_DESCRIBE))
......@@ -3631,7 +3632,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* reset our state */
gst_segment_init (&src->segment, GST_FORMAT_TIME);
src->position = 0.0;
src->need_range = TRUE;
/* can't continue without a valid url */
if (G_UNLIKELY (src->url == NULL))
......@@ -3988,7 +3989,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res;
gchar *rtpinfo, *range;
gchar *hval;
GST_RTSP_STATE_LOCK (src);
......@@ -4007,13 +4008,30 @@ gst_rtspsrc_play (GstRTSPSrc * src)
if (res < 0)
goto create_request_failed;
if (src->position == 0.0)
range = g_strdup_printf ("npt=0-");
else
range = g_strdup_printf ("npt=%f-", src->position);
if (src->need_range) {
if (src->segment.last_stop == 0)
hval = g_strdup_printf ("npt=0-");
else
hval =
g_strdup_printf ("npt=%f-",
((gdouble) src->segment.last_stop) / GST_SECOND);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
g_free (hval);
src->need_range = FALSE;
}
if (src->segment.rate != 1.0) {
hval = g_strdup_printf ("%f", src->segment.rate);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
g_free (hval);
}
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, range);
g_free (range);
if (src->segment.applied_rate != 1.0) {
hval = g_strdup_printf ("%f", src->segment.applied_rate);
gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
g_free (hval);
}
if ((res = gst_rtspsrc_send (src, &request, &response, NULL)) < 0)
goto send_error;
......@@ -4022,16 +4040,40 @@ gst_rtspsrc_play (GstRTSPSrc * src)
/* parse RTP npt field. This is the current position in the stream (Normal
* Play Time) and should be put in the NEWSEGMENT position field. */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &range,
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
0) == GST_RTSP_OK)
gst_rtspsrc_parse_range (src, range);
gst_rtspsrc_parse_range (src, hval);
/* parse Speed header. This is the intended playback rate of the stream
* and should be put in the NEWSEGMENT rate field. */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
0) == GST_RTSP_OK) {
gfloat fval;
if (sscanf (hval, "%f", &fval) > 0)
src->segment.rate = fval;
} else {
src->segment.rate = 1.0;
}
/* parse Scale header. This is the playback rate as sent by the server
* and should be put in the NEWSEGMENT applied_rate field. */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE, &hval,
0) == GST_RTSP_OK) {
gfloat fval;
if (sscanf (hval, "%f", &fval) > 0)
src->segment.applied_rate = fval;
} else {
src->segment.applied_rate = 1.0;
}
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
* This is info for the RTP session manager that we pass to it in caps. */
if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
&rtpinfo, 0) == GST_RTSP_OK)
gst_rtspsrc_parse_rtpinfo (src, rtpinfo);
&hval, 0) == GST_RTSP_OK)
gst_rtspsrc_parse_rtpinfo (src, hval);
gst_rtsp_message_unset (&response);
......
......@@ -130,7 +130,7 @@ struct _GstRTSPSrc {
GStaticRecMutex *stream_rec_lock;
GstSegment segment;
gboolean running;
gdouble position;
gboolean need_range;
gint free_channel;
GstEvent *close_segment;
GstEvent *start_segment;
......
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