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48 commits

Author SHA1 Message Date
John Slade
30fa95c6e2 rtmpsrc: Fix indentation with gst-indent 
https://bugzilla.gnome.org/show_bug.cgi?id=755732
 2015-10-02 15:06:02 +03:00
Edward Hervey
86c500a47a rtmpsink: Initialize GstMapInfo 
Avoids doing a call to unmap with it uninitialized

CID #1302834
 2015-06-01 13:56:03 +02:00
Vivia Nikolaidou
fba7c97135 rtmpsink: Do not crash when receiving buffers after GST_FLOW_ERROR 
If the RTMP URI is invalid, the rtmpsink will return GST_FLOW_ERROR.
If it still receives buffers after that, it shouldn't crash.

https://bugzilla.gnome.org/show_bug.cgi?id=750104
 2015-05-30 00:25:37 +10:00
Sebastian Dröge
a4d8efde0f rtmpsink: Declare sink variable that was forgotten in last commit 2014-10-20 09:47:27 +02:00
Havard Graff
a1e948cddd rtmpsink: Free URI string in finalize() 
https://bugzilla.gnome.org/show_bug.cgi?id=738674
 2014-10-20 09:36:40 +02:00
Jan Alexander Steffens (heftig)
86080cb5cc rtmpsrc: Report limited bandwidth 
Makes uridecodebin treat this source as a stream source,
allowing timeshifting.

https://bugzilla.gnome.org/show_bug.cgi?id=732335
 2014-07-01 15:02:37 +02:00
Edward Hervey
5300e59f09 rtmp: proxy logging from librtmp 
Helps with debugging various librtmp issues
 2014-06-05 09:41:31 +02:00
Edward Hervey
3cb5bc8868 rtmpsrc: Fix position querying 
It's the position we're querying, not the duration :)
 2014-06-05 09:41:31 +02:00
Tim-Philipp Müller
ab783acd7f rtmpsrc: error out if we get EOS immediately without any data 
It's not really right to just go EOS as if nothing was wrong.
 2014-05-10 12:57:29 +01:00
Jan Schmidt
6b784cf808 rtmpsink: Remove URL check for valid playpath. 
The playpath is an optional component of the URL - don't require it.
 2014-03-26 09:05:55 +11:00
Sebastian Dröge
e51cd4fe2f gst: Add better support for static plugins 2013-04-15 15:59:22 +02:00
David Schleef
94ed6caec4 rtmpsrc: Implement basesrc->unlock() 
This fixes ->NULL transition problems if librtmp is stuck in a
recv or send call that never returns.
 2013-04-01 19:53:01 -07:00
Alessandro Decina
62879bdd38 rtmpsrc: disable seeking if the configured url specifies live=true 
Disable seeking when live=true is set in the location URL (eg:
"rtmp://example.net/stream live=true")
 2012-12-01 17:11:43 +01:00
Tim-Philipp Müller
9e1b75fda3 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:09:59 +00:00
Tim-Philipp Müller
32ba17cd0f Use gst_element_class_set_static_metadata() 
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
 2012-10-17 17:46:34 +01:00
Sergey N. Gorshkov
6d0d209a28 rtmpsink: handle RTMP_Write() return value correctly 
Error might also be negative (-1). Unclear if 0 should
be fatal as well though.

https://bugzilla.gnome.org/show_bug.cgi?id=681111
https://bugzilla.gnome.org/show_bug.cgi?id=686009
 2012-10-12 23:29:53 +01:00
David Régade
65add5533a rtmpsink: fix memory leak from URI verification via RTMP_ParseURL() 
In gst_rtmp_sink_uri_set_uri(), a test is performed in order
to be sure uri is correct for librtmp. This test calls
RTMP_ParseURL with 3 AVal pointers as parameters: host,
playpath and app.

AVal is a struct with a char* + int. After RTMP_ParseURL call,
host.av_val and app.av_val both refer a substring of "uri". But
playpath.av_val may be the result of a malloc so it needs to
be freed.

https://bugzilla.gnome.org/show_bug.cgi?id=681459
 2012-10-12 23:09:06 +01:00
Mark Nauwelaerts
578861abea replace gst_element_class_set_details_simple with gst_element_class_set_metadata 2012-09-14 17:27:49 +02:00
Mike Ruprecht
96b7059d24 rtmpsrc: Fix element losing data at the end of buffers 
rtmpsrc outputs truncated buffers because, when enough data is
read to fill the buffer, the amount read that time (todo) is set
to zero before it's added to the cumulative buffer size (bsize).
The buffer is then truncated to bsize resulting in lost data.
This patch adds todo to bsize before setting todo to zero.

Fixes #678509
 2012-06-21 08:36:35 +01:00
Tim-Philipp Müller
b87f7345db Add WINSOCK2_LIBS, remove WIN32_LIBS, fix rtmp build on Windows some more 
One way of passing -lws2_32 to plugins should be enough..
 2012-05-05 18:20:33 +01:00
Sebastian Dröge
cda192b3b7 gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 18:02:56 +02:00
Wim Taymans
a9ec4d62a8 update for buffer changes 2012-03-28 12:53:09 +02:00
Wim Taymans
6cbb840385 update for memory api changes 2012-03-15 13:37:36 +01:00
Tim-Philipp Müller
658cbeac06 rtmp: don't use gst_element_class_install_std_props() 
It's about to be removed.
 2012-02-09 00:09:36 +00:00
Mark Nauwelaerts
12ee41829c port some more to new memory API 
Fixes #668677.
 2012-01-25 18:50:40 +01:00
Wim Taymans
acfa55df6c GST_FLOW_UNEXPECTED -> GST_FLOW_EOS 2012-01-04 10:02:28 +01:00
Tim-Philipp Müller
2a78a3010d Merge commit '26d6add9457f00ce8ec13844368466f0e3816e5d' into 0.11 
Conflicts:
	ext/rtmp/gstrtmpsink.c
 2011-11-28 23:20:02 +00:00
Julien Isorce
26d6add945 rtmp: add WSAStartup and WSACleanup on Win32 
https://bugzilla.gnome.org/show_bug.cgi?id=661098
 2011-11-28 10:34:45 +00:00
Tim-Philipp Müller
026af880b5 faac, rtmp: more printf format fixes in debug messages 
https://bugzilla.gnome.org/show_bug.cgi?id=662618
 2011-11-23 23:43:48 +00:00
Tim-Philipp Müller
357d7bdfed Update for GstURIHandler get_protocols() changes 2011-11-13 23:55:56 +00:00
Wim Taymans
9ddfdfe60c rtmp: port to 0.11 2011-10-08 11:40:25 +02:00
Alessandro Decina
f33b78abd1 rtmpsink: don't block the main thread with RTMP_Connect 
Move the RTMP_Connect call from the main thread (::start) to the streaming
thread (::render).
 2011-09-12 11:23:03 +02:00
Jan Schmidt
38bf3169ff RTMP: add rtmpsink element for output to an RTMP server 2011-06-18 01:09:51 +10:00
Tim-Philipp Müller
d6c908ea59 rtmpsrc: fix wrong use of GST_ELEMENT_ERROR 2010-09-02 22:39:33 +01:00
Alessandro Decina
fc9cfb0c00 rtmpsrc: fix warning on osx. 2010-07-30 23:59:10 +02:00
Sebastian Dröge
af4c066bc3 rtmp: All read return values smaller than zero are failures 2010-06-23 22:19:33 +02:00
Sebastian Dröge
c15487961b rtmpsrc: Do some sanity checks before accepting an URI 
Fixes bug #622369.
 2010-06-23 21:46:42 +02:00
Sebastian Dröge
f0e7bd298c rtmpsrc: Fix timestamps after a seek 2010-06-09 20:49:10 +02:00
Sebastian Dröge
5417900a0e rtmpsrc: Remove page-url and swf-url properties 
It's possible to include all those options in the URL already
by appending the options and separating them by spaces, e.g.
rtmp://somewhere/something opt1=val1 opt2=val2
 2010-06-07 17:39:07 +02:00
Sebastian Dröge
6aa4a71604 rtmpsrc: Fix memory leaks 2010-06-07 17:31:40 +02:00
Sebastian Dröge
370a5049ba rtmpsrc: Add some braces to improve readability 2010-06-06 15:32:39 +02:00
Sebastian Dröge
d0ce1ff675 rtmpsrc: Improve timestamp handling a bit 2010-06-06 15:29:34 +02:00
Sebastian Dröge
827ecadb81 rtmpsrc: Add support for seeking 2010-06-06 15:24:23 +02:00
Sebastian Dröge
fdf1598173 rtmpsrc: Handle timestamps and the position query 
This is not very accurate but better than nothing. The demuxer
after the source knows more accurate timestamps.
 2010-06-06 13:57:06 +02:00
Sebastian Dröge
21f976066c rtmpsrc: Allocate and free the RTMP instance in start/stop 2010-06-06 08:30:09 +02:00
Sebastian Dröge
d289105409 rtmpsrc: Add properties for setting the swfUrl and pageUrl properties 
These are required for some streams unfortunately.
 2010-06-05 18:02:39 +02:00
Sebastian Dröge
c3d10ed72a rtmpsrc: Major cleanup and reorganization 2010-06-05 18:02:39 +02:00
Sebastian Dröge
547f037ea4 rtmp: Move to ext and drop internal librtmp copy 
We really don't want this in gst-plugins-bad because of
legal complexities around RTMP and possible problems
for distributions.

Add README that explains how to build librtmp to be suitable
for linking to the GStreamer plugin.
 2010-06-05 18:02:39 +02:00