Thibault Saunier
78022a6e0c
docs: Port all docstring to gtk-doc markdown
2017-04-12 12:57:57 -03:00
Edward Hervey
a76ad40c6c
rtmpsrc: Remove dead assignments
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* read is only used within the while loop
* todo and bsize only need to be assigned once
2016-05-15 14:18:23 +02:00
Yann Jouanin
9554e1c666
rtmpsrc plugin : add timeout option
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https://bugzilla.gnome.org/show_bug.cgi?id=764251
2016-03-27 11:54:36 +03:00
Vineeth TM
8cdfb13658
bad: use new gst_element_class_add_static_pad_template()
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https://bugzilla.gnome.org/show_bug.cgi?id=763081
2016-03-24 14:56:51 +02:00
Reynaldo H. Verdejo Pinochet
00587eb561
rtmpsrc: check for failed RTMP context alloc
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Avoids an unlikely crash.
Arguably, if allocation fails we have no chance of
recovering but nonetheless, RTMP_Alloc can fail and
librtmp's RTMP_init() (called next) assumes a non-NULL
pointer is passed without checking.
Additionally, unify exit path on error.
2015-12-30 17:22:54 -08:00
Vineeth TM
7c42ba97d7
plugins-bad: Fix example pipelines
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rename gst-launch --> gst-launch-1.0
replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
fix caps in examples
https://bugzilla.gnome.org/show_bug.cgi?id=759432
2015-12-15 10:30:49 +00:00
Havard Graff
cb20105aa5
rtmpsrc: plug memory-leaks
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https://bugzilla.gnome.org/show_bug.cgi?id=756001
2015-10-03 17:52:51 +01:00
John Slade
30fa95c6e2
rtmpsrc: Fix indentation with gst-indent
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https://bugzilla.gnome.org/show_bug.cgi?id=755732
2015-10-02 15:06:02 +03:00
Jan Alexander Steffens (heftig)
86080cb5cc
rtmpsrc: Report limited bandwidth
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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
3cb5bc8868
rtmpsrc: Fix position querying
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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
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It's not really right to just go EOS as if nothing was wrong.
2014-05-10 12:57:29 +01:00
David Schleef
94ed6caec4
rtmpsrc: Implement basesrc->unlock()
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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
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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
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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()
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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
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
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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
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()
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It's about to be removed.
2012-02-09 00:09:36 +00:00
Mark Nauwelaerts
12ee41829c
port some more to new memory API
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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
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Conflicts:
ext/rtmp/gstrtmpsink.c
2011-11-28 23:20:02 +00:00
Julien Isorce
26d6add945
rtmp: add WSAStartup and WSACleanup on Win32
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https://bugzilla.gnome.org/show_bug.cgi?id=661098
2011-11-28 10:34:45 +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
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
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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
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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
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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
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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
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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