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path: root/ext/rtmp/gstrtmpsrc.c
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* rtmpsrc: mention setting librtmp flags in docsR S Nikhil Krishna2021-08-091-0/+10
| | | | Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2424>
* rtmp: allow per feature registrationStéphane Cerveau2021-03-231-0/+3
| | | | | | | | | | | | | Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2038>
* rtmp, transcodebin: fix i18n header includesTim-Philipp Müller2020-07-071-1/+1
| | | | | | Fixes #1351 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1416>
* documentation: fixed a heap o' typosAaron Boxer2019-11-051-1/+1
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* rtmpsrc: fix buffer leak on read error or EOSIlya Smelykh2019-03-201-0/+2
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* rtmpsrc: fix flushing seekGeorge Kiagiadakis2017-08-081-20/+47
| | | | | | | | | | | | | Previously this was broken, because a flushing seek causes unlock() to be called and in the implementation of unlock() we close the socket, so the seek errors out. This patch fixes it by re-connecting before the seek. Unfortunately, a seek does not work properly right after re-connecting, so a small hack is also in place: we read 1 buffer before seeking to allow librtmp to do its processing in RTMP_Read() https://bugzilla.gnome.org/show_bug.cgi?id=785941
* rtmpsrc: remove unused macroGeorge Kiagiadakis2017-08-081-7/+0
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* docs: Port all docstring to gtk-doc markdownThibault Saunier2017-04-121-3/+3
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* rtmpsrc: Remove dead assignmentsEdward Hervey2016-05-151-8/+6
| | | | | * read is only used within the while loop * todo and bsize only need to be assigned once
* rtmpsrc plugin : add timeout optionYann Jouanin2016-03-271-2/+18
| | | | https://bugzilla.gnome.org/show_bug.cgi?id=764251
* bad: use new gst_element_class_add_static_pad_template()Vineeth TM2016-03-241-2/+1
| | | | https://bugzilla.gnome.org/show_bug.cgi?id=763081
* rtmpsrc: check for failed RTMP context allocReynaldo H. Verdejo Pinochet2015-12-301-6/+15
| | | | | | | | | | | 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.
* plugins-bad: Fix example pipelinesVineeth TM2015-12-151-1/+1
| | | | | | | | 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
* rtmpsrc: plug memory-leaksHavard Graff2015-10-031-4/+4
| | | | https://bugzilla.gnome.org/show_bug.cgi?id=756001
* rtmpsrc: Fix indentation with gst-indentJohn Slade2015-10-021-2/+2
| | | | https://bugzilla.gnome.org/show_bug.cgi?id=755732
* rtmpsrc: Report limited bandwidthJan Alexander Steffens (heftig)2014-07-011-0/+9
| | | | | | | Makes uridecodebin treat this source as a stream source, allowing timeshifting. https://bugzilla.gnome.org/show_bug.cgi?id=732335
* rtmpsrc: Fix position queryingEdward Hervey2014-06-051-1/+1
| | | | It's the position we're querying, not the duration :)
* rtmpsrc: error out if we get EOS immediately without any dataTim-Philipp Müller2014-05-101-2/+8
| | | | It's not really right to just go EOS as if nothing was wrong.
* rtmpsrc: Implement basesrc->unlock()David Schleef2013-04-011-1/+19
| | | | | This fixes ->NULL transition problems if librtmp is stuck in a recv or send call that never returns.
* rtmpsrc: disable seeking if the configured url specifies live=trueAlessandro Decina2012-12-011-6/+7
| | | | | Disable seeking when live=true is set in the location URL (eg: "rtmp://example.net/stream live=true")
* Fix FSF addressTim-Philipp Müller2012-11-041-2/+2
| | | | https://bugzilla.gnome.org/show_bug.cgi?id=687520
* Use gst_element_class_set_static_metadata()Tim-Philipp Müller2012-10-171-1/+1
| | | | | where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.
* replace gst_element_class_set_details_simple with gst_element_class_set_metadataMark Nauwelaerts2012-09-141-1/+1
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* rtmpsrc: Fix element losing data at the end of buffersMike Ruprecht2012-06-211-1/+1
| | | | | | | | | | 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
* update for memory api changesWim Taymans2012-03-151-1/+1
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* rtmp: don't use gst_element_class_install_std_props()Tim-Philipp Müller2012-02-091-4/+10
| | | | It's about to be removed.
* port some more to new memory APIMark Nauwelaerts2012-01-251-3/+6
| | | | Fixes #668677.
* GST_FLOW_UNEXPECTED -> GST_FLOW_EOSWim Taymans2012-01-041-1/+1
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* Merge commit '26d6add9457f00ce8ec13844368466f0e3816e5d' into 0.11Tim-Philipp Müller2011-11-281-0/+16
|\ | | | | | | | | Conflicts: ext/rtmp/gstrtmpsink.c
| * rtmp: add WSAStartup and WSACleanup on Win32Julien Isorce2011-11-281-0/+16
| | | | | | | | https://bugzilla.gnome.org/show_bug.cgi?id=661098
* | Update for GstURIHandler get_protocols() changesTim-Philipp Müller2011-11-131-10/+16
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* | rtmp: port to 0.11Wim Taymans2011-10-081-44/+33
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* RTMP: add rtmpsink element for output to an RTMP serverJan Schmidt2011-06-181-15/+2
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* rtmpsrc: fix wrong use of GST_ELEMENT_ERRORTim-Philipp Müller2010-09-021-1/+1
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* rtmpsrc: fix warning on osx.Alessandro Decina2010-07-301-1/+1
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* rtmp: All read return values smaller than zero are failuresSebastian Dröge2010-06-231-1/+1
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* rtmpsrc: Do some sanity checks before accepting an URISebastian Dröge2010-06-231-1/+2
| | | | Fixes bug #622369.
* rtmpsrc: Fix timestamps after a seekSebastian Dröge2010-06-091-5/+6
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* rtmpsrc: Remove page-url and swf-url propertiesSebastian Dröge2010-06-071-42/+0
| | | | | | 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
* rtmpsrc: Fix memory leaksSebastian Dröge2010-06-071-6/+4
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* rtmpsrc: Add some braces to improve readabilitySebastian Dröge2010-06-061-2/+2
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* rtmpsrc: Improve timestamp handling a bitSebastian Dröge2010-06-061-3/+6
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* rtmpsrc: Add support for seekingSebastian Dröge2010-06-061-8/+118
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* rtmpsrc: Handle timestamps and the position querySebastian Dröge2010-06-061-7/+21
| | | | | This is not very accurate but better than nothing. The demuxer after the source knows more accurate timestamps.
* rtmpsrc: Allocate and free the RTMP instance in start/stopSebastian Dröge2010-06-061-57/+55
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* rtmpsrc: Add properties for setting the swfUrl and pageUrl propertiesSebastian Dröge2010-06-051-3/+56
| | | | These are required for some streams unfortunately.
* rtmpsrc: Major cleanup and reorganizationSebastian Dröge2010-06-051-308/+118
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* rtmp: Move to ext and drop internal librtmp copySebastian Dröge2010-06-051-0/+652
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.
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