mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-26 19:51:11 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-rtsp-server/ChangeLog

14309 lines
508 KiB
Text
Raw Blame History

This file contains ambiguous Unicode characters

This file contains Unicode characters that might be confused with other characters. If you think that this is intentional, you can safely ignore this warning. Use the Escape button to reveal them.

=== release 1.19.2 ===
2021-09-23 01:35:27 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* gst-rtsp-server.doap:
* meson.build:
Release 1.19.2
2021-07-05 11:54:18 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-sink/gstrtspclientsink.c:
Protection against early RTCP packets.
When receiving RTCP packets early the funnel is not ready yet and
GST_FLOW_FLUSHING will be returned when pushing data to it's srcpad.
This causes the thread that handle RTCP packets to go to pause mode.
Since this thread is in pause mode there will be no further callbacks to
handle keep-alive for incoming RTCP packets. This will make the session
time out if the client is not using another keep-alive mechanism.
Change-Id: Idb29db05f59c06423fa693a2aeeacbe3a1883fc5
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/211>
2021-06-21 08:34:35 +0000 Corentin Damman <c.damman@intopix.com>
* COPYING:
* COPYING.LIB:
Update COPYING.LIB, COPYING files
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/210>
2021-06-01 15:29:07 +0100 Tim-Philipp Müller <tim@centricular.com>
* docs/gst_plugins_cache.json:
* meson.build:
Back to development
=== release 1.19.1 ===
2021-06-01 00:15:08 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.19.1
2021-05-24 18:58:00 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: use new gst_buffer_new_memdup()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/208>
2021-05-04 20:47:18 -0400 Doug Nazar <nazard@nazar.ca>
* gst/rtsp-server/rtsp-media-factory-uri.c:
rtsp-media: fix leak when adding converter
Free the previous caps before reusing the variable for the converter caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
2021-05-04 20:45:19 -0400 Doug Nazar <nazard@nazar.ca>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix leak adding headers
gst_rtsp_message_add_header() makes a copy of the header, instead
of taking ownership.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/204>
2021-04-21 10:43:41 +0200 François Laignel <fengalin@free.fr>
* gst/rtsp-server/rtsp-stream.c:
Use gst_element_request_pad_simple...
Instead of the deprecated gst_element_get_request_pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/195>
2021-04-29 03:07:42 -0400 Doug Nazar <nazard@nazar.ca>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Ensure the bus watch is removed during unprepare
It's possible for the destruction of the source to be delayed.
Instead of relying on the dispose() to remove the bus watch, do
it ourselves.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/202>
2021-04-27 09:22:21 +0200 Marc Leeman <m.leeman@televic.com>
* docs/README:
docs: minor spelling correction in README
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
2021-04-27 09:05:39 +0200 Marc Leeman <m.leeman@televic.com>
* examples/test-replay-server.c:
test-replay-server: minor spelling corrections
Bumped on these while investigating the example code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/200>
2021-04-22 23:26:02 -0400 Doug Nazar <nazard@nazar.ca>
* tests/check/gst/stream.c:
tests: Don't fail tests if IPv6 not available.
On computers with IPv6 disabled it shouldn't result in a test failure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/196>
2021-04-23 07:18:48 +0200 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Add one more case to seek avoidance
This is an extension to the previous commit. There can also be cases where the
start position is not specified, in those cases we should also avoid doing
seeking unless it's forced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/197>
2021-04-16 14:35:02 -0400 Doug Nazar <nazard@nazar.ca>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Improve skipping trickmode seek.
We can also skip the seek if the end range is already
correct.
Avoids initial seek on play start if playing full stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/194>
2021-03-19 10:36:01 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Don't run signal class handlers during the CLEANUP stage
It's sufficient to run them during the FIRST stage instead of in both.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/193>
2021-02-15 12:07:15 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspclientsink.c:
tests: rtspclientsink: fix some leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
2021-02-15 12:26:30 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: mark cached caps as maybe-leaked to make leaks tracer happy
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/190>
2021-02-15 12:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspclientsink.c:
rtspclientsink: add unit test for potential shutdown deadlock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
2021-02-15 12:01:34 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: fix deadlock on shutdown before preroll
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/130
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/189>
2021-02-01 12:16:46 +0100 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: avoid deadlock in send_func
Currently the send_func() runs in a thread of its own which is started
the first time we enter handle_new_sample(). It runs in an outer loop
until priv->continue_sending is FALSE, which happens when a TEARDOWN
request is received. We use a local variable, cont, which is initialized
to TRUE, meaning that we will always enter the outer loop, and at the
end of the outer loop we assign it the value of priv->continue_sending.
Within the outer loop there is an inner loop, where we wait to be
signaled when there is more data to send. The inner loop is exited when
priv->send_cookie has changed value, which it does when more data is
available or when a TEARDOWN has been received.
But if we get a TEARDOWN before send_func() is entered we will get stuck
in the inner loop because no one will increase priv->session_cookie
anymore.
By not entering the outer loop in send_func() if priv->continue_sending
is FALSE we make sure that we do not get stuck in send_func()'s inner
loop should we receive a TEARDOWN before the send thread has started.
Change-Id: I7338a0ea60ea435bb685f875965f5165839afa20
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/187>
2021-01-22 08:58:23 +0100 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: cleanup transports during TEARDOWN
When tunneling RTP over RTSP the stream transports are stored in a hash
table in the GstRTSPClientPrivate struct. They are used for, among other
things, mapping channel id to stream transports when receiving data from
the client. The stream tranports are created and added to the hash table
in handle_setup_request(), but unfortuately they are not removed in
handle_teardown_request(). This means that if the client sends data on
the RTSP connection after it has sent the TEARDOWN, which is often the
case when audio backchannel is enabled, handle_data() will still be able
to map the channel to a session transport and pass the data along to it.
Which eventually leads to a failing assert in gst_rtsp_stream_recv_rtp()
because the stream is no longer joined to a bin.
We avoid this by removing the stream transports from the hash table when
we handle the TEARDOWN request.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/184>
2020-12-15 11:07:01 +0200 Sebastian Dröge <sebastian@centricular.com>
* docs/gst_plugins_cache.json:
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Add "update-sdp" signal that allows updating the SDP before sending it to the server
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/178>
2020-12-23 13:54:54 -0500 John Lindgren <john.lindgren@avasure.com>
* tests/check/gst/client.c:
Add test cases for mountpoint of '/'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
2020-11-05 16:02:49 -0500 John Lindgren <john.lindgren@avasure.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-session-media.c:
Make a mount point of "/" work correctly.
As far as I can tell, this is neither explicitly allowed nor
forbidden by RFC 7826.
Meanwhile, URLs such as rtsp://<IP>:554 or rtsp://<IP>:554/ are in
use in the wild (presumably with non-GStreamer servers).
GStreamer's prior behavior was confusing, in that
gst_rtsp_mount_points_add_factory() would appear to accept a mount
path of "" or "/", but later connection attempts would fail with a
"media not found" error.
This commit makes a mount path of "/" work for either form of URL,
while an empty mount path ("") is rejected and logs a warning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/168>
2020-12-15 10:18:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* docs/gst_plugins_cache.json:
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Use proper types instead of G_TYPE_POINTER for the RTSP messages in the "handle-request" signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/177>
2020-12-17 15:27:27 +0100 Tobias Ronge <tobiasr@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only count senders when counting blocked streams
Only sender streams sends the GstRTSPStreamBlocking message, so only
these should be counted before setting media status to prepared.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/180>
2020-10-21 15:38:43 +0200 Jimmi Holst Christensen <jimmi.christensen@aivero.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink add proper support for uri queries
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/166>
2020-12-14 14:12:38 +1300 Lawrence Troup <lawrence.troup@teknique.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Only unref client watch context on finalize, to avoid deadlock
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/127
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/176>
2020-11-18 20:36:50 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: collect a clock_rate when blocking
This lets us provide a clock_rate in a fashion similar to the
other code paths in get_rtpinfo()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/174>
2020-11-16 10:34:41 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Use guint64 for setting the size-time property on rtpstorage
Otherwise this will cause memory corruption as the property expects a 64
bit integer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/169>
2020-11-03 16:56:28 +0100 David Phung <davidph@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-media: Ignore GstRTSPStreamBlocking from incomplete streams
To prevent cases with prerolling when the inactive stream prerolls first
and the server proceeds without waiting for the active stream, we will
ignore GstRTSPStreamBlocking messages from incomplete streams. When
there are no complete streams (during DESCRIBE), we will listen to all
streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
2020-10-28 21:48:06 +0100 Kristofer Björkström <kristofb@axis.com>
* tests/check/gst/media.c:
* tests/check/meson.build:
* tests/files/test.avi:
media test: Add test for seeking one active stream with a demuxer
Add another seek_one_active_stream test but with a demuxer. The demuxer
will flush both streams in opposed to the existing test which only
flushes the active stream. This will help exposing problems with the
prerolling process after a flushing seek.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/167>
2018-10-29 09:19:33 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/meson.build:
* meson.build:
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp-server.pc.in:
* pkgconfig/meson.build:
Meson: Use pkg-config generator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/1>
2020-10-19 11:25:25 +0300 Sebastian Dröge <sebastian@centricular.com>
* meson.build:
meson: update glib minimum version to 2.56
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/164>
2020-09-04 21:14:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-launch.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
rtsp-media-factory: expose API to disable RTCP
This is supported by the RFC, and can be useful on systems where
allocating two consecutive ports is problematic, and RTCP is not
necessary.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/159>
2020-10-08 23:45:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* hooks/pre-commit.hook:
* meson.build:
git: use our standard pre commit hook
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/162>
2020-10-08 22:17:16 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: make use of blocked_running_time in query_position
When blocking, the sink element will not have received a buffer
yet and the position query will fail. Instead, we make use of
the running time of the buffer we blocked on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
2020-10-06 00:04:17 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: collect rtp info when blocking
We don't unblock the stream anymore before replying to the
play request (883ddc72bb5bc57c95a9e167814d1ac53fe1b443),
so the sinks don't have a last-sample after potentially flush
seeking. seek_trickmode waits for preroll however, which means
the stream will block and wait for a first buffer. Subsequent
calls to get_rtpinfo() can thus make use of the information.
See https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/115
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/160>
2020-09-27 20:09:22 +0900 Seungha Yang <seungha@centricular.com>
* examples/meson.build:
* examples/test-replay-server.c:
* examples/test-replay-server.h:
examples: Add an example for loop playback
This demo example shows a way of file loop playback of a given source.
Note that client seek request is not properly implemented yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/154>
2020-09-28 22:03:47 +0200 David Phung <davidph@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Plug memory leak
The get-storage signal of rtpbin increases the ref count of the storage.
So we have to unref it after usage.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/155>
2020-09-11 15:46:41 +0200 Guiqin Zou <guiqinzu@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Get rates only on sender streams
When play a media with both sender and receiver stream, like ONVIF
back channel audio in, gst_rtsp_media_get_rates call
gst_rtsp_stream_get_rates for each stream to set the rates. But
gst_rtsp_stream_get_rates return false for the receiver steam, which
lead a g_assert crash.
Instead to get rates on all streams, now just get rates on sender
streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/150>
2020-09-05 00:30:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream.c:
rtsp-media: set a 0 storage size for TCP receivers
ulpfec correction is obviously useless when receiving a stream
over TCP, and in TCP modes the rtp storage receives non
timestamped buffers, causing it to queue buffers indefinitely,
until the queue grows so large that sanity checks kick in and
warnings start to get emitted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/149>
2020-08-21 03:02:40 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: preroll on gap events
This allows negotiating a SDP with all streams present, but only
start sending packets at some later point in time
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/146>
2020-08-25 16:10:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: do not unblock on unsuspend
rtsp_media_unsuspend() is called from handle_play_request()
before sending the play response. Unblocking the streams here
was causing data to be sent out before the client was ready
to handle it, with obvious side effects such as initial packets
getting discarded, causing decoding errors.
Instead we can simply let the media streams be unblocked when
the state of the media is set to PLAYING, which occurs after
sending the play response.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/147>
2020-09-08 17:30:49 +0100 Tim-Philipp Müller <tim@centricular.com>
* .gitlab-ci.yml:
ci: include template from gst-ci master branch again
2020-09-08 16:58:58 +0100 Tim-Philipp Müller <tim@centricular.com>
* docs/gst_plugins_cache.json:
* meson.build:
Back to development
=== release 1.18.0 ===
2020-09-08 00:08:29 +0100 Tim-Philipp Müller <tim@centricular.com>
* .gitlab-ci.yml:
* ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.18.0
=== release 1.17.90 ===
2020-08-20 16:15:06 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.17.90
2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
* gst/rtsp-server/rtsp-thread-pool.c:
rtsp-thread-pool.c: fix clang 10 warning
clang 10 is complaining about incompatible types due to the
glib typesystem.
```
../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-thread-pool.c:534:10: error: incompatible pointer types passing 'typeof ((((void *)0))) *' (aka 'void **') to parameter of type 'GThreadPool **' (aka 'struct _GThreadPool **') [-Werror,-Wincompatible-pointer-types]
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
2020-08-03 19:34:30 +0300 Jordan Petridis <jordan@centricular.com>
* gst/rtsp-server/rtsp-thread-pool.c:
rtsp-thread-pool.c: fix clang 10 warning
clang 10 is complaining about incompatible types due to the
glib typesystem.
```
../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-thread-pool.c:534:10: error: incompatible pointer types passing 'typeof ((((void *)0))) *' (aka 'void **') to parameter of type 'GThreadPool **' (aka 'struct _GThreadPool **') [-Werror,-Wincompatible-pointer-types]
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/145>
2020-07-15 11:19:40 +0200 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Fix resource leak in mikey messsage
Fixed a resource leak for mikey message while adding crypto session
failed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/144>
2020-07-08 17:28:57 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
* scripts/extract-release-date-from-doap-file.py:
meson: set release date from .doap file for releases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/143>
2020-07-02 23:52:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: explicitly set caps on udpsrc elements
This causes them to send caps events before data flow, which is
usually a pretty correct thing to do!
Not doing so manifested in a bug where ssrcdemux wouldn't forward
the caps it had received with an extra ssrc field, as it hadn't
received any caps event.
Fixes #85
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/141>
2020-07-03 02:04:04 +0100 Tim-Philipp Müller <tim@centricular.com>
* docs/gst_plugins_cache.json:
* meson.build:
Back to development
=== release 1.17.2 ===
2020-07-03 00:33:54 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.17.2
2020-06-19 22:55:54 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/gst_plugins_cache.json:
doc: Stop documenting properties from parents
2020-06-22 20:04:45 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/gst_plugins_cache.json:
docs: Fix version in the plugins cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
2020-06-22 12:33:32 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Don't call gst_ghost_pad_construct() anymore
It's deprecated, unneeded and doesn't do anything anymore.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/138>
2020-06-20 00:28:28 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
Back to development
=== release 1.17.1 ===
2020-06-19 19:24:38 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* docs/gst_plugins_cache.json:
* gst-rtsp-server.doap:
* meson.build:
Release 1.17.1
2020-06-15 19:45:38 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Add/configure transports when completing the pipeline
Otherwise the transports are not set up yet during the PLAY request
handling when unsuspending (and thus unblocking) the media.
In case of live pipelines this then causes the first few packets to go
to the sinks before they know what to do with them, and they simply
discard them which is rather suboptimal in case of keyframes.
For non-live pipelines this is not a problem because the sink will still
be PAUSED and as such not send out the data yet but wait until it goes
to PLAYING, which is late enough.
Adding the transports multiple times is not a problem: if the transport
is already added it won't be added another time and TRUE will be
returned.
This fixes a regression introduced by a7732a68e8bc6b4ba15629c652056c240c624ff0
before 1.14.0.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/107
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
2020-06-15 19:45:21 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix misleading comment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
2020-06-15 18:29:13 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Make sure to also unblock pads when going to PLAYING while buffering
The pad probes are not needed anymore at this point and later when
reaching buffering 100% only the state is changed, no unblocking
happens.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
2020-06-15 18:17:40 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Remove duplicated media_unblock() function
It does literally the same as media_streams_set_blocked(FALSE).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/135>
2020-06-12 15:38:45 +0200 Lenny Jorissen <lennyjorissen@gmail.com>
* examples/test-onvif-server.c:
test-onvif-server: cast ntp-offset property value to 64 bit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/134>
2020-06-09 15:21:24 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/gst_plugins_cache.json:
docs: Update plugins cache
2020-06-10 13:45:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-server.c:
* examples/test-onvif-server.h:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
onvif-media-factory: define autoptr cleanup function
And have the factory in the onvif-server example inherit from
GstRTSPOnvifMediaFactory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/133>
2020-06-08 10:59:34 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/gst_plugins_cache.json:
docs: Update plugins cache
2020-06-08 09:45:15 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.com>
* tests/check/gst/rtspserver.c:
tests: enforce I420 format
Test was not enforcing a video format on videotestsrc. I420 was picked as it
was the first format in GST_VIDEO_FORMATS_ALL which will no longer be
true (gst-plugins-base!689).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/129>
2020-06-06 00:41:51 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
plugins: uddate gst_type_mark_as_plugin_api() calls
2020-06-03 18:36:25 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/meson.build:
doc: Require hotdoc >= 0.11.0
2020-05-27 17:00:05 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/gst_plugins_cache.json:
docs: Update gst_plugins_cache.json
2020-05-30 23:23:51 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types
2020-05-27 23:38:06 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/meson.build:
meson: gir: remove bogus sources_top_dir kwarg
Doesn't actually exist. Was fixed differently in Meson
so that the user doesn't have to specify it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/127>
2020-05-27 17:43:43 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/meson.build:
tests: put registry into tests/check not the gst/ subdir
Underscorify the test name before setting GST_REGISTRY,
so the registry actually ends up in the current build dir
and not some subdir.
For consistency with the other modules, but should also
avoid problems on windows.
Also fix indentation of environment block.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
2020-05-27 17:33:24 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/meson.build:
tests: fix meson test env setup to make sure we use the right gst-plugin-scanner
If core is built as a subproject (e.g. as in gst-build), make sure to use
the gst-plugin-scanner from the built subproject. Without this, gstreamer
might accidentally use the gst-plugin-scanner from the install prefix if
that exists, which in turn might drag in gst library versions we didn't
mean to drag in. Those gst library versions might then be older than
what our current build needs, and might cause our newly-built plugins
to get blacklisted in the test registry because they rely on a symbol
that the wrongly-pulled in gst lib doesn't have.
This should fix running of unit tests in gst-build when invoking
meson test or ninja test from outside the devenv for the case where
there is an older or different-version gst-plugin-scanner installed
in the install prefix.
In case no gst-plugin-scanner is installed in the install prefix, this
will fix "GStreamer-WARNING: External plugin loader failed. This most
likely means that the plugin loader helper binary was not found or
could not be run. You might need to set the GST_PLUGIN_SCANNER
environment variable if your setup is unusual." warnings when running
the unit tests.
In the case where we find GStreamer core via pkg-config we use
a newly-added pkg-config var "pluginscannerdir" to get the right
directory. This has the benefit of working transparently for both
installed and uninstalled pkg-config files/setups.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
2020-05-27 17:32:02 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/meson.build:
tests: gst-plugins-base and -bad plugins are required for the unit tests
Make hard requirement until we have more fine-grained control
in the unit tests. Of course the presence of the .pc file doesn't
imply that the plugins we need are actually there, but it's at
least a step in the right direction.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
2020-05-27 17:29:18 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/meson.build:
tests: pick up rtsp-server plugins from build directory only
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/126>
2020-05-26 15:31:22 +0200 Ludvig Rappe <ludvigr@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: wait for all GstRTSPStreamBlocking messages
Make sure rtsp-media have received a GstRTSPStreamBlocking message from
each active stream when checking if all streams are blocked.
Without this change there will be a race condition when using two or
more streams and rtsp-media receives a GstRTSPStreamBlocking message
from one of the streams. This is because rtsp-media then checks if all
streams are blocked by calling gst_rtsp_stream_is_blocking() for each
stream. This function call returns TRUE if the stream has sent a
GstRTSPStreamBlocking message, however, rtsp-media may have yet to
receive this message. This would then result in that rtsp-media
erroneously thinks it is blocking all streams which could result in
rtsp-media changing state, from PREPARING to PREPARED. In the case of a
preroll, this could result in that rtsp-media thinks that the pipeline
is prerolled even though that might not be the case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/124>
2020-05-04 13:43:00 +0200 Ludvig Rappe <ludvigr@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: update expected_async_done during suspend
Set expected_async_done to FALSE in default_suspend() if a state change
occurs and the return value from set_target_state() is something other
than GST_STATE_CHANGE_ASYNC.
Without this change there is a risk that expected_async_done will be
TRUE even though no asynchronous state change is taking place. This
could happen if the pipeline is set to PAUSED using
media_set_pipeline_state_locked(), an asynchronous state change starts
and then the media is suspended (which could result in a state change,
aborting the asynchronous state change). If the media is suspended
before the asynchronous state change ends then expected_async_done will
be TRUE but no asynchronous state change is taking place.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/123>
2020-05-25 13:49:45 +0200 Kristofer Björkström <kristofb@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix race condition in rtsp ctrl timeout by WeakRef client
There was a race condition where client was being finalized and
concurrently in some other thread the rtsp ctrl timout was relying on
client data that was being freed.
When rtsp ctrl timeout is setup, a WeakRef on Client is set.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/121>
2015-03-03 14:42:07 +0100 Gregor Boirie <gregor.boirie@parrot.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media-factory: complete DSCP QoS setting support
add dscp_qos setting support at factory and media level to setup IP DSCP
field of bounded UDP sinks.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
2020-05-14 10:08:32 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix some race conditions around timeout source removal
We always need to take the lock while accessing it as otherwise another
thread might've removed it in the meantime. Also when destroying and
creating a new one, ensure that the mutex is not shortly unlocked in
between as during that time another one might potentially be created
already.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/119>
2020-05-03 16:29:31 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-media: Mark out parameters accordingly in gst_rtsp_media_get_rates()
And the same for gst_rtsp_stream_get_rates().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/118>
2020-05-03 10:17:41 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-onvif-server.c:
examples: test-onvif-server: fix compiler warnings on raspbian
Fix printf format for 64-bit variables.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/117>
2020-05-01 10:42:17 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks
The old API is preserved now and new API was added that provides the
additional parameter to the callback.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
2020-04-28 23:33:49 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Store the timeout source by pointer instead of id
That way we don't have to retrieve it again from the main context when
destroying it but can directly do so.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
2020-04-28 23:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Clean up watch/watch context and related state consistently
And assert that it was cleaned up properly before the client is
finalized. If something is still around when the client is shut down
then something went very wrong before.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
2020-04-27 23:25:22 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/rtspserver.c:
rtsp-client: Combine the pre-session and post-session timeout
They previously used the same state but different mechanisms and
functions, which was difficult to follow, error prone and simply
confusing.
Also adjust the test for the post-session timeout a bit to be less racy
now that the timing has slightly changed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
2020-04-27 19:47:15 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Don't ever close the client connection directly when a session is torn down
There might be other sessions that are running over the same RTSP
connection and we should not simply close the client directly if one of
them is torn down.
By default the connection will be closed once the client closes it or
the OS does. This behaviour can be adjusted with the
post-session-timeout property, which allows to close it automatically
from the server side after all sessions are gone and the given timeout
is reached.
This reverts the previous commit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/115>
2020-04-27 13:49:55 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: If the TEARDOWN response can be sent directly, directly close the client
Instead of closing it never at all. Previously there was only code that
closed the client asynchronously if sending the response happened
asynchrously at a later time.
Thanks to Christian M for debugging this issue.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/102
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/114>
2020-03-23 14:51:28 +0100 Michael Olbrich <m.olbrich@pengutronix.de>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: use mcast_udpsink[0] last-sample if available for rtpinfo
Otherwise no sink is found for multicast sreams and the less accurate
fallback is used to determine the current sequence number and timestamp.
2020-03-23 16:06:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
rtsp-auth: Fix NULL pointer dereference when handling an invalid basic Authorization header
When using the basic authentication scheme, we wouldn't validate that
the authorization field of the credentials is not NULL and pass it on
to g_hash_table_lookup(). g_str_hash() however is not NULL-safe and will
dereference the NULL pointer and crash.
A specially crafted (read: invalid) RTSP header can cause this to
happen.
As a solution, check for the authorization to be not NULL before
continuing processing it and if it is simply fail authentication.
This fixes CVE-2020-6095 and TALOS-2020-1018.
Discovered by Peter Wang of Cisco ASIG.
2020-03-09 14:17:34 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Use watch_context before unref
Move the usage of priv->watch_context to beginning of function
gst_rtsp_client_finalize. Instead of use it after
g_main_context_unref (priv->watch_context).
2020-02-14 14:59:43 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix deadlock on transport removal
We cannot take the RTSPStream lock while holding a transport backlog
lock, as remove_transport may be called externally, which will
take first the RTSPStream lock then the transport backlog lock.
2020-02-14 14:59:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: clear backlog when removing transport
This ensures we don't end up calling any of transports' callbacks
with a potentially unreffed user_data (in practice, a client that
may have been removed)
2020-02-06 22:46:18 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: marshal calls to send_tcp_message to a single thread
In order to address the race condition pointed out at
https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/merge_requests/108#note_403579
we get rid of the send thread pool, and instead spawn and manage
a single thread to pull samples from app sinks and add them to
the transport's backlogs.
Additionally, we now also always go through the backlogs in order
to simplify the logic.
2020-02-05 20:28:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: properly protect TCP backlog access
Fixes #97
We cannot hold stream->lock while pushing data, but need
to consistently check the state of the backlog both from
the send_tcp_message function and the on_message_sent function,
which may or may not be called from the same thread.
This commit introduces internal API to allow for potentially
recursive locking of transport streams, addressing a race
condition where the RTSP stream could push items out of order
when popping them from the backlog.
2020-02-22 00:41:32 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Sink pipeline in gst_rtsp_media_take_pipeline()
It's taken ownership of by the media, and returned with `transfer none`
from the GstRTSPMedia::create_pipeline() vfunc. If we don't sink it
first then any bindings will wrongly take ownership of the pipeline once
it arrives in bindings code.
2020-02-05 16:51:14 +0100 Bastian Bouchardon <bastian.bouchardon@gmail.com>
* examples/test-onvif-client.c:
Add initialization for context and params (gchar *) Insert define (DEFAULT_*) into help to have to modify only the constants
2020-02-03 12:30:14 +0000 Marc Leeman <marc.leeman@gmail.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix default latency
2020-01-15 17:06:41 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: make closing more thread safe
+ Take the watch lock prior to using priv->watch
+ Flush both the watch and connection before closing / unreffing
gst_rtsp_connection_close() is not threadsafe on its own, this is
a workaround at the client level, where we control both the watch
and the connection
2020-01-23 16:41:26 +0200 Jordan Petridis <jordan@centricular.com>
* gst/rtsp-server/rtsp-latency-bin.c:
rtsp-latency-bin: replace G_TYPE_INSTANCE_GET_PRIVATE as it's been deprecated
from glib
```
Deprecated: 2.58: Use %G_ADD_PRIVATE and the generated
`your_type_get_instance_private()` function instead
```
2019-12-17 16:08:19 +0100 Zoltán Imets <zoltani@axis.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/rtspserver.c:
rtsp-client: add property post-session-timeout
This is a TCP connection timeout for client connections, in seconds.
If a positive value is set for this property, the client connection
will be kept alive for this amount of seconds after the last session
timeout. For negative values of this property the connection timeout
handling is delegated to the system (just as it was before).
Fixes #83
2020-01-11 22:58:48 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: check for NULL transports prior to ref'ing
2020-01-09 14:10:44 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix checking of TCP backpressure
The internal index of our appsinks, while it can be used to
determine whether a message is RTP or RTCP, is not necessarily
the same as the interleaved channel. Let the stream-transport
determine the channel to check backpressure for, the same way
it determines the channel according to whether it is sending
RTP or RTCP.
2019-12-10 19:16:51 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: Butcher the file to please gst-indent in the CI
This should be reverted once the CI has an updated gst-indent.
2019-12-10 18:39:32 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtsp-session & client: Remove deprecated GTimeVal
GTimeVal won't work past 2038
2019-12-12 17:56:18 +0100 Nicola Murino <nicola.murino@gmail.com>
* gst/rtsp-server/rtsp-auth.c:
rtsp-auth: fix default token leak
2019-12-09 14:17:05 +0100 Adam x Nilsson <adamni@axis.com>
* gst/rtsp-sink/gstrtspclientsink.c:
gstrtspclientsink: unref transports when closing bin
Fixes #91
2019-12-06 10:44:35 +0100 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Force seek when flush flag is set
The commit "rtsp-client: define all seek accuracy flags from
setup_play_mode" changed the behaviour of when doing a seek.
Before that commit, having the flush flag set would result in a seek
(forced seek).
Even if no seek was needed. One reason to force seek is to flush old buffers
created in Describe requests.
Thus adding force seek also for flush flag will result in play request
with fresh buffers.
2019-11-21 17:12:45 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Revitalize dead code
Leftover from 65d9aa327cd1844934836249cd4463edf09c725d
CID: 1455379
2019-11-27 15:22:35 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Don't try to use non-initialized values
Only attempt to use the various timing values iif gst_rtsp_stream_get_info()
returns TRUE. Also avoid the whole clock signalling block if we're not
dealing with senders.
CID: 1439524
CID: 1439536
CID: 1439520
2019-11-01 12:01:41 +0100 Adam x Nilsson <adamni@axis.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/stream.c:
rtsp-stream: Removing invalid transports returns false
When removing transports an assertion was that the transports passed in
for removal are present in the list, however that can't be assumed.
As an example if a transport was removed from a thread running
send_tcp_message, the main thread can try to remove the same transport
again if it gets a handle_pause_request. This will not effect the
transport list but it will effect n_tcp_transports as it will be
decrement and then have the wrong value.
2019-11-06 14:17:48 +0100 Zoltán Imets <zoltani@axis.com>
* tests/check/gst/client.c:
client test: add scale and speed negative tests
Negative tests for scale and speed should be done as well, verify that
the response code is "400 Bad request" when a bad request is done.
2019-08-29 07:34:26 +0200 Niels De Graef <nielsdegraef@gmail.com>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-sink/gstrtspclientsink.c:
Don't pass default GLib marshallers for signals
By passing NULL to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.
Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
2019-09-05 19:51:06 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-mount-points.c:
GstRTSPMountPoints: Remove any existing factory before adding a new one
The documentation of gst_rtsp_mount_points_add_factory() says "Any
previous mount point will be freed" which was true when it was
implemented using a GHashTable. But in 2012 it got rewrote using a
GSequence and since then it could have 2 factories for the same path.
Which one gets used is random, depending on the sorting order of 2
identical items.
2019-10-15 19:08:32 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server-internal.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
stream: refactor TCP backpressure handling
The previous implementation stopped sending TCP messages to
all clients when a single one stopped consuming them, which
obviously created problems for shared media.
Instead, we now manage a backlog in stream-transport, and slow
clients are removed once this backlog exceeds a maximum duration,
currently hardcoded.
Fixes #80
2019-10-18 00:42:12 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: build gir even when cross-compiling if introspection was enabled explicitly
This can be made to work in certain circumstances when
cross-compiling, so default to not building g-i stuff
when cross-compiling, but allow it if introspection was
enabled explicitly via -Dintrospection=enabled.
See gstreamer/gstreamer#454 and gstreamer/gstreamer#381.
2019-10-18 09:19:59 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: clean up comment extra-timeout
2019-10-17 12:15:42 +0200 Muhammet Ilendemli <mi@tailored-apps.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Generate correct URI for MIKEY in ANNOUNCE responses
Instead of hardcoding the URI, take the actual URI (and especially the correct port)
from the RTSP context.
Fixes #84
2019-10-16 13:20:54 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-client: Lock shared media
For shared media we got race conditions. Concurrently rtsp clients might
suspend or unsuspend the shared media and thus change the state without
the clients expecting that.
By introducing a lock that can be taken by callers such as rtsp_client
one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media,
to handle the media sequentially thus allowing one client to finish its
rtsp call before another client calls on the same media.
https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86
Fixes #86
2019-10-15 07:33:29 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: add property extra-timeout
Extra time to add to the timeout, in seconds. This only
affects the time until a session is considered timed out
and is not signalled in the RTSP request responses.
Only the value of the timeout property is signalled in the
request responses.
2019-10-07 12:13:47 +0200 Adam x Nilsson <adamni@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream : fix race condition in send_tcp_message
If one thread is inside the send_tcp_message function and are done
sending rtp or rtcp messages so the n_outstanding variable is zero
however have not exit the loop sending the messages. While sending its
messages, transports have been added or removed to the transport list,
so the cache should be updated. If now an additional thread comes to
the function send_tcp_message and trying to send rtp messages it will
first destroy the rtp cache that is still being iterated trough by the
first thread.
Fixes #81
2019-05-24 14:32:50 +0200 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
* .gitmodules:
* Makefile.am:
* autogen.sh:
* common:
* configure.ac:
* docs/.gitignore:
* examples/.gitignore:
* examples/Makefile.am:
* gst/Makefile.am:
* gst/rtsp-server/.gitignore:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-sink/Makefile.am:
* pkgconfig/.gitignore:
* pkgconfig/Makefile.am:
* tests/.gitignore:
* tests/Makefile.am:
* tests/check/Makefile.am:
Remove autotools build
Replaced by Meson.
Maybe we can now use the meson pkgconfig module
for .pc files? (Does it support uninstalled now?)
2019-10-07 10:27:36 +0200 Göran Jönsson <goranjn@axis.com>
* tests/check/gst/client.c:
client: fix test mem leak in attach_rate_tweaking_probe
2019-10-07 10:14:52 +0200 Göran Jönsson <goranjn@axis.com>
* tests/check/gst/media.c:
media: remove memleak in test test_media_seek
2019-10-07 10:07:54 +0200 Göran Jönsson <goranjn@axis.com>
* tests/check/gst/rtspserver.c:
rtspserver: Remove memleak in test test_double_play
2019-09-17 13:45:57 +0200 Adam x Nilsson <adamni@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Use lock in gst_rtsp_media_is_receive_only
2018-10-29 17:02:41 +0100 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/rtspserver.c:
rtsp-media: Unblock all streams
When unsuspending and going to PLAYING, unblock all streams instead of
only those that are linked (the linked streams are the ones for which
SETUP has been called). GST_FLOW_NOT_LINKED will be returned when
pushing buffers on unlinked streams.
This change is because playback using single-threaded demuxers like
matroska-demux could be blocked if SETUP was not called for all media.
Demuxers that use GstFlowCombiner (including gstoggdemux, gstavidemux,
gstflvdemux, qtdemux, and matroska-demux) will handle
GST_FLOW_NOT_LINKED automatically.
Fixes #39
2019-09-11 07:08:37 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/rtspserver.c:
rtsp-media: Wait on async when needed.
Wait on asyn-done when needed in gst_rtsp_media_seek_trickmode.
In the unit test the pause from adjust_play_mode will cause a preroll
and after that async-done will be produced.
Without this patch there are no one consuming this async-done and when
later when seek fluch is done in gst_rtsp_media_seek_trickmode then it
wait for async-done. But then it wrongly find the async-done prodused by
adjus_play_mode and continue executing without waiting for the preroll
to finish.
2019-09-30 15:13:15 +0200 Kristofer Bjorkstrom <kristofb@pc36402-1937.se.axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: RTP Info when completed_sender
Change condition that should be fulfilled regarding RTPInfo.
Replace !gst_rtsp_media_is_receive_only with
gst_rtsp_media_has_completed_sender. It is more correct to actually look
for a sender pipeline that is complete. Only then a RTPInfo should
exist.
gst_rtsp_media_is_receive_only gives different answears depending on
state of server.
If Describe is called wth URL+options for backchannel SDP will give only
audio and only backchannel a=sendonly
If Describe is called on URL+options that gives both audio and video
direction from server to client, pipelines are created. Thus
receive_only will return false, even though Setup only would setup
backchannel.
RTP-Info is only for outgoing streams. Thus one should look if outgoing
streams are complete.
2019-09-25 09:14:08 +0000 Kristofer <kristofer.bjorkstrom@axis.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
rtsp-client: RTP Info exists conditionally in PLAY
If RTP Info is missing and it is not a receiver only, eg. audio
backchannel. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR.
In rfc2326 it says RTP-info is req. but in RFC7826 it is conditional.
Since 1.14 there is audio backchannel support. Thus RTP-info is
conditional now. When audio backchannel only mode, there is no RTP-info.
Fixes #82
2019-09-05 16:23:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-client.c:
test-onvif-client: remove unused query
2019-08-30 14:00:52 +0200 Kristofer Björkström <kristofb@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: RTP Info must exist in PLAY response
If RTP Info is missing. Then return GST_RTSP_STS_INTERNAL_SERVER_ERROR
Fixes #76
2019-08-29 21:37:24 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-client.c:
test-onvif-client: perform accurate seeks
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/336
Also, modify how we compute the position: position queries in
PAUSED mode fail to account for the newly-prerolled frame, leading
to frame skips when performing seeks in that state. Instead,
compute the current position from the last sample.
2019-08-21 14:57:25 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* tests/check/gst/rtspserver.c:
Use complete streams for scale and speed.
Without this patch it's always stream0 that is used to get segment event
that is used to set scale and speed. This even if client not doing SETUP
for stream0. At least in suspend mode reset this not working since then
it's just random if send_rtp_sink have got any segment event. There are
no check if send_rtp_sink for stream0 got any data before media is
prerolled after PLAY request.
2019-08-26 22:24:12 +1000 Matthew Waters <matthew@centricular.com>
* examples/test-onvif-server.c:
* examples/test-onvif-server.h:
examples/onvif-server: fix werror build with clang
../subprojects/gst-rtsp-server/examples/test-onvif-server.c:346:65: warning: implicit conversion from enumeration type 'const GstSegmentFlags' to different enumeration type 'GstSeekFlags' [-Wenum-conversion]
self->incoming_segment->format, self->incoming_segment->flags,
~~~~~~~~~~~~~~~~~~~~~~~~^~~~~
../subprojects/gst-rtsp-server/examples/test-onvif-server.c:53:1: warning: unused function 'REPLAY_IS_BIN' [-Wunused-function]
G_DECLARE_FINAL_TYPE (ReplayBin, replay_bin, REPLAY, BIN, GstBin);
^
/usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
^
<scratch space>:77:1: note: expanded from here
REPLAY_IS_BIN
^
../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_FACTORY' [-Wunused-function]
G_DECLARE_FINAL_TYPE (OnvifFactory, onvif_factory, ONVIF, FACTORY,
^
/usr/include/glib-2.0/gobject/gtype.h:1405:33: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
static inline ModuleObjName * MODULE##_##OBJ_NAME (gpointer ptr) { \
^
<scratch space>:9:1: note: expanded from here
ONVIF_FACTORY
^
../subprojects/gst-rtsp-server/examples/test-onvif-server.c:525:1: warning: unused function 'ONVIF_IS_FACTORY' [-Wunused-function]
/usr/include/glib-2.0/gobject/gtype.h:1407:26: note: expanded from macro 'G_DECLARE_FINAL_TYPE'
static inline gboolean MODULE##_IS_##OBJ_NAME (gpointer ptr) { \
^
<scratch space>:12:1: note: expanded from here
ONVIF_IS_FACTORY
^
2019-08-23 16:21:36 +1000 Matthew Waters <matthew@centricular.com>
* docs/meson.build:
meson: Don't generate doc cache when no plugins are enabled
Fixes gst-build with -Dauto-features=disabled -Drtsp_server=enabled
2019-08-16 13:38:01 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* examples/test-onvif-client.c:
test-onvif-client: stdin is not defined in MSVC
2019-08-12 18:03:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: add missing Since tag
2019-08-08 15:52:53 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-client.c:
test-onvif-client: STDIN_FILENO is not portable
If not defined, define it to _fileno(stdin) on Windows, 0
everywhere else
2019-08-07 21:04:33 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/test-onvif-server.c:
test-onvif-server: downgrade logging
2019-07-27 05:14:49 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/meson.build:
* examples/test-onvif-client.c:
* examples/test-onvif-server.c:
examples: add ONVIF client / server example
2019-07-27 05:14:28 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-client: define all seek accuracy flags from setup_play_mode
We then pass those to adjust_play_mode, which needs to operate
on the "final" seek flags, as previously the code in rtsp-media
was assuming that accuracy seek flags (accurate / key_unit) should
not be set if the flags passed to the seek method were already set.
2019-07-22 19:32:43 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Try to get dynamic payloaders by name from their bin first
First try "pay", then "pay_%s" (where %s == pad name). And only then
fall back to the code that simply takes the first payloader that is
found.
The current code usually works (but is racy) because it will always take
the payloader that was last added (due to g_list_prepend() when adding
elements) in pad-added and that's usually the correct one. But if a new
payloader is added between pad-added and us trying to get it, we would
get the wrong payloader.
2019-07-17 15:51:08 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* tests/check/gst/client.c:
client test: expect any port in transport
setup_multicast_client sets a 5000-5010 range for the client
ports, it is incorrect to expect the transport to always use
5000-5001
Fixes #73
2019-07-15 17:06:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* tests/check/gst/onvif.c:
onvif tests: use g_cond_wait() correctly
g_cond_wait() has to be called in a loop until required conditions
are met
Fixes #71
2019-06-28 12:28:41 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Not wait on receiver streams when pre-rolling
Without this patch there are problem pre-rolling when using audio back
channel.
Without this patch a probe will be created for all streams including
the stream for audio backchannel. To pre-roll all this pads have to
receive data. Since the stream for audio backchannel is a receiver this
will never happen.
The solution is to never create any probes for streams that are for
incomming data and instead set them as blocking already from beginning.
2019-06-25 13:19:44 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-onvif-media-factory.c:
* gst/rtsp-server/rtsp-onvif-media.c:
onvif-media: fix "void function returning a value" compiler warning
2019-06-12 22:19:27 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: make sure streams are blocked when sending seek
The recent ONVIF work exposed a race condition when dealing with
multiple streams: one of the sinks may preroll before other streams
have started flushing. This led to the pipeline posting async-done
prematurely, when some streams were actually still in the middle
of performing a flushing seek. The newly-added code looks up a
sticky segment event on the first stream in order to respond to
the PLAY request with accurate Scale and Speed headers. In the
failure condition, the first stream was flushing, and thus had
no sticky segment event, leading to the PLAY request failing,
and in turn the test.
2019-06-07 10:51:19 +0200 Michael Bunk <bunk@iat.uni-leipzig.de>
* docs/README:
* gst/rtsp-server/rtsp-media-factory-uri.h:
Fix typos
2019-04-05 00:48:07 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-onvif-client.c:
* gst/rtsp-server/rtsp-onvif-client.h:
* gst/rtsp-server/rtsp-onvif-media-factory.c:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
* gst/rtsp-server/rtsp-onvif-media.c:
* gst/rtsp-server/rtsp-onvif-server.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/media.c:
* tests/check/gst/onvif.c:
* tests/check/meson.build:
onvif: Implement and test the Streaming Specification
https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
2018-11-05 15:34:20 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: add gst_rtsp_client_get_stream_transport()
This will be used in the onvif tests in order to validate the
data transmitted over TCP: for streaming to continue after a
data message has been provided to client->send_func, the client
is responsible for marking the message as sent on the relevant
stream transport.
2018-11-07 00:33:01 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
client: Scale implies TRICK_MODE
2018-11-07 00:32:29 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
client: compare booleans, not pointers to them
2018-11-13 21:28:45 +0100 Nikita Bobkov <NikitaDBobkov@gmail.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/media.c:
Reverse playback support
GStreamer plays segment from stop to start when doing reverse playback.
RTSP implies that media should be played from start of Range header to
its stop. Hence we swap start and stop times before passing them to
gst_element_seek.
Also make gst_rtsp_stream_query_stop always return value that can be
used as stop time of Range header.
2018-10-12 08:53:04 +0200 Branko Subasic <branko@subasic.net>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* tests/check/gst/client.c:
rtsp-client: add support for Scale and Speed header
Add support for the RTSP Scale and Speed headers by setting the rate in
the seek to (scale*speed). We then check the resulting segment for rate
and applied rate, and use them as values for the Speed and Scale headers
respectively.
https://bugzilla.gnome.org/show_bug.cgi?id=754575
2018-10-01 18:51:49 +0200 Branko Subasic <branko@subasic.net>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: allow sub classes to adjust the seek
Adds a new virtual function, adjust_play_mode(), that allows
sub classes to adjust the seek done on the media. The sub class can
modify the values of the the seek flags and the rate.
https://bugzilla.gnome.org/show_bug.cgi?id=754575
2018-09-27 19:09:01 +0200 Branko Subasic <branko@subasic.net>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/media.c:
rtsp-media: allow specifying rate when seeking
Add new function gst_rtsp_media_seek_full_with_rate() which allows the
caller to specify the rate for the seek. Also added functions in
rtsp-stream and rtsp-media for retreiving current rate and applied rate.
https://bugzilla.gnome.org/show_bug.cgi?id=754575
2019-06-02 21:39:33 +0200 Niels De Graef <niels.degraef@barco.com>
* configure.ac:
* meson.build:
meson: Bump minimal GLib version to 2.44
This means we can use some newer features and get rid of some
boilerplate code using the G_DECLARE_* macros.
As discussed on IRC, 2.44 is old enough by now to start depending on it.
2019-05-31 18:53:36 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/.gitignore:
* docs/libs/Makefile.am:
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* docs/libs/gst-rtsp-server.types:
docs: remove obsolete gtk-doc related files
2019-05-29 23:20:09 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
doc: remove xml from comments
2019-05-16 09:23:53 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/gst_plugins_cache.json:
* docs/meson.build:
docs: Stop building the doc cache by default
And update the cache
Fixes https://gitlab.freedesktop.org/gstreamer/gst-docs/issues/36
2019-05-13 22:59:57 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/gst_plugins_cache.json:
docs: Update plugins documentation cache
2019-04-23 12:30:02 -0400 Thibault Saunier <tsaunier@igalia.com>
* docs/meson.build:
* gst/rtsp-server/rtsp-context.c:
* gst/rtsp-server/rtsp-session-pool.c:
doc: Fix some docstrings
2018-10-22 11:29:24 +0200 Thibault Saunier <tsaunier@igalia.com>
* .gitignore:
* Makefile.am:
* configure.ac:
* docs/Makefile.am:
* docs/gst_plugins_cache.json:
* docs/index.md:
* docs/meson.build:
* docs/plugin-index.md:
* docs/plugin-sitemap.txt:
* docs/sitemap.md:
* docs/sitemap.txt:
* docs/version.entities.in:
* gst/rtsp-server/meson.build:
* gst/rtsp-sink/meson.build:
* meson.build:
* meson_options.txt:
docs: Port to hotdoc
2019-04-23 15:09:34 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-server: Fix various Since markers
2019-04-23 15:01:32 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: Add various Since: 1.14 markers
2019-04-23 14:38:05 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: Add various missing Since: 1.16 markers
2019-04-15 20:54:24 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Set async-handling=false for the internal bins
Without this we can easily run into a race condition with async state changes:
- the pipeline is doing an async state change
- we set the internal bins to PLAYING but that's ignored because an
async state change is currently pending
- the async state change finishes but does not change the state of the
internal bins because of locked_state==TRUE
- the internal bins stay in PAUSED forever
2019-04-15 20:51:30 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Use write_messages() API to send buffer lists in one go
And to write messages with multiple memories also via writev().
2019-03-27 16:21:03 +0100 Kristofer Bjorkstrom <kristofb@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server-object.h:
* gst/rtsp-server/rtsp-server.c:
rtsp-client: Handle Content-Length limitation
Add functionality to limit the Content-Length.
API addition, Enhancement.
Define an appropriate request size limit and reject requests
exceeding the limit with response status 413 Request Entity Too Large
Related to !182
2019-04-19 10:40:29 +0100 Tim-Philipp Müller <tim@centricular.com>
* RELEASE:
* configure.ac:
* meson.build:
Back to development
=== release 1.16.0 ===
2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.16.0
2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Notify the stream transport about each written message
Otherwise it will never try to send us the next one: it tries to keep
exactly one message in-flight all the time.
In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
in the client sink we always write data out synchronously.
2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp_server: Free thread pool before clean transport cache
If not waiting for free thread pool before clean transport caches, there
can be a crash if a thread is executing in transport list loop in
function send_tcp_message.
Also add a check if priv->send_pool in on_message_sent to avoid that a
new thread is pushed during wait of free thread pool. This is possible
since when waiting for free thread pool mutex have to be unlocked.
=== release 1.15.90 ===
2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.15.90
2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Add support for GCM (RFC 7714)
Follow-up to !198
2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
* gst/rtsp-server/rtsp-session-pool.c:
session pool: fix missing klass-> in klass->create_session
2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
g-i: pass --quiet to g-ir-scanner
This suppresses the annoying 'g-ir-scanner: link: cc ..' output
that we get even if everything works just fine.
We still get g-ir-scanner warnings and compiler warnings if
we pass this option.
2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
g-i: silence 'nested extern' compiler warnings when building scanner binary
We need a nested extern in our init section for the scanner binary
so we can call gst_init to make sure GStreamer types are initialised
(they are not all lazy init via get_type functions, but some are in
exported variables). There doesn't seem to be any other mechanism to
achieve this, so just remove that warning, it's not important at all.
2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
rtsp-media: Handle set state when preparing.
Handle the situation when a call to gst_rtsp_media_set_state is done
when media status is preparing.
Also add unit test for this scenario.
The unit test simulate on a media level when two clients share a (live)
media.
Both clients have done SETUP and got responses. Now client 1 is doing
play and client 2 is just closing the connection.
Then without patch there are a problem when
client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
And client2 is doing closing connection we can end up in a call
to gst_rtsp_media_set_state when
priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
shut down media is jumped over .
With this patch and this scenario we wait until
priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
execute after that and now we will execute the logic for
shut down media.
2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
* NEWS:
* RELEASE:
* configure.ac:
* meson.build:
Back to development
=== release 1.15.2 ===
2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.15.2
2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/client.c:
rtsp-media: Fix multicast use case with common media
Use case
client 1: SETUP
client 1: PLAY
client 2: SETUP
client 1: TEARDOWN
client 2: PLAY
client 2: TEARDOWN
2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-server: remove recursive behavior
Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Only allow to set either a send_func or send_messages_func but not both
And route all messages through the send_func if no send_messages_func
was provided.
We otherwise break backwards compatibility.
2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-stream.c:
rtsp-client: Add support for sending buffer lists directly
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-sink/gstrtspclientsink.c:
rtsp-server: Add support for buffer lists
This adds new functions for passing buffer lists through the different
layers without breaking API/ABI, and enables the appsink to actually
provide buffer lists.
This should already reduce CPU usage and potentially context switches a
bit by passing a whole buffer list from the appsink instead of
individual buffers. As a next step it would be necessary to
a) Add support for a vector of data for the GstRTSPMessage body
b) Add support for sending multiple messages at once to the
GstRTSPWatch and let it be handled internally
c) Adding API to GOutputStream that works like writev()
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: Fix crash in close handler
The close handler could trigger a crash because it invalidated the
watch_context while still leaving a source attached to it which would be
cleaned up at a later point.
2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Use cached address when allocating sockets
If an address/port was previously decided upon (ex: multicast in the
SDP), then use that instead of re-creating another one
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix race codition in finish_unprepare
The previous fix for race condition around finish_unprepare where the
function could be called twice assumed that the status wouldn't change
during execution of the function. This assumption is incorrect as the
state may change, for example if an error message arrives from the
pipeline bus.
Instead a flag keeping track on whether the finish_unprepare function
is currently executing is introduced and checked.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
=== release 1.15.1 ===
2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.15.1
2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
Add source elements to the pipeline before activation
In plug_src we changed the element state before adding it to
the owner container. This prevented the pipeline from intercepting
a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
to assign a custom task pool.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
* common:
Automatic update of common submodule
From ed78bee to 59cb678
2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
* examples/test-appsrc.c:
examples: test-appsrc: fix coding style error
2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
* examples/test-appsrc.c:
examples: test-appsrc: fix buffer leak
2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Update priv->blocked when linked streams are unblocked.
Media is considered to be blocked when all streams that belong to
that media are blocked.
This patch solves the problem of inconsistent updates of
priv->blocked that are not synchronized with the media state.
2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't block streams before seeking
Before the seek operation is performed on media, it's required that
its pipeline is prepared <=> the pipeline is in the PAUSED state.
At this stage, all transport parts (transport sinks) have been successfully
added to the pipeline and there is no need for blocking the streams.
2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
* tests/check/gst/rtspserver.c:
tests: rtspserver: Add shared media test case for TCP
2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Use seqnum-offset for rtpinfo
The sequence number in the rtpinfo is supposed to be the first RTP
sequence number. The "seqnum" property on a payloader is supposed to be
the number from the last processed RTP packet. The sequence number for
payloaders that inherit gstrtpbasepayload will not be correct in case of
buffer lists. In order to fix the seqnum property on the payloaders
gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
"seqnum-offset" from the "stats" property contains the value of the
very first RTP packet in a stream. The server will, however, try to look
at the last simple in the sink element and only use properties on the
payloader in case there no sink elements yet, and by looking at the last
sample of the sink gives the server full control of which RTP packet it
looks at. If the payloader does not have the "stats" property, "seqnum"
is still used since "seqnum-offset" is only present in as part of
"stats" and this is still an issue not solved with this patch.
Needed for gst-plugins-base!17
2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Plug memory leak
Attaching a GSource to a context will increase the refcount. The idle
source will never be free'd since the initial reference is never
dropped.
2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
* .gitlab-ci.yml:
Add Gitlab CI configuration
This commit adds a .gitlab-ci.yml file, which uses a feature
to fetch the config from a centralized repository. The intent is
to have all the gstreamer modules use the same configuration.
The configuration is currently hosted at the gst-ci repository
under the gitlab/ci_template.yml path.
Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
* .gitmodules:
* gst-rtsp-server.doap:
Update git locations to gitlab
2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/meson.build:
meson: add new onvif types
2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/meson.build:
Add ONVIF subclass headers to the installed headers in meson.build too
2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-server-object.h:
* gst/rtsp-server/rtsp-server.h:
rtsp-server: Declare GstRTSPServer struct before anything else
It's needed by all kinds of other headers, including the ones that are
required for defining the GstRTSPServer struct itself and its API.
2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-onvif-client.h:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
* gst/rtsp-server/rtsp-onvif-media.h:
* gst/rtsp-server/rtsp-onvif-server.h:
Mark all ONVIF-specific subclasses as Since 1.14
2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/meson.build:
* gst/rtsp-server/rtsp-context.h:
* gst/rtsp-server/rtsp-onvif-server.c:
* gst/rtsp-server/rtsp-onvif-server.h:
* gst/rtsp-server/rtsp-server-object.h:
* gst/rtsp-server/rtsp-server-prelude.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session.h:
Include ONVIF types from single-include rtsp-server.h
... by actually making it a single-include header and moving everything
related to the GstRTSPServer type to rtsp-server-object.h instead.
Otherwise there are too many circular includes.
https://bugzilla.gnome.org/show_bug.cgi?id=797361
2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-latency-bin.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: use idle source in on_message_sent
When the underlying layers are running on_message_sent, this sometimes
causes the underlying layer to send more data, which will cause the
underlying layer to run callback on_message_sent again. This can go on
and on.
To break this chain, we introduce an idle source that takes care of
sending data if there are more to send when running callback
https://bugzilla.gnome.org/show_bug.cgi?id=797289
2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Remove timeout GSource on cleanup
Avoids ending up with races where a timeout would still be around
*after* a client was gone. This could happen rather easily in
RTSP-over-HTTP mode on a local connection, where each RTSP message
would be sent as a different HTTP connection with the same tunnelid.
If not properly removed, that timeout would then try to free again
a client (and its contents).
2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/Makefile.am:
autotools: fix distcheck
2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/meson.build:
* gst/rtsp-server/rtsp-latency-bin.c:
* gst/rtsp-server/rtsp-latency-bin.h:
* gst/rtsp-server/rtsp-onvif-media.c:
onvif: encapsulate onvif part into a bin
...and thus do not let onvif affect pipelines latency
https://bugzilla.gnome.org/show_bug.cgi?id=797174
2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
* tests/check/gst/client.c:
tests: client: Avoid bind() failures in tests
https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/mediafactory.c:
New property for socket binding to mcast addresses
By default the multicast sockets are bound to INADDR_ANY,
as it's not allowed to bind sockets to multicast addresses
in Windows. This default behaviour can be changed by setting
bind-mcast-address property on the media-factory object.
https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/meson.build:
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-context.c:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-server-prelude.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-token.c:
* meson.build:
libs: fix API export/import and 'inconsistent linkage' on MSVC
Export rtsp-server library API in headers when we're building the
library itself, otherwise import the API from the headers.
This fixes linker warnings on Windows when building with MSVC.
Fix up some missing config.h includes when building the lib which
is needed to get the export api define from config.h
https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: Add missing break statements
This resulted in warnings/assertions whenever one accessed the
max-mcast-ttl property.
CID #1439515
CID #1439523
2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
* meson_options.txt:
meson: add gobject-cast-checks, glib-asserts, glib-checks options
2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/meson.build:
* meson_options.txt:
* tests/check/meson.build:
meson: add option to disable build of rtspclientsink plugin
2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson_options.txt:
meson: re-arrange options
2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* meson.build:
* meson_options.txt:
* tests/check/meson.build:
* tests/meson.build:
meson: Use feature option for tests option
This was somehow missed the last time around.
2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* gst/rtsp-server/meson.build:
* meson.build:
meson: Maintain macOS ABI through dylib versioning
Requires Meson 0.48, but the feature will be ignored on older versions
so it's safe to add it without bumping the requirement.
Documentation:
https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
* gst/rtsp-sink/meson.build:
* meson.build:
meson: add pkg-config file for the rtspclientsink plugin
2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
rtsp-client: Avoid reuse of channel numbers for interleaved
If a (strange) client would reuse interleaved channel numbers in
multiple SETUP requests, we should not accept them. The channel
numbers are used for looking up stream transports in the
priv->transports hash table, and transports disappear from the table
if channel numbers are reused.
RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
server to change the channel numbers suggested by the client.
https://bugzilla.gnome.org/show_bug.cgi?id=796988
2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
* tests/check/gst/client.c:
rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
Allow regex for matching transport header against expected pattern.
https://bugzilla.gnome.org/show_bug.cgi?id=796988
2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* tests/check/meson.build:
meson: There is no gstreamer-plugins-good-1.0.pc
There is no installed version of that, only an uninstalled version.
2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/stream.c:
Fix indentation again
2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/stream.c:
stream: Added a list of multicast client addresses
When media is shared, the same media stream can be sent
to multiple multicast groups. Currently, there is no API
to retrieve multicast addresses from the stream.
When calling gst_rtsp_stream_get_multicast_address() function,
only the first multicast address is returned.
With this patch, each multicast destination requested in SETUP
will be stored in an internal list (call to
gst_rtsp_stream_add_multicast_client_address()).
The list of multicast groups requested by the clients can be
retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
There still exist some problems with the current implementation
in the multicast case:
1) The receiving part is currently only configured with
regard to the first multicast client (see
https://bugzilla.gnome.org/show_bug.cgi?id=796917).
2) Secondly, of security reasons, some constraints should be
put on the requested multicast destinations (see
https://bugzilla.gnome.org/show_bug.cgi?id=796916).
Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
stream: Choose the maximum ttl value provided by multicast clients
The maximum ttl value provided so far by the multicast clients
will be chosen and reported in the response to the current
client request.
Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
rtsp-stream: Don't require address pool in the transport specific case
If "transport.client-settings" parameter is set to true, the client is
allowed to specify destination, ports and ttl.
There is no need for pre-configured address pool.
Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
client: Don't reserve multicast address in the client setting case
When two multicast clients request specific transport
configurations, and "transport.client-settings" parameter is
set to true, it's wrong to actually require that these two
clients request the same multicast group.
Removed test_client_multicast_invalid_transport_specific test
cases as they wrongly require that the requested destination
address is supposed to be present in the address pool, also in
the case when "transport.client-settings" parameter is set to true.
Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/mediafactory.c:
Add new API for setting/getting maximum multicast ttl value
Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: avoid duplicating the first multicast client
In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
clients were dynamically added and removed to the multicast
udp sinks, as such we should no longer add a first client in
set_multicast_socket_for_udpsink
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
Revert "rtsp-stream: avoid duplicating the first multicast client"
This reverts commit 33570944401747f44d8ebfec535350651413fb92.
Commits where accidentially squashed together
2018-08-14 14:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/mediafactory.c:
Revert "Add new API for setting/getting maximum multicast ttl value"
This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
Commits where accidentially squashed together
2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
Revert "rtsp-stream: Don't require address pool in the transport specific case"
This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
Commits where accidentially squashed together
2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/stream.c:
Revert "stream: Choose the maximum ttl value provided by multicast clients"
This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
Commits where accidentially squashed together
2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
* examples/test-auth-digest.c:
examples: Fix indentation
2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/stream.c:
stream: Choose the maximum ttl value provided by multicast clients
The maximum ttl value provided so far by the multicast clients
will be chosen and reported in the response to the current
client request.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
rtsp-stream: Don't require address pool in the transport specific case
If "transport.client-settings" parameter is set to true, the client is
allowed to specify destination, ports and ttl.
There is no need for pre-configured address pool.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/mediafactory.c:
Add new API for setting/getting maximum multicast ttl value
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: avoid duplicating the first multicast client
In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
clients were dynamically added and removed to the multicast
udp sinks, as such we should no longer add a first client in
set_multicast_socket_for_udpsink
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
* gst/rtsp-server/Makefile.am:
rtsp-server: Add gstreamer-base gir dir in autotools
2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-client: always allocate both IPV4 and IPV6 sockets
multiudpsink does not support setting the socket* properties
after it has started, which meant that rtsp-server could no
longer serve on both IPV4 and IPV6 sockets since the patches
from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
merged.
When first connecting an IPV6 client then an IPV4 client,
multiudpsink fell back to using the IPV6 socket.
When first connecting an IPV4 client, then an IPV6 client,
multiudpsink errored out, released the IPV4 socket, then
crashed when trying to send a message on NULL nevertheless,
that is however a separate issue.
This could probably be fixed by handling the setting of
sockets in multiudpsink after it has started, that will
however be a much more significant effort.
For now, this commit simply partially reverts the behaviour
of rtsp-stream: it will continue to only create the udpsinks
when needed, as was the case since the patches were merged,
it will however when creating them, always allocate both
sockets and set them on the sink before it starts, as was
the case prior to the patches.
Transport configuration will only error out if the allocation
of UDP sockets fails for the actual client's family, this
also downgrades the GST_ERRORs in alloc_ports_one_family
to GST_WARNINGs, as failing to allocate is no longer
necessarily fatal.
https://bugzilla.gnome.org/show_bug.cgi?id=796875
2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* meson.build:
* meson_options.txt:
meson: Convert common options to feature options
These are necessary for gst-build to set options correctly. The
remaining automagic option is cgroup support in examples.
https://bugzilla.gnome.org/show_bug.cgi?id=795107
2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Slightly simplify locking
2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
Limit queued TCP data messages to one per stream
Before, the watch backlog size in GstRTSPClient was changed
dynamically between unlimited and a fixed size, trying to avoid both
unlimited memory usage and deadlocks while waiting for place in the
queue. (Some of the deadlocks were described in a long comment in
handle_request().)
In the previous commit, we changed to a fixed backlog size of 100.
This is possible, because we now handle RTP/RTCP data messages differently
from RTSP request/response messages.
The data messages are messages tunneled over TCP. We allow at most one
queued data message per stream in GstRTSPClient at a time, and
successfully sent data messages are acked by sending a "message-sent"
callback from the GstStreamTransport. Until that ack comes, the
GstRTSPStream does not call pull_sample() on its appsink, and
therefore the streaming thread in the pipeline will not be blocked
inside GstRTSPClient, waiting for a place in the queue.
pull_sample() is called when we have both an ack and a "new-sample"
signal from the appsink. Then, we know there is a buffer to write.
RTSP request/response messages are not acked in the same way as data
messages. The rest of the 100 places in the queue are used for
them. If the queue becomes full of request/response messages, we
return an error and close the connection to the client.
Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Use fixed backlog size
Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
Preparation for the next commit, which changes to a different way of
avoiding both deadlocks and unlimited memory usage with the watch
backlog.
2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: unref clock (if set) when finalizing
https://bugzilla.gnome.org/show_bug.cgi?id=796814
2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
* docs/libs/gst-rtsp-server-sections.txt:
rtsp-media: add gst_rtsp_media_*_set_clock to docs
https://bugzilla.gnome.org/show_bug.cgi?id=796814
2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: unref old clock when setting new clock
https://bugzilla.gnome.org/show_bug.cgi?id=796724
2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: unref clock in finalize
https://bugzilla.gnome.org/show_bug.cgi?id=796724
2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-onvif-media.c:
rtsp-onvif-media: fix g-ir-scanner warnings
2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
.gitignore: add another example binary
2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
* examples/meson.build:
meson: add new test-appsrc2 example to meson build
2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
* examples/Makefile.am:
examples: fix build of new test-appsrc2 example
Need to link against libgstapp-1.0.
2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-appsrc2.c:
examples: Add test-appsrc2
Add an example of feeding both audio and video into an RTSP
pipeline via appsrc.
2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
* gst/rtsp-server/rtsp-client.c:
client: Strip transport parts as whitespaces could be around commas
https://bugzilla.gnome.org/show_bug.cgi?id=758428
2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
Fix race when setting up source elements.
Since we set the source element(s) to PLAYING state before hooking
them up to the downstream funnel, it's possible for the source element
to receive packets before we actually get to linking it to the funnel,
in which case buffers would be pushed out on an unlinked pad, causing
it to error out and stop receiving more data.
We fix this by blocking the source's srcpad until we have linked it.
https://bugzilla.gnome.org/show_bug.cgi?id=796160
2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix mismatch between allowed and configured protocols
https://bugzilla.gnome.org/show_bug.cgi?id=796679
2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Emit a signal when the SRTP decoder is created
https://bugzilla.gnome.org/show_bug.cgi?id=778080
2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't require presence of sinks in _get_*_socket()
Transport specific sink elements are added to the pipeline
in PLAY request and sockets are already created in SETUP so
it's actually wrong to require the presence of sinks in
_get_*_socket() functions.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Update transport for multicast clients as well
If a multicast client requests different transport settings
than the existing one make sure that this new transport
configuruation is propagated to the multicast udp sink.
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
And not on unicast udp sinks
https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-thread-pool.c:
Update for g_type_class_add_private() deprecation in recent GLib
2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
Fix indentation
2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
* examples/Makefile.am:
* examples/test-video-disconnect.c:
examples: Add test-video-disconnect example
Simple example which cuts off all clients 10 seconds
after the first one connects.
2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* examples/test-auth-digest.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
rtsp-auth: Add support for parsing .htdigest files
Passwords are usually not stored in clear text, but instead
stored already hashed in a .htdigest file.
Add support for parsing such files, add API to allow setting
a custom realm in RTSPAuth, and update the digest example.
https://bugzilla.gnome.org/show_bug.cgi?id=796637
2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtspclientsink: fix waiting for multiple streams
We were previously only ever waiting for a single stream to notify it's
blocked status through GstRTSPStreamBlocking. Actually count streams to
wait for.
Fixes rtspclientsink sending SDP's without out some of the input
streams.
https://bugzilla.gnome.org/show_bug.cgi?id=796624
2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
docs: add missing auth methods
2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: only create funnel if it didn't exist already.
This precented using multiple protocols for the same stream.
https://bugzilla.gnome.org/show_bug.cgi?id=796634
2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* examples/meson.build:
meson: build auth-digest example
2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
Get payloader stats only for the sending streams
Get/set payloader properties only for streams that actually
contain a payloader element.
https://bugzilla.gnome.org/show_bug.cgi?id=796523
2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/Makefile.am:
Makefile: Don't hardcode libtool for g-i build
Similar to the other commits in core/base/bad
2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
* gst/rtsp-server/rtsp-onvif-media-factory.h:
rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
https://bugzilla.gnome.org/show_bug.cgi?id=796229
2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Don't deadlock in preroll on early close
If the connection is closed very early, the flushing
marker might not get set and rtspclientsink can get
deadlocked waiting for preroll forever.
https://bugzilla.gnome.org/show_bug.cgi?id=786961
2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* meson.build:
* meson_options.txt:
meson: Update option names to omit disable_ and with- prefixes
Also yield common options to the outer project (gst-build in our case)
so that they don't have to be set manually.
2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: use -Wl,-Bsymbolic-functions where supported
Just like the autotools build.
2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
* tests/check/Makefile.am:
configure: check for -good and -bad plugins only in uninstalled setup
Avoids confusing configure messages looking or a -good .pc file
that doesn't exist.
Also use plugindir variables that common macros set while at it.
https://bugzilla.gnome.org/show_bug.cgi?id=795466
2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix session timeout
When streaming data over TCP then is not the keep-alive
functionality working.
The reason is that the function do_send_data have changed
to boolean but the code is still checking the received result
from send_func with GST_RTSP_OK.
The result is that a successful send_func will always lead to
that do_send_data is returning false and the keep-alive will
not be updated.
https://bugzilla.gnome.org/show_bug.cgi?id=795321
2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
Implement support for ULP Forward Error Correction
In this initial commit, interface is only exposed for RECORD,
further work will be needed in rtspsrc to support this for
PLAY.
https://bugzilla.gnome.org/show_bug.cgi?id=794911
2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-onvif-media.c:
Revert "rtsp-server: Switch around sendonly/recvonly attributes"
This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
the opposite, just like the ONVIF standard.
Let's follow those RFCs as we're doing RTSP here, and add a property at
a later time if needed to switch to the SDP RFC behaviour.
https://bugzilla.gnome.org/show_bug.cgi?id=793964
2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From 3fa2c9e to ed78bee
2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/rtspclientsink.c:
gst: Run everything through gst-indent again
2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
rtsp-media: query the position on active streams if media is complete
If the media is complete, i.e. one or more streams have been configured
with sinks, then we want to query the position on those streams only.
A query on an incomplete stream may return a position that originates from
an earlier preroll.
https://bugzilla.gnome.org/show_bug.cgi?id=794964
2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: make sure not to use freed string
Set transport string to NULL after freeing it, so that
at worst we get a NULL pointer if constructing a new
transport string fails (which shouldn't really fail here).
Also check return value of that, just in case.
CID 1433768.
2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: do not free string passed to take_header
2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: do not take lock in request_aux_receiver
Added it right before pushing the previous commit, it is
incorrect and deadlocks because this function gets called
from the join_bin thread, which already holds the lock,
that's the reason why request_aux_sender didn't take the
lock either.
2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-server: add API to enable retransmission requests
"do-retransmission" was previously set when rtx-time != 0,
which made no sense as do-retransmission is used to enable
the sending of retransmission requests, where as rtx-time
is used by the peer to enable storing of buffers in order
to respond to retransmission requests.
rtsp-media now also provides a callback for the
request-aux-receiver signal.
https://bugzilla.gnome.org/show_bug.cgi?id=794822
2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: add rtx ssrc to mikey's crypto sessions
https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
This in order to be able to decrypt the RTCP backchannel
https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Send KeyMgmt header in ANNOUNCE response
When sending back an encrypted RTCP back channel, it is useful
for the client to know the encryption key.
https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: extract handle_keymgmt from rtsp-client
rtspclientsink will also need to parse KeyMgmt headers
sent by the server to decrypt the RTCP backchannel stream
https://bugzilla.gnome.org/show_bug.cgi?id=794813
2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* tests/check/gst/rtspclientsink.c:
rtspclientsink: Fix client ports for the RTCP backchannel
This was broken since the work for delayed transport creation
was merged: the creation of the transports string depends on
calling stream_get_server_port, which only starts returning
something meaningful after a call to stream_allocate_udp_sockets
has been made, this function expects a transport that we parse
from the transport string ...
Significant refactoring is in order, but does not look entirely
trivial, for now we put a band aid on and create a second transport
string after the stream has been completed, to pass it in
the request headers instead of the previous, incomplete one.
https://bugzilla.gnome.org/show_bug.cgi?id=794789
2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client:Error handling when equal http session cookie
There are some clients that are sending same session cookie on random
basis.
https://bugzilla.gnome.org/show_bug.cgi?id=753616
2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory-uri.c:
rtsp-media-factory-uri: Fix compilation with latest GLib
rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
data->factory = g_object_ref (factory);
^
2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
* NEWS:
* RELEASE:
* configure.ac:
* meson.build:
Back to development
=== release 1.14.0 ===
2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.14.0
=== release 1.13.91 ===
2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.13.91
2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/meson.build:
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-context.h:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-onvif-client.h:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
* gst/rtsp-server/rtsp-onvif-media.h:
* gst/rtsp-server/rtsp-onvif-server.h:
* gst/rtsp-server/rtsp-params.h:
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-sdp.h:
* gst/rtsp-server/rtsp-server-prelude.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-server/rtsp-thread-pool.h:
* gst/rtsp-server/rtsp-token.h:
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API
We need different export decorators for the different libs.
For now no actual change though, just rename before the release,
and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-onvif-media-factory.c:
rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
https://bugzilla.gnome.org/show_bug.cgi?id=794143
=== release 1.13.90 ===
2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.13.90
2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-permissions.c:
permissions: add Since tags and example for new API
2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-permissions.h:
* tests/check/gst/permissions.c:
permissions: more bindings-friendly API
https://bugzilla.gnome.org/show_bug.cgi?id=793975
2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* meson.build:
meson: enable more warnings
2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Place netaddress meta on packets received via TCP
This allows us to later map signals from rtpbin/rtpsource back to the
corresponding stream transport, and allows to do keep-alive based on
RTCP packets in case of TCP media transport.
https://bugzilla.gnome.org/show_bug.cgi?id=789646
2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: if OPEN failed, unqueue next command
As READY_TO_PAUSED can no longer return async, the RECORD
command will be queued before the OPEN command fails
(for example in case the server could not be connected),
and record then waits for ever.
https://bugzilla.gnome.org/show_bug.cgi?id=793896
2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: fix retrieval of custom payloader caps
If a bin is passed as the custom payloader, the caps of
its factory will be empty, the correct way to obtain the caps
is to query its sinkpad.
2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: fix extra unref of custom payloader
2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rspclientsink: fix recent code indentation
2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: add missing get_type prototype
2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: allow setting payloader as pad property
This was a FIXME item, and can be quite useful, also
allowing to specify payloader properties from the command
line, which is always nice.
https://bugzilla.gnome.org/show_bug.cgi?id=793776
2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Replace g_print() log line
https://bugzilla.gnome.org/show_bug.cgi?id=793838
2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/rtspclientsink.c:
rtsp-media: fix RECORD getting stuck
The test_record case was working because async=false had
been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
but that was incorrect, as it should not be needed.
Removing async=false made the test fail as expected, this is
fixed by not trying to preroll when preparing the media for
RECORD, as start_prepare is called upon receiving ANNOUNCE,
and our peer will not start sending media until it has received
a response to that request, and sent and received a response
to RECORD as well, thus obviously preventing preroll.
https://bugzilla.gnome.org/show_bug.cgi?id=793738
2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
rtsp-auth: fix set_tls_authentication_mode annotation
2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
* gst/rtsp-server/rtsp-onvif-media.c:
rtp-server: remove redefined variable
res is a boolean variable which is defined in the function scope and
redefined, with no reason, in the loop scope. This patch removes the
redefinition.
https://bugzilla.gnome.org/show_bug.cgi?id=793592
2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: Add functions for checking if stream is receiver or sender
...and replace all checks for RECORD in GstRTSPMedia which are really
for "sender-only". This way the code becomes more generic and introducing
support for onvif-backchannel later on will require no changes in
GstRTSPMedia.
2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-onvif-media-factory.c:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
onvif: Make requires_backchannel() public
...in order to let subclasses building the onvif part of the pipeline
check whether backchannel shall be included or not.
2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-onvif-media.c:
rtsp-server: Switch around sendonly/recvonly attributes
They are wrong in the ONVIF streaming spec. The backchannel should be
recvonly and the normal media should be sendonly: direction is always
from the point of view of the SDP offerer (the server) according to
RFC 3264.
2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-onvif-backchannel.c:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-onvif-client.c:
* gst/rtsp-server/rtsp-onvif-client.h:
* gst/rtsp-server/rtsp-onvif-media-factory.c:
* gst/rtsp-server/rtsp-onvif-media-factory.h:
* gst/rtsp-server/rtsp-onvif-media.c:
* gst/rtsp-server/rtsp-onvif-media.h:
* gst/rtsp-server/rtsp-onvif-server.c:
* gst/rtsp-server/rtsp-onvif-server.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
rtsp: Add support for ONVIF backchannel
This adds a new RTSP server, client, media-factory and media subclass
for handling the specifics of the backchannel. Ideally this later can be
extended with other ONVIF specific features.
2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Add support for sending+receiving medias
We need to add an appsrc/appsink in that case because otherwise the
media bin will be a sink and a source for rtpbin, causing a pipeline
loop.
https://bugzilla.gnome.org/show_bug.cgi?id=788950
2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
* meson.build:
Back to development
=== release 1.13.1 ===
2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* NEWS:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.13.1
2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-session-pool.c:
session-pool: remove nullable return annotation
create_watch can only return NULL from the API guards, no
need for nullable.
2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
set_clock functions: Add nullable annotations
2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-thread-pool.c:
All around: add annotations and API guards
2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* tests/test-cleanup.c:
test-cleanup: bind any port
The meson test suite runs tests in parallel, trying to bind
a single port made the test fail.
2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: make version numbers ints and fix int/string comparison
WARNING: Trying to compare values of different types (str, int).
The result of this is undefined and will become a hard error
in a future Meson release.
2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-context.c:
gst_rtsp_context_get_current: add (skip) annotation
The return value type is defined with G_DEFINE_POINTER_TYPE,
and gi emits the following warning:
Invalid non-constant return of bare structure or union; register as
boxed type or (skip)
2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: add type annotations
gi doesn't seem to be able to figure out the type of the
signal parameters when defined with G_DEFINE_POINTER_TYPE
2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
autotools: use -fno-strict-aliasing where supported
https://bugzilla.gnome.org/show_bug.cgi?id=769183
2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: use -fno-strict-aliasing where supported
https://bugzilla.gnome.org/show_bug.cgi?id=769183
2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-mount-points.c:
mount-points: bail out of loop again when matching mount points
Previous patch led to us iterating the entire sequence. Bail out
of the loop again if we have a match but are moving away from it.
https://bugzilla.gnome.org/show_bug.cgi?id=771555
2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/mountpoints.c:
tests: mountpoints: add more checks for mount point path matching
https://bugzilla.gnome.org/show_bug.cgi?id=771555
2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
* gst/rtsp-server/rtsp-mount-points.c:
mount-points: fix matching of paths where there's also an entry with a common prefix
e.g. with the following mount points
/raw
/raw/snapshot
/raw/video
_match() would not match /raw/video and /raw/snapshot correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=771555
2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-permissions.h:
* tests/check/gst/permissions.c:
permissions: add some new API to make this usable from bindings
https://bugzilla.gnome.org/show_bug.cgi?id=787073
2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-token.c:
rtsp-token: annotate constructors for bindings
This maps _new_empty() to _new(), which also makes RTSPToken()
work properly now. Since this API wasn't usable from bindings
before, this should hopefully be fine.
https://bugzilla.gnome.org/show_bug.cgi?id=787073
2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-token.c:
* gst/rtsp-server/rtsp-token.h:
* tests/check/gst/token.c:
rtsp-token: add some API to set fields from bindings
The existing functions are all vararg-based and as such
not usable from bindings.
https://bugzilla.gnome.org/show_bug.cgi?id=787073
2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspclientsink.c:
* tests/check/gst/rtspserver.c:
* tests/check/gst/sessionpool.c:
* tests/check/gst/stream.c:
tests: fix indentation
Fix and "fix".
2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: rtspserver: fix another ref leak
Even if this didn't show up in valgrind.
2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspclientsink.c:
tests: rtspclientsink: fix leak
2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
* tests/check/gst/rtspserver.c:
test: rtspserver: plug memory leak in test_no_session_timeout
In test_no_session_timeout, unref the rtsp session object when the
test is done.
https://bugzilla.gnome.org/show_bug.cgi?id=792127
2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtpsclientsink: Initialize and clear newly added mutex and cond
While it *did* work, glib would automatically create new mutex and cond
... which never got freed
2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set multicast TTL on the multicast sockets
And not if we do unicast UDP.
https://bugzilla.gnome.org/show_bug.cgi?id=791743
2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
In the multicast case (as in test-multicast, not test-multicast2), the
address could be allocated/reserved (and thus set) already without
allocating the actual socket. We need to allocate the socket here still
instead of just claiming that it was already allocated.
See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtspclientsink: Use the new rtsp-stream API
https://bugzilla.gnome.org/show_bug.cgi?id=790412
2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtspclientsink: Wait until OPEN has been scheduled
Make sure that the sink thread has started opening connection
to the server before continuing.
https://bugzilla.gnome.org/show_bug.cgi?id=790412
2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
* common:
Automatic update of common submodule
From e8c7a71 to 3fa2c9e
2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: Minor doc fixes
Mostly for g-i
2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
* tests/Makefile.am:
tests: disable all tests when --disable-tests is used
Move conditional subdir include into top level.
Based on patch by: Joel Holdsworth
https://bugzilla.gnome.org/show_bug.cgi?id=757703
2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
* meson_options.txt:
* tests/meson.build:
meson: build more tests and add options to disable tests and examples
2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
* gst/rtsp-server/rtsp-session.c:
Fix build when -Werror=deprecated-declarations is on
As gst_rtsp_session_next_timeout is deprecated.
```
../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:760:3: error: ‘gst_rtsp_session_next_timeout’ is deprecated: Use 'gst_rtsp_session_next_timeout_usec' instead [-Werror=deprecated-declarations]
res = (gst_rtsp_session_next_timeout (session, now) == 0);
^~~
../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
```
2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
* common:
Automatic update of common submodule
From 3f4aa96 to e8c7a71
2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* tests/check/gst/media.c:
check/media: Add seekability test case: not all streams are active
Media contains two streams but only one is complete and prepared
for playing.
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Do not reset 'blocking' if stream is already blocked
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: remove vs_module_defs_dir variable which is no longer needed
2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-session.h:
rtsp: fix distcheck
2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
* gst/rtsp-server/meson.build:
* win32/MANIFEST:
* win32/common/libgstrtspserver.def:
win32: remove .def file with exports
They're no longer needed, symbol exporting is now explicit
via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
autotools: stop controlling symbol visibility with -export-symbols-regex
Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
This should result in consistent behaviour for the autotools and
Meson builds.
2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp-server: add missing GST_EXPORT and export deprecated funcs
2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
* tests/check/gst/media.c:
check: Add seekability testing on medias
Make sure that once GstRTSPMedia are prepared they returned
the expected seekability results
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* win32/common/libgstrtspserver.def:
rtsp-media: Enable seeking query before pipeline is complete
SDP are now provided *before* the pipeline is fully complete. In order
to know whether a media is seekable or not therefore requires asking
the invididual streams.
API: gst_rtsp_stream_seekable
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix handling in default_unsuspend()
Handle the case when streams are not blocked and media
is suspended from PAUSED.
Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
* tests/check/gst/media.c:
check/media: Fix thread pool leak.
Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Removed fakesink elements
There is not need of adding fakesink elements to the media
pipeline in the dynamic-payloader case.
The media pipeline itself is dynamically updated with
the receiver and sender parts that are based on the client
transport information known after SETUP has been received.
Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Corrected ASYNC_DONE handling
Media is complete when all the transport based parts are
added to the media pipeline. At this point ASYNC_DONE is
posted by the media pipeline and media is ready to enter
the PREPARED state.
Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
https://bugzilla.gnome.org/show_bug.cgi?id=790674
2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
* tests/check/gst/media.c:
check/media: Check that prepared media can provide a SDP
Whenever a RTSPMedia is prepared, it should be able to provide a SDP
2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Don't leak addr
CID #1422260
2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream.c:
Run gst-indent
2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't unblock with remaining dynamic payloaders
If we still have some dynamic paylaoders which haven't posted
no-more-pads yet, don't go to PREPARED if one of the streams
blocked.
The risk was that we would end up not exposing/using all specified
streams.
The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
then it will take a bit more time to start. But only if those 3
conditions are present.
https://bugzilla.gnome.org/show_bug.cgi?id=769521
2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix doc
2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't set float on a gint64 variable
Just use 0. Fixes 'undefined' behaviour from clang
2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix previous commit
We only want to count dynamic payloaders
2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
rtsp-media: Handle multiple dynamic elements
If we have more than one dynamic payloader in the pipeline, we need
to wait until the *last* one emits 'no-more-pads' before switching
to PREPARED.
Failure to do so would result in a race where some of the streams
wouldn't properly be prepared
https://bugzilla.gnome.org/show_bug.cgi?id=769521
2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
* win32/common/libgstrtspserver.def:
win32: Fix exported symbols list
2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Only update the RTP udpsink if it actually exists
For send-only streams it does not exist, but the RTCP udpsink might.
2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
* win32/common/libgstrtspserver.def:
win32: Update exports
2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-media: seek on media pipelines that are complete
Make sure that a seek is performed on pipelines that
contain at least one sink element.
Change-Id: Icf398e10add3191d104b1289de612412da326819
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/client.c:
* tests/check/gst/media.c:
* tests/check/gst/rtspserver.c:
* tests/check/gst/stream.c:
Dynamically reconfigure pipeline in PLAY based on transports
The initial pipeline does not contain specific transport
elements. The receiver and the sender parts are added
after PLAY.
If the media is shared, the streams are dynamically
reconfigured after each PLAY.
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: obtain stream position from pad
If no sinks have been added yet, obtain the current and
the stop position of the stream from the send_src pad.
Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
rtsp-session-media: add function to get a list of transports
Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: add functions to get rtp and rtcp multicast sockets
Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
stream: set async=sync=false only for RTCP appsink
Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: return minimum value in query position case
The minimum position should be returned as we are interested
in the whole interval.
Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
* gst/rtsp-server/rtsp-session.c:
* tests/check/gst/rtspserver.c:
rtsp-session: Handle the case when timeout=0
According to the documentation, a timeout of value 0 means
that the session never timeouts. This adds handling of that.
If timeout=0 we just return with a -1 from
gst_rtsp_session_next_timeout_usec ().
https://bugzilla.gnome.org/show_bug.cgi?id=785058
2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
https://bugzilla.gnome.org/show_bug.cgi?id=785024
2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media-factory.c:
docs: add media factory transport mode accessors
and fix the documentation for the return value of the getter
2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: unref 'pipelined_requests' in finalize
The hash table priv->pipelined_requests is not unref:ed in the
finalize funktion. Make sure it is.
https://bugzilla.gnome.org/show_bug.cgi?id=788704
2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Initialize scalar variable
CID 1418985
2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
* win32/common/libgstrtspserver.def:
win32: Update export file
2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Start support for RTSP 2.0
This adds basic support for new 2.0 features, though the protocol is
subposdely backward incompatible, most semantics are the sames.
This commit adds:
- features:
* version negotiation
* pipelined requests support
* Media-Properties support
* Accept-Ranges support
- APIs:
* gst_rtsp_media_seekable
The RTSP methods that have been removed when using 2.0 now return
BAD_REQUEST.
https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Use stream duration as stream-stop if segment was not configured with a stop
Allowing client to know stream duration when no seeking happened.
https://bugzilla.gnome.org/show_bug.cgi?id=783435
2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: Don't cache any media if NULL was returned as key
The docs already mentioned this, but we actually stored it in the hash
table with key==NULL and leaked its reference forever.
2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
rtspclientsink: Use a mutex for protecting against concurrent send/receives
This is a simple port of:
* a722f6e8329032c6eda4865d6a07f4ba5981d7ea
* c438545dc9e2f14f657bc0ef261fff726449867b
* cd17c71dcea5c9310d21f1347c7520983e5869ac
in gst-plugins-good.
2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
* gst/rtsp-server/rtsp-sdp.c:
sdp: fix Memory leak in error case
https://bugzilla.gnome.org/show_bug.cgi?id=787059
2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
* pkgconfig/meson.build:
meson: don't install -uninstalled.pc file
https://bugzilla.gnome.org/show_bug.cgi?id=786457
2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From 48a5d85 to 3f4aa96
2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix typo in debug message
2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: hide symbols by default unless explicitly exported
2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
Fixes meson warning about undefined @srcdir@.
2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/meson.build:
meson: skip tests on windows for now
As we do in the other modules. As libgstcheck is currently not
built on windows. Fixes "Fallback variable 'gst_check_dep' in
the subproject 'gstreamer' does not exist"" Meson error.
2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix connection delay due to wrong assumption on last-sample
Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
multiudpsink's last-sample always comes from the payloader. Which
is wrong if auxiliary streams are multiplexed in the same stream.
So check the buffer's ssrc against the caps'ssrc before to use its
seqnum. If not the same ssrc just use the payloader as done prior
the commit above or when there is no last-sample yet.
https://bugzilla.gnome.org/show_bug.cgi?id=784094
2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
* meson.build:
meson: Allow using glib as a subproject
2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: fix with-package-name option
https://bugzilla.gnome.org/show_bug.cgi?id=784082
2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* Makefile.am:
Distribute meson_options.txt
2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* Makefile.am:
And config.h.meson is no longer dist either
2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
* config.h.meson:
* meson.build:
meson: config.h.meson is no longer needed
2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
* tests/check/meson.build:
* tests/meson.build:
meson: Fix building tests and activate them again
2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
* tests/check/meson.build:
meson: Do not use path separator in test names
Avoiding warnings like:
WARNING: Target "elements/audioamplify" has a path separator in its name.
2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
* meson_options.txt:
meson: add options to set package name and origin
https://bugzilla.gnome.org/show_bug.cgi?id=782172
2017-05-18 10:35:18 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-context.h:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-params.h:
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-sdp.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-server/rtsp-thread-pool.h:
* gst/rtsp-server/rtsp-token.h:
Mark symbols explicitly for export with GST_EXPORT
2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* configure.ac:
* gst/rtsp-sink/Makefile.am:
Remove plugin specific static build option
Static and dynamic plugins now have the same interface. The standard
--enable-static/--enable-shared toggle are sufficient.
2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
* meson.build:
Back to development
=== release 1.12.0 ===
2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.12.0
=== release 1.11.91 ===
2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.11.91
2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From 60aeef6 to 48a5d85
2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream.c:
gi: Fix some annotations and docstrings
2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
* gst/rtsp-server/meson.build:
* meson.build:
* meson_options.txt:
meson: Build gir
2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
* autogen.sh:
* common:
Automatic update of common submodule
From 39ac2f5 to 60aeef6
=== release 1.11.90 ===
2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.11.90
2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
* examples/test-launch.c:
examples: make test-launch pipeline shared by default as well
2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
gstreamer-rtsp-server: Add both srcdir and builddir to the include path
Just the build dir is not going to work for srcdir!=builddir.
2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
* meson.build:
meson: Update version
2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.11.2 ===
2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.11.2
2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
meson: dist meson build files
Ship meson build files in tarballs, so people who use tarballs
in their builds can start playing with meson already.
2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
* examples/test-record.c:
examples/test-record: Add extra line to initial printout
Add an example line of how to deliver a stream to the
RTSP RECORD example
2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
If there is no Content-Length header, no body would be allocated and the
'\0' would also not be appended to the body.
2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
While they logically have 0 bytes length, GstRTSPConnection is appending
a '\0' to everything making the size be 1 instead.
2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: bump version
2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
affected.
2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.11.1 ===
2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* win32/common/libgstrtspserver.def:
Release 1.11.1
2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: corrected if-statement in _get_server_port()
This bug was accidentally introduced while fixing a segfault
in _get_server_port() function.
https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/stream.c:
rtsp-stream: fixed segmenation fault in _get_server_port()
Calling function gst_rtsp_stream_get_server_port() results in
segmenation fault in the RTP/RTSP/TCP case.
Port that the server will use to receive RTCP makes only
sense in the UDP case, however the function should handle
the TCP case in a nicer way.
https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
* gst/rtsp-server/rtsp-media-factory.c:
dosc: Fix a little typo
https://bugzilla.gnome.org/show_bug.cgi?id=777037
2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
* pkgconfig/meson.build:
meson: generate pkg-config -uninstalled pc files
Generating those files is useful for users building the GStreamer stack
using meson and having to link it to another project which is still
using the autotools.
https://bugzilla.gnome.org/show_bug.cgi?id=776810
2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
pkgconfig: fix -uninstalled pc file
pcfiledir was never defined so the paths were wrong.
https://bugzilla.gnome.org/show_bug.cgi?id=776867
2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/rtspserver.c:
rtsp-stream: Fixed TCP transport case
Make sure that the appsink element is actually added to
the bin before trying to link it with the elements in it.
https://bugzilla.gnome.org/show_bug.cgi?id=776343
2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
* Makefile.am:
* configure.ac:
* gst-rtsp.spec.in:
Remove generated .spec file
Likely extremely bitrotten, and we should not ship this anyway.
2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From f980fd9 to 39ac2f5
2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
media: Fix pt map caps
Since decryption is handled within rtpbin, all outcoming stream
caps will be application/x-rtp (i.e. regular rtp)
Fixes RECORD with SRTP streams
2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: Create media objects with the proper transport mode
The function called immediately afterwards (collect_streams()) will
need it to work properly
2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: Don't create a pipeline for the media pipeline string
We're going to put a pipeline into a pipeline otherwise, which is not
exactly ideal.
2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
* gst/rtsp-server/rtsp-media.c:
media: Fix race condition around finish_unprepare() if called multiple time
https://bugzilla.gnome.org/show_bug.cgi?id=755329
2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Don't leave stale pointer after unref
Fix a warning on shutdown - don't keep a pointer to an
alread-unreffed object.
2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitmodules:
common: use https protocol for common submodule
https://bugzilla.gnome.org/show_bug.cgi?id=775110
2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
stream: block the output of rtpbin instead of the source pipeline
85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
detection of the srtp rollover counter to add to the SDP.
Unfortunately, it was incomplete for live pipelines where the logic
blocks the source bin before creating the SDP and thus would never have
the necessary informaiton to create a correct SDP with srtp encryption.
Move the pad blocks to rtpbin's output pads instead so that the
necessary information can be created before we need the information for
the SDP.
https://bugzilla.gnome.org/show_bug.cgi?id=770239
2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: add IDLE timeout, before session exists
The RTSP server will not timeout an idle RTSP connection
(note this is different from doing timeout on a RTSP
session).
At least for Apache this is a problem when running RTSP over
HTTPS since it uses one of the threads (there is a rather
limited number) that are available for handling requests.
https://bugzilla.gnome.org/show_bug.cgi?id=771830
2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
.gitignore more
2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set close-socket FALSE on UDP src:es
With this RTSP server can use the sockets independent on the udpsrc
state.
When the udp src is finalized it will unref socket and when g_socket
is finalized the socket will be closed.
https://bugzilla.gnome.org/show_bug.cgi?id=765673
2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Move to new helper function to parse authentication responses
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/Makefile.am:
* examples/test-auth-digest.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* win32/common/libgstrtspserver.def:
rtsp-auth: Add support for Digest authentication
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
* Makefile.am:
* gst/rtsp-server/meson.build:
* meson.build:
* tests/check/meson.build:
* win32/MANIFEST:
* win32/common/libgstrtspserver.def:
Enable building with MSVC
https://bugzilla.gnome.org/show_bug.cgi?id=774640
2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
* meson.build:
meson: gstreamer gst_check_dep does not exist on windows
2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
* gst/rtsp-server/rtsp-client.c:
client: update do_send_message to match type GstRTSPClientSendFunc
This type mismatch fails building with MSVC
https://bugzilla.gnome.org/show_bug.cgi?id=774640
2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Fix indentation
2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only signal "new-state" if the state has actually changed
https://bugzilla.gnome.org/show_bug.cgi?id=774173
2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: emit signal in the beginning of each rtsp request
These signals let the application validate the requests, configure the
media/stream in a certain way and also generate error status code in
case of error or bad request.
https://bugzilla.gnome.org/show_bug.cgi?id=758062
2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: update version
=== release 1.11.0 ===
2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.10.0 ===
2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.10.0
2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
* tests/check/gst/stream.c:
tests: try to avoid using the same ports in different tests
Causes problems with client multicast tests otherwise if
tests are run in parallel.
https://bugzilla.gnome.org/show_bug.cgi?id=773640
2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/client.c:
tests: client: use fail_unless_equals_foo() for better failure reporting
2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Session filter in unwatch session
Call session filter with filter_session_media as paramer in
client_unwatch_session if using drop_backlog = FALSE.
In client_unwatch_session its allowed to grow the watchs backlog.
If using drop_backlog = FALSE and the backlog is full it will cause
a deadlock when setting session media state to NULL
if the backlog is not allowed to grow.
https://bugzilla.gnome.org/show_bug.cgi?id=771983
2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: add fallbacks for gst modules
For gst-all.
2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix factory leaking in find_media() in error cases
https://bugzilla.gnome.org/show_bug.cgi?id=771488
2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Fix randomly missing streams from SDP with dynamic elements
When using dynamic elements, gst_rtsp_stream_join_bin() is called from
"pad-added" signal. In that case priv->srcpad could already have its caps,
and they'll be sent to priv->send_src[0] pad. That means that when it
connects "notify::caps" signal, that pad could already have received its
caps and the signal won't be emitted anymore.
In that case priv->caps stay to NULL and when building the SDP that stream
gets ignored. Leading to missing video or audio when playing in client side.
https://bugzilla.gnome.org/show_bug.cgi?id=772478
2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: update version
=== release 1.9.90 ===
2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.9.90
2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: Hint that set_multicast_iface expects the name of the interface
To prevent any possibly confusion with IPs or anything else.
https://bugzilla.gnome.org/show_bug.cgi?id=771530
2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
configure: Depend on gstreamer 1.9.2.1
2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
* autogen.sh:
* common:
Automatic update of common submodule
From b18d820 to f980fd9
2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
* autogen.sh:
* common:
Automatic update of common submodule
From 6f2d209 to b18d820
2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Remove unused _locked() variant of a function
It was added during refactoring.
2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: cosmetic cleanup
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Compare IP addresses case insensitive in more places
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* common:
* gst/rtsp-server/rtsp-stream.c:
stream: Fix leaked joined_bin
There is no need to keep a strong ref on it, and _leave_bin() was
setting it to NULL before calling g_clear_object() so it was leaked.
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Compare IP address strings case insensitive
Otherwise IPv6 addresses might fail this comparision.
2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Bind multicast sockets to ANY as before
https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: Fix segfault when doing keep-alive after removing the session
If keep-alive happens after removing the session but before finalizing the
stream transport, we would segfault.
https://bugzilla.gnome.org/show_bug.cgi?id=750544
2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Always create multicast UDP elements if the protocol flag is set
Adding them later will cause deadlocks due to
1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
2) adding the multicast sink
3) waiting for it to get data to preroll again
3) never happens because the queues after the tee are full.
2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix up various multicast related issues
2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
* tests/check/gst/stream.c:
tests: Fix compilation
2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/stream.c:
stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
This is basically reverting changes introduced in commit f62a9a7,
because it was introducing various regressions:
- It introduces a leak of udpsrc elements that got wrongly fixed by adding
an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
- If a mcast client connects, it creates a new socket in SETUP to try to respect
the destination/port given by the client in the transport, and overrides the
socket already set on the udpsink element. That means that if we already had a
client connected, the source address on the udp packets it receives suddenly
changes.
- If a 2nd mcast client connects, the destination/port in its transport is
ignored but its transport wasn't updated.
What this patch does:
- Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
- Always have a tee+queue when udp is enabled. This could be optimized
again in a later patch, but is more complicated. If no unicast clients
connects then those elements are useless, this could be also optimized
in a later patch.
- When mcast transport is added, it creates a new set of udpsrc/udpsink,
seperated from those for unicast clients. Since we already support only
one mcast address, we also create only one set of elements.
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: factor our plug_src function
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: factor out plug_sink function
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: small documentation clarification
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Keep a ref on joined bin
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: code cleanup
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: small fix in error code path
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
but keeps unit tests.
https://bugzilla.gnome.org/show_bug.cgi?id=766612
2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.9.2 ===
2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.9.2
2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
* config.h.meson:
* examples/meson.build:
* gst/meson.build:
* gst/rtsp-server/meson.build:
* gst/rtsp-sink/meson.build:
* meson.build:
* pkgconfig/meson.build:
* tests/check/meson.build:
* tests/meson.build:
Add support for Meson as alternative/parallel build system
https://github.com/mesonbuild/meson
2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
* configure.ac:
* tests/check/Makefile.am:
build: silence error about pthread for 'make check' in osx
Fixes "clang: error: argument unused during compilation: '-pthread'"
2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix leaking of media in error cases
With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
and myself to make the media refcounting a bit easier to follow.
https://bugzilla.gnome.org/show_bug.cgi?id=755632
2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix leaking of session in error cases
https://bugzilla.gnome.org/show_bug.cgi?id=755632
2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From f363b32 to f49c55e
2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.9.1 ===
2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.9.1
2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* configure.ac:
configure: Need to add -DGST_STATIC_COMPILATION when building only statically
https://bugzilla.gnome.org/show_bug.cgi?id=767463
2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* common:
Automatic update of common submodule
From ac2f647 to f363b32
2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
sdp: add rollover counters for all sender SSRC
We add different crypto sessions in MIKEY, one for each sender
SSRC. Currently, all of them will have the same security policy, 0.
The rollover counters are obtained from the srtpenc element using the
"stats" property.
https://bugzilla.gnome.org/show_bug.cgi?id=730539
2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-server.h:
docs: fix some typos
2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/Makefile.am:
g-i: pass compiler env to g-ir-scanner
It's what introspection.mak does as well. Should
fix spurious build failures on gnome-continuous
(caused by g-ir-scanner getting compiler details
via python which is broken in some environments
so passing the compiler details bypasses that).
2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
This works with rtspsrc and live555, but fails with e.g. ffmpeg.
https://bugzilla.gnome.org/show_bug.cgi?id=766619
2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Check return value of sscanf
And just make sure we always have 0/0 if we have an error
CID #1352031
2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/rtspserver.c:
* tests/check/gst/stream.c:
rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
- Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
- Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
- Create unit test for shared media.
https://bugzilla.gnome.org/show_bug.cgi?id=764744
2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
For IPv6 addresses, binding to a multicast group does not work on Linux
either. Always bind to ANY and then later join the multicast group.
https://bugzilla.gnome.org/show_bug.cgi?id=764679
2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
* common:
Automatic update of common submodule
From 6f2d209 to ac2f647
2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-thread-pool.c:
rtsp-thread-pool: explained why GSource is a part of ThreadImpl
Clarified why it is necessary to add source information to
GstRTSPThreadImpl. See the reported bug in GLib:
https://bugzilla.gnome.org/show_bug.cgi?id=720186
for more information.
https://bugzilla.gnome.org/show_bug.cgi?id=761702
2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
* examples/Makefile.am:
examples: Clean up CFLAGS/LDADD even more
The internal .la should come first and is part of LDADD, as is
GST_CFLAGS/LIBS.
2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
* examples/Makefile.am:
examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/Makefile.am:
rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-server: Implement clock signalling according to RFC7273
For NTP and PTP clocks we signal the actual clock that is used and signal
the direct media clock offset.
For all other clocks we at least signal that it's the local sender clock.
This allows receivers to know which clock was used to generate the media and
its RTP timestamps. Receivers can then implement network synchronization,
either absolute or at least relative by getting the sender clock rate directly
via NTP/PTP instead of estimating it from RTP timestamps and packet receive
times.
https://bugzilla.gnome.org/show_bug.cgi?id=760005
2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Add support for setting the multicast interface
https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-media: Add support for setting the multicast interface
https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: use new gst_element_class_add_static_pad_template()
https://bugzilla.gnome.org/show_bug.cgi?id=763196
2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.8.0 ===
2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.8.0
2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
This would get us NO_PREROLL in the bin again and break seeking.
Thanks to Carlos Rafael Giani for helping to debug this!
https://bugzilla.gnome.org/show_bug.cgi?id=740509
=== release 1.7.91 ===
2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.7.91
2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
Without this, RECORD pipelines are broken because
a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
added later. Previously it was there earlier and due to NO_PREROLL caused the
pipeline to preroll immediately
b) the udpsrc for the pipeline is added later and never set to PLAYING state,
as the corresponding code previously was only for PLAY pipelines.
https://bugzilla.gnome.org/show_bug.cgi?id=763281
2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix typo in the docstring
gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Disable multicast loopback for all our sockets
On Windows this is a receiver-side setting, on Linux a sender-side setting. As
we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
loopback setting on the socket... while udpsink does which unfortunately has
no effect here on Windows but on Linux.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
* tests/check/gst/stream.c:
stream tests: added new tests
Test a case when the address pool only contains multicast addresses
and the client is requesting unicast udp.
Added tests for multicast ports allocation.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Only bind multicast sockets to ANY on Windows
On Linux it is still needed to bind to the multicast address
to filter out random other packets, while on Windows binding
to multicast addresses just fails.
2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
Otherwise we fail to allocate UDP ports if the pool only contains multicast
addresses, which is something that used to work before. For unicast addresses
if the pool contains none, we just allocate them as if there is no pool at
all.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: Fix indentation
2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't bind the sockets to multicast addresses
This works on Linux but fails completely on Windows. You're supposed
to bind to ANY and then join the multicast group.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
=== release 1.7.90 ===
2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.7.90
2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From b64f03f to 6f2d209
2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
* tests/check/gst/rtspclientsink.c:
rtspsink: Fix some leaks in rtspclientsink and the unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=762525
2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
* tests/check/gst/media.c:
* tests/check/gst/rtspclientsink.c:
* tests/check/gst/rtspserver.c:
* tests/check/gst/stream.c:
tests: unit test fixes
Removed port allocation test from the media suite.
The port allocation failure is now in the stream suite.
rtspserver:
Make sure that the media is suspended after the DESCRIBE request
before reconfiguring the UDP sinks.
rtspclientsink:
In the RECORD case we have to set async property to false
for the appsink element in the test in order to make sure
that the media pipeline doesn't hang in start_preroll().
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: postpone UDP socket allocation until SETUP
Postpone the allocation of the UDP sockets until we know
what transport has been chosen by the client.
Both unicast and multicast UDP sources are created in one
function.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: postpone the creation of the UDP sources
Code refactoring: allocate the UDP ports after the sender and
the reciver parts have been created.
We postpone the creation of the UDP sources until the UDP
ports have been allocated.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: added function for setting UDP sources to PLAYING state
Code refactoring: Introduced a function for setting UDP sources
to PLAYING state.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: added function for creating and configuring UDP sources
Code refactoring: create and configure UDP sources in a separate function.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: added function for RTP/RTCP socket configuration
Code refactoring: configure RTP and RTCP sockets for UDP sinks
in a separate function.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: added function for creating and configuring UDP sinks
Code refactoring: create and configure UDP sinks in a separate function.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: added helper function for creating the sender/receiver parts
Code refactoring: introduced helper function for creating
the receiver and the sender parts of the streaming pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.7.2 ===
2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.7.2
2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
uninstalled.pc: add support for non libtool build systems
Currently the .la path is provided which requires to use libtool as
mentioned in the GStreamer manual section-helloworld-compilerun.html.
It is fine as long as the application is built using libtool.
So currently it is not possible to compile a GStreamer application
within gst-uninstalled with CMake or other build system different
than autotools.
This patch allows to do the following in gst-uninstalled env:
gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
gstreamer-rtsp-server-1.0)
Previously it required to prepend libtool --mode=link
https://bugzilla.gnome.org/show_bug.cgi?id=720778
2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: remove check for impossible condition
Goto error label checks stream to see if it needs to be unreferenced before
returning, but this goto jumps happens before the stream is ever set, so it
will always be NULL in this error label.
CID #1352034
2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: clean switch statements
Coverity demands for fallthrough statements to be clearly commented,
to distinguish from accidental fall throughs. And it also needs all
cases to finish with a break, even if the break is never going to be
executed like in the case of a continue jump.
CID #1352039
CID #1352040
2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* tests/check/Makefile.am:
tests: extend the AM_TESTS_ENVIRONMENT from check.mak
To get the CK_DEFAULT_TIMEOUT defined for all tests
Also removes a 120 seconds timeout that was set as default
explicitly in this module
https://bugzilla.gnome.org/show_bug.cgi?id=761472
2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* autogen.sh:
* common:
Automatic update of common submodule
From 86e4663 to b64f03f
2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix state_lock not locked again when preroll fails
https://bugzilla.gnome.org/show_bug.cgi?id=761399
2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
configure: Move plugin specific flags below all the others
They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
-no-undefined. And -no-undefined is required on Windows to build DLLs.
2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Simplify slightly using new -base API
Use the new Mikey and SDP API in the base plugins libs
to simplify some code.
https://bugzilla.gnome.org/show_bug.cgi?id=758180
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* .gitignore:
* configure.ac:
* gst/Makefile.am:
* gst/rtsp-sink/Makefile.am:
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
* gst/rtsp-sink/plugin.c:
* tests/check/Makefile.am:
* tests/check/gst/rtspclientsink.c:
rtspsink: Add rtspclientsink element
Add an rtspclientsink element that accepts streams for which
there is a registered payloader and sends them to
an RTSP server using RECORD.
Sending is synchronised to the pipeline clock. Payload-types
are automatically selected. The 'new-payloader' signal is fired
for custom configuration of payloaders when they are created.
Can now stream a movie like this:
receiver:
./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
decodebin name=depay1 ! audioconvert ! autoaudiosink )"
sender:
gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
https://bugzilla.gnome.org/show_bug.cgi?id=758180
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: Add functions for using rtsp-stream from the client
Add a boolean to indicate that the rtsp-stream is running on the
'client' side of an RTSP connection, for sending streams via
RECORD. In that case, the roles of the client/server ports
in transport setup are swapped.
https://bugzilla.gnome.org/show_bug.cgi?id=758180
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
rtsp-sdp: Add gst_rtsp_sdp_from_stream()
A new function that adds info from a GstRTSPStream into an SDP message.
https://bugzilla.gnome.org/show_bug.cgi?id=758180
2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix mutex beeing unlocked while they should be locked
https://bugzilla.gnome.org/show_bug.cgi?id=761226
2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: add missing break in "clock" property setter
CID 1348453
2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fixed assert during update transport
When RTSP server trying update transport during multicast, it throws an
assert. The assert is thrown because it is trying to get the parent of
an non-existing funnel element.
https://bugzilla.gnome.org/show_bug.cgi?id=760150
2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-thread-pool.h:
* gst/rtsp-server/rtsp-token.h:
docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
gtk-doc can handle static inline functions just fine these days,
there's no need for this stuff any more.
2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
sdp: replace duplicated codes to call new base sdp apis
https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock.c:
test-netclock: Use the new API to configure a clock directly
2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Add API to directly configure a clock on the media pipelines
2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: Add FIXME for 2.0
2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix indentation
2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Do not prepare media after media times out
Deferred calls to start_prepare() can be deferred past the point until
which wait_preroll() and by proxy gst_rtsp_media_get_status() is
prepared to wait. Previously there was no lock and no check for this
situation. This meant that a media could be prepared and unprepared
simultaneously by two different threads. Now a lock is in place and a
suitable check is done.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
Without TEARDOWN it might be desireable to keep the media running and continue
sending data to the client, even if the RTSP connection itself is
disconnected.
Only do this for session medias that have only UDP transports. If there's at
least on TCP transport, it will stop working and cause problems when the
connection is disconnected.
https://bugzilla.gnome.org/show_bug.cgi?id=758999
2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.7.1 ===
2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.7.1
2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
* configure.ac:
configure: Make -Bsymbolic check work with clang.
Update the -Bsymbolic check with the version glib has. This version
works with clang.
https://bugzilla.gnome.org/show_bug.cgi?id=759713
2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-session-pool.c:
rtsp-session-pool: Avoid dollar sign ($) in session ids
Live555 in VLC strips off dollar signs and then gets very confused,
we don't loose too much entropy by just skipping it.
2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-server/rtsp-thread-pool.h:
* gst/rtsp-server/rtsp-token.h:
rtsp-server: Add g_autoptr() support to all types
https://bugzilla.gnome.org/show_bug.cgi?id=754464
2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fixed valgrind error
Fixed the valgrind error in unit test. The UDP source created during
gst_rtsp_stream_join_bin() was not released while destroying the rtp
bin.
https://bugzilla.gnome.org/show_bug.cgi?id=759010
2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* autogen.sh:
* common:
Automatic update of common submodule
From b319909 to 86e4663
2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: suspend media during setup request
SETUP request from clients needs to suspend the media to clear the
prerolled buffers. Otherwise it will not affect the prerolled buffer
and the prerolled buffers will be incorrect (for example block-size
from setup request will not affect the prerolled buffer unless the
media is suspended).
https://bugzilla.gnome.org/show_bug.cgi?id=758268
2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: create stream pipeline based on transport
Based on the protocol, create the rtsp stream pipeline. If only TCP or
only UDP is set as the transport protocol, it will not add the extra tee
or queue element to the pipeline. Both these elements will be added, if
it supports both TCP and UDP protocols. This improves the pipeline
performance when one protocol is present.
https://bugzilla.gnome.org/show_bug.cgi?id=758179
2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
Adding them when not needed will start some logic inside rtpbin that might be
problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
would start up a rtpjitterbuffer and behave in weird ways.
We still set up the UDP sources for RTP receiving for a sender media to be
able to receive any packets sent by the client for NAT traversal. They will
all go to a fakesink though.
Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
NO_PREROLL, which will cause deadlocks when seeking the media as it will never
receive ASYNC_DONE after a seek.
https://bugzilla.gnome.org/show_bug.cgi?id=758319
2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Disable multicast loopback for the multicast udp sources too
On POSIX this setting is for sender sockets, on Windows for receiver sockets.
Previously we were only setting this for sender sockets, which caused looped
back packets to be received on Windows if a multicast transport was used.
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* examples/test-record-auth.c:
* examples/test-record.c:
examples: Actually use the provided port in the record examples
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* examples/test-record-auth.c:
test-record-auth: Add the option to build in TLS support
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* examples/test-auth.c:
test-auth: Use an 'anonymous' user for unauthenticated default
There's a comment on one of the resources that 'user' and 'admin'
shouldn't even be able to see it, but they can if the default
token is 'admin2', since that gives them access anyway.
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-record-auth.c:
Add test-record-auth example
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
* gst/rtsp-server/rtsp-server.c:
rtsp-server: Change the logic so we don't pop a NULL context
When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
will sometimes fail. This call is made before any context is pushed
resulting in an attempt to pop a NULL context.
https://bugzilla.gnome.org/show_bug.cgi?id=757949
2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
* tests/check/gst/rtspserver.c:
rtspserver: Add udp-mcast transport SETUP test
Refactor utility functions in the test file so they can handle
more than UDP and TCP as lower transport.
https://bugzilla.gnome.org/show_bug.cgi?id=756969
2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Always unref return value of gst_object_get_parent()
Fixes a leak of a GstBin in the udp-mcast case.
https://bugzilla.gnome.org/show_bug.cgi?id=756968
2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From b99800a to b319909
2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Use new GST_ENABLE_EXTRA_CHECKS #define
https://bugzilla.gnome.org/show_bug.cgi?id=756870
2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From 6babecd to b99800a
2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Update GLib dependency to 2.40.0
2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* examples/test-mp4.c:
* gst/rtsp-server/rtsp-stream.c:
stream: listen to sender ssrc signals
https://bugzilla.gnome.org/show_bug.cgi?id=746747
2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
common: update for new suppression
Makes check-valgrind pass with glib 2.46
2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Take reference to media that will be prepared
default_prepare() takes a transfer-none reference GstRTSPMedia object.
Later on a g_idle_source_new() is created and a pointer to the media
object is passed as user data. If the media is freed before the idle
source is dispatched the media object pointer is invalid, but the idle
source callback expects it to still be valid. To fix this a reference to
the media object is taken when registering the source callback function
and a corresponding release of the reference is done when the souce is
destroyed.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-record.c:
* examples/test-uri.c:
rtsp-server: Fix memory leaks when context parse fails
When g_option_context_parse fails, context and error variables are not getting free'd
which results in memory leaks. Free'ing the same.
And replacing g_error_free with g_clear_error, which checks if the error being passed
is not NULL and sets the variable to NULL on free'ing.
https://bugzilla.gnome.org/show_bug.cgi?id=753863
2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.6.0 ===
2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.6.0
=== release 1.5.91 ===
2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.5.91
2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-stream.c:
stream: fix docs for recently-added get/set_buffer_size API
https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't crash on encrypted RTX SDP
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).
https://bugzilla.gnome.org/show_bug.cgi?id=754753
2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
* examples/test-mp4.c:
test-mp4: Support filenames with spaces in them. Error out on too few arguments
2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
* examples/test-record.c:
test-record: Check parameter count and print out help
If no launch pipeline was supplied, print out some help
2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: Implement UDP buffer size setting.
Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
UDP TX buffer size.
Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Fix small typo causing gtk-doc to complain
=== release 1.5.90 ===
2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.5.90
2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: get port number through gst_rtsp_url_get_port
https://bugzilla.gnome.org/show_bug.cgi?id=753473
2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
* tests/check/gst/media.c:
media-test: Removing unnecessary assertion
https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-server.c:
Document that source keeps a ref on server until it's destroyed
https://bugzilla.gnome.org/show_bug.cgi?id=749227
2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* tests/check/gst/media.c:
media-test: Test for multiple dynamic payload
https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: Only add fakesink once per pipeline
The intention is to prevent going PLAYING state before pads are created.
If there was mutilple dynamic payload, it would leak few fakesink and
actually prevent from ever reaching playing state.
https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
Revert "rtsp-media: Only add 1 fakesink per pipeline"
This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only add 1 fakesink per pipeline
There should be only one fakesink per pipeline, not per dynpay. This
would lead to element naming clash.
2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: assertion error due to wrong condition check
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Strip keys from the fmtp that we use internally in our caps
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps
https://bugzilla.gnome.org/show_bug.cgi?id=753009
2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-thread-pool.c:
threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
https://bugzilla.gnome.org/show_bug.cgi?id=752640
2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From f74b2df to 9aed1d7
2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.5.2 ===
2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.5.2
2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* tests/check/gst/client.c:
rtsp-client: allow application to decide what requirements are supported
Add "check-requirements" signal and vfunc to allow application
(and subclasses) to check the requirements.
Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
https://bugzilla.gnome.org/show_bug.cgi?id=749417
2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* common:
Automatic update of common submodule
From 6015d26 to f74b2df
2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Always use real payloader when creating streams
A bin that contains the real payloader might be used as payloader. In this
case we have to get the real payloader for the various properties it provides.
Example use cases for this are bins that payload some media and then have
additional elements that add metadata or RTP extension headers to the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=750800
2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
* examples/test-netclock.c:
test-netclock: Use new ntp-time-source property on rtpbin
Select the clock time to be used as NTP time source. This allows proper
synchronization between receivers, independent of sharing base times, and just
requires them to use the same clock.
2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
* examples/test-netclock.c:
test-netclock: Setting the same base time on sender and receiver is not necessary
It's going to be fixed up by rtpbin when using ntp-sync=TRUE
2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* docs/libs/gst-rtsp-server.types:
docs: add missing types
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* docs/libs/gst-rtsp-server-sections.txt:
docs: add missing apis
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
GstRTSPAuth: Add client certificate authentication support
https://bugzilla.gnome.org/show_bug.cgi?id=750471
2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
test-netclock-client: Use new GstClock API to wait for clock synchronization
2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock-client.c:
test-netclock-client: Use a GMainLoop and playbin's source-setup signal
A mainloop is needed to get glimagesink to display something on OSX, and
the source-setup signal just makes things a little bit easier.
2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From d9a3353 to 6015d26
2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From d37af32 to d9a3353
2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 21ba2e5 to d37af32
2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From c408583 to 21ba2e5
2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
* docs/libs/Makefile.am:
docs: remove variables that we define in the snippet from common
This is syncing our Makefile.am with upstream gtkdoc.
2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 44a3517 to c408583
2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.5.1 ===
2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.5.1
2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: No flush during Teardown.
When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
backlog is empty it can happen that just a part of a message will be
sent and rest is in backlog queue. If then flush during teardown
just a part of message will be sent.This can lead to client miss
teardown response since it expect to get the last part of message.
The flushing during teardown was introduced to fix a deadlock that now
is fixed more generally in handle_request by temporary setting backlog
size to unlimited.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/Makefile.am:
tests: Use AM_TESTS_ENVIRONMENT
Needed by the new automake test runner and the
current version of the common submodule.
2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.h:
rtsp-server: Use single-include rtsp header to make sure we get all definitions
2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Mark some more functions static
2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only unblock the media in suspend() when actually changing the state
Otherwise we're going to lose a few packets for live streams during DESCRIBE.
2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-video-rtx.c:
examples: Use AVPF profile for the RTX example
2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Only add RTX to the SDP when using a feedback profile
2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: get valid clock-rate from last-sample
clock-rate in last-sample's caps is integer, not unsigned.
To get this value properly, variable needs to be type-casted to int.
https://bugzilla.gnome.org/show_bug.cgi?id=747614
2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
* autogen.sh:
* common:
autogen.sh: only run autopoint if gettext requested in configure.ac
Not just because there happens to be a po directory.
https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
Revert "configure.ac: uncomment gettext version setup"
This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
We don't need a gettext setup here and there's no po
directory either, so no reason why autopoint would be
run in the first place.
See https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
* examples/test-multicast.c:
* examples/test-multicast2.c:
* examples/test-sdp.c:
* examples/test-video-rtx.c:
* examples/test-video.c:
* tests/test-cleanup.c:
* tests/test-reuse.c:
Fix timeout function signatures across tests and examples
2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/Makefile.am:
tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
Make sure the test environment is set up.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: bump automake requirement to 1.14 and autoconf to 2.69
This is only required for builds from git, people can still
build tarballs if they only have older autotools.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* configure.ac:
configure.ac: uncomment gettext version setup
Fixes autogen.sh. It would run autopoint, which would complain
that it could not find the gettext version in configure.ac.
https://bugzilla.gnome.org/show_bug.cgi?id=748058
2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* examples/test-video-rtx.c:
test-video-rtx: set exact payload type to PCMA payloader
Setting wrong payload type causes failure to do retransmission through audio stream
https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: fix to get valid each stream data for request-aux-sender signal
Because of duplicated g_signal_connect for request-aux-sender signal,
wrong stream pointer is passed to the signal handler.
Instead of passing each stream, pass stream array and get the relevant stream.
https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
* acinclude.m4:
* autogen.sh:
Update autogen.sh to latest version from common
Fixes build after aclocal_check etc. helpers have been removed.
2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From bc76a8b to c8fb372
2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Limit the queues to 1 buffer
We only need them to be able to pre-roll, queueing up more data here
is only going to harm latency and memory usage.
2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Update comment and ASCII art to the latest code
We have a queue in front of the udpsink too to prevent the pipeline from
locking up.
2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-media: Properly return first rtptime
Instead we where returning first GstBuffer timestamp. This would result
in clock skew and unwanted behaviour in RTSP playback.
https://bugzilla.gnome.org/show_bug.cgi?id=746479
2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't leave buffer mapped
If the seq is NULL, the RTP buffer was left mapped. We should always
unmap the buffer.
2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
* README:
Fix typo in README
2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* tests/check/gst/client.c:
Fix double semicolons
2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
This gives more accurate values than asking the payloader. There might be
queueing happening between the payloader and the sink.
https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't seek for PLAY if the position will not change
https://bugzilla.gnome.org/show_bug.cgi?id=745704
2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't include payload type in the caps for framesize
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: add payload type to the sdp framesize attribute
The sdp framesize attribute is desribed in RFC6064. It is specified
for payloading of H263 and has the following form
a=framesize:<payload type> <width>-<height>. The <width>-<height> part
should be added to the caps in a payloader and the <payload type> should
be added by the rtsp-server.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
* examples/test-uri.c:
examples: test-uri: fix tainted variable
Insignificant but this keeps Coverity happy.
CID #1268404
2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-netclock-client.c:
* examples/test-netclock.c:
examples: Add a simple example of network synch for live streams.
An example server and client that works for synchronising live streams
only - as it can't support pause/play.
2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
rtsp-media-factory: Add functions to set/get the media gtype
Allow specifying the GType of a GstRtspMedia subclass to create
as a simpler way to get the factory to create a custom
GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix double unlock in _get_buffer_size()
Fixes an abort when calling gst_rtsp_media_get_buffer_size()
because of double g_mutex_unlock () usage.
https://bugzilla.gnome.org/show_bug.cgi?id=745434
2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp-session: Use monotonic time for RTSP session timeout
Changed RTSP session timeout handling to monotonic time
and deprecating the API for current system time.
This fixes timeouts when the system time changes.
https://bugzilla.gnome.org/show_bug.cgi?id=743346
2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-client: Only error out in PLAY if seeking actually failed
If the media was just not seekable, we continue from whatever position we are
and let the client decide if that is what is wanted or not.
Only if the actual seek failed, we can't really recover and should error out.
2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Add necessary queues between tee and multiudpsink
https://bugzilla.gnome.org/show_bug.cgi?id=744379
2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-media: If seeking fails, don't wait forever for the media to preroll again
Instead error out properly the same way as if the SEEKING query already
failed.
2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: minor code formatting fix
2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix logic for collect_streams
Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
all streams it knows if it got any, and can check if the transport mode is OK.
CID #1268400
2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Don't set the transport mode based on what elements we find
Just print a warning if the one that was set before disagrees with what
elements we found. It must already be set to something before as this
function is called after we received the SDP from ANNOUNCE in RECORD mode,
and we would reject ANNOUNCE if the RECORD flag was not set.
2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: rtspserver: rename shadowed variable
We have two different 'sink' variables here,
rename one of them for clarity.
2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix awkward if clause
2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-uri.c:
examples: test-uri: improve uri argument handling and accept file names
Print an error if the argument passed is not a URI and can't
be converted into one, or no arguments have been provided.
2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-uri.c:
examples: test-uri: don't remove mount point after 10 seconds
It's very irritating when trying to test stuff repeatedly
and serves no real purpose other than showing that it can
be done.
2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/.gitignore:
examples: add new test-record to .gitignore
2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* tests/check/gst/rtspserver.c:
rtsp-media: Use flags to distinguish between PLAY and RECORD media
2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
test-record: Set latency for playback-style example to 2s instead of 200ms
2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: add some unit tests for ANNOUNCE and RECORD
https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix a couple of leaks in handle_announce
2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Expose latency setting for setting the rtpbin latency
2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-record.c:
test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/Makefile.am:
* examples/test-record.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
Add initial support for RECORD
We currently only support media that is RECORD or PLAY only, not both at once.
https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: RTCP and RTP transport cache cookies seperated
RTCP packets were not sent because the same tr_cache_cookie was used for
both RTP and RTCP. So only one of the tr_cache lists were populated
depending on which one was sent first. If the tr_cache list is not
populated then no packets can be sent. Most often this happened to be
RTCP. Now seperate RTCP and RTP transport cache cookies are added which
resulted in both the tr_cache_lists to be populated regardless of which
one was sent first.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fix false compiler warning
rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: log interleaved data received
2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Use a random session ID in the SDP
RFC4566 Section 5.2 says that it should make the username, session id,
nettype, addrtype and unicast address tuple globally unique. Always using
1188340656180883 is not going to guarantee that: https://xkcd.com/221/
Instead let's create a 64 bit random number, which at least brings us
closer to the goal of global uniqueness.
https://tools.ietf.org/html/rfc4566#section-5.2
2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-uri.c:
examples: Don't call gst_init() and gst_get_option_group()
The latter calls the former at the appropriate time.
2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Drop trailing \0 of RTSP DATA messages
We add a trailing \0 in GstRTSPConnection to make parsing of
string message bodies easier (e.g. the SDP from DESCRIBE) but
for actual data this means we have to drop it or otherwise
create invalid data.
2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
Fixes crash when two threads access handle_new_sample() at the same
time, one for RTP, one for RTCP.
Otherwise, when iterating over the transports cache, it might be modified by
another thread at the same time if the transports cookie has changed.
https://bugzilla.gnome.org/show_bug.cgi?id=742954
2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set format=TIME on our app sources for TCP
2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-session-pool.c:
Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
RFC 2326 states that session IDs may consist of alphanumeric as well as
the safe characters $-_.+ -- N.B. the percent character is not allowed.
Previously the session ID was URI-escaped, this meant that any character
which was not alphanumeric or any of the characters +-._~ would be
percent encoded. While the RFC (surprisingly) mentions that linear white
space in session IDs should be URI-escaped, it does not say anything
about other characters. Moreover no white space is allowed in the
session ID. Finally the percent character which is the result of
URI-escaping is not allowed in a session ID.
So there is no reason to do any URI-escaping, and now it is removed.
https://bugzilla.gnome.org/show_bug.cgi?id=742869
2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From f2c6b95 to bc76a8b
2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
Fix 'make check' from top-level directory
2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-uri.c:
examples: Add command-line parsing and take a 'port' argument
This allows users to run multiple servers on different ports for testing.
Only done for examples that actually take arguments and hence are capable of
outputting different streams for each instance on each port.
https://bugzilla.gnome.org/show_bug.cgi?id=742115
2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: Add a send_message default signal handler
This allows subclasses to easily hook into the response sending
mechanism without doing everything from a signal, which seems
awkward from subclasses.
2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From ef1ffdc to f2c6b95
2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* Makefile.am:
* configure.ac:
configure: add --disable-examples switch
https://bugzilla.gnome.org/show_bug.cgi?id=741678
2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-video-rtx.c:
examples: add a retransmisison example implementing RFC4588
Currently only SSRC-multiplexed rtx streams are supported
2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix some minor memory leaks
2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Some minor cleanup
2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix compiler warnings
rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
^
rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
g_return_if_fail (GST_IS_RTSP_STREAM (stream));
^
2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
media: implement ssrc-multiplexed retransmission support
based off RFC 4588 and the server-rtpaux example in -good
2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp: Ref transports in hash table.
Also ref streams for transports.
This solves a crash when reciving a rtcp after teardown but before
client finalize.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From 7bb2bce to ef1ffdc
2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: refactor cleanup of cached media
2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/client.c:
tests: Remove FIXME
The session leak is now fixed, lets remove those FIXME comments.
2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/rtspserver.c:
tests: Test to setup two sessions on one connection
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/rtspserver.c:
tests: Test setup with tcp transport
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Configure transport after creating session media
The default implementation of configure_client_transport() in
rtsp-client uses the session media when it chooses channels for
interleaved traffic.
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
client: Stop caching media in client when doing setup
If the media has been managed by a session media, it should not be
cached in the client any longer. The GstRTSPSessionMedia object is now
responsible for unpreparing the GstRTSPMedia object using
gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
session media.
https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: unref srtp decoder when leaving bin
https://bugzilla.gnome.org/show_bug.cgi?id=739481
2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: mikey memory leaks
https://bugzilla.gnome.org/show_bug.cgi?id=739383
2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From 84d06cd to 7bb2bce
2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
* Makefile.am:
Parallelise 'make check-valgrind'
2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From a8c8939 to 84d06cd
2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 36388a1 to a8c8939
2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: deactivate media when shutting down from paused
This was only done when going directly from playing.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-context.h:
rtsp-client: add stream transport to context
We add the stream transport to the context so we can get the configured
client stream transport in the setup request signal.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-stream.c:
stream: release lock even not all transports have been removed
We don't want to keep the lock even we return FALSE because not all the
transports have been removed. This could lead into a deadlock.
https://bugzilla.gnome.org/show_bug.cgi?id=737797
2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
client: set session media to NULL without the lock
We need to set session medias to NULL without the client lock otherwise
we can end up in a deadlock if another thread is waiting for the lock
and media unprepare is also waiting for that thread to end.
https://bugzilla.gnome.org/show_bug.cgi?id=737690
2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Set state to UNPREPARING in all cases
2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
* gst/rtsp-server/rtsp-media.c:
media: set state to unpreparing when unprepare is initiated
https://bugzilla.gnome.org/show_bug.cgi?id=737675
2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Remove backlog limit while processings requests
If the backlog limit is kept two cases of deadlocks may be
encountered when streaming over TCP. Without the backlog
limit this deadlocks can not happen, at the expence of
memory usage.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: do not free main context before rtsp watch
https://bugzilla.gnome.org/show_bug.cgi?id=737110
2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
* tests/check/gst/rtspserver.c:
tests: Extend unit test timeout to accomodate for valgrind
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
rtsp-*: Treat sending packets to clients as keepalive
As long as gst-rtsp-server can successfully send RTP/RTCP data to
clients then the client must be reading. This change makes the server
timeout the connection if the client stops reading.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Allow backlog to grow while expiring session
Allow the send backlog in the RTSP watch to grow to unlimited size while
attempting to bring the media pipeline to NULL due to a session
expiring. Without this change the appsink element cannot change state
because it is blocked while rendering data in the new_sample callback.
This callback will block until it has successfully put the data into the
send backlog. There is a chance that the send backlog is full at this
point which means that the callback may block for a long time, possibly
forever. Therefore the media pipeline may also be prevented from
changing state for a long time.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Make old compilers happy
rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
Just in case that guint8 doesn't fit in a pointer. Just in case ...
2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: raise the backlog limits before pausing
We need to raise the backlog limits before pausing the pipeline or else
the appsink might be blocking in the render method in wait_backlog() and
we would deadlock waiting for paused.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: make define for the WATCH_BACKLOG
See https://bugzilla.gnome.org/show_bug.cgi?id=736322
2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: simplify session transport handling
link/unlink of the transport in a session was done to keep track of all
TCP transports and to send RTP/RTCP data to the streams. We can simplify
that by putting all the TCP transports in a hashtable indexed with the
channel number.
We also don't need to link/unlink the transports when we pause/resume
the streams. The same effect is already achieved when we pause/play the
media. Indeed, when we pause the media, the transport is removed from
the media and the callbacks will not be called anymore.
See https://bugzilla.gnome.org/show_bug.cgi?id=736041
2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
stream-transport: make method to handle received data
Make a method to handle the data received on a channel. It sends the
data to the stream of the transport on the RTP or RTCP pads based on
the channel number.
2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
* examples/test-mp4.c:
test: add example of dumping RTCP reports
2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-media: Make sure that sequence numbers are monotonic after pause
The sequence number is not monotonic for RTP packets after pause. The
reason is basepayloader generates a randon sequence number when the
pipeline goes from ready to pause. With this fix generation of sequence
number will be monotonic when going from pause to play request.
https://bugzilla.gnome.org/show_bug.cgi?id=736017
2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Protect saved clients watch with a mutex
Fixes a crash when close() is called while merging clients
in handle_tunnel(). In that case close() would destroy the
watch while it is still being used in handle_tunnel().
https://bugzilla.gnome.org/show_bug.cgi?id=735570
2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Remove the multicast group udp sources when removing from the bin
2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-media: Query position and stop time only on the RTP parts of the pipeline
The RTCP parts, in specific the RTCP udpsinks, are not flushed when
seeking and will always continue counting the time. This leads to
the NPT after a backwards seek to be something completely different
to the actual seek position.
https://bugzilla.gnome.org/show_bug.cgi?id=732644
2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
* examples/test-appsrc.c:
examples: fix another reference leak
gst_rtsp_media_get_element() returns a new ref.
2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* examples/test-appsrc.c:
examples: unref element after usage
gst_bin_get_by_name_recurse_up() returns an element
reference that must be unreffed after usage.
https://bugzilla.gnome.org/show_bug.cgi?id=734546
2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
* gst/rtsp-server/rtsp-media.c:
signals: Fix copy-pasto in target-state signal offset
2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
* Makefile.am:
* common:
Makefile: Add usage of build-checks step
Allows building checks without running them
2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Listen on the multicast group for RTP/RTCP packets
When a UDP multicast transport is used it is expected that the server listens
for RTP and RTCP packets on the multicast group with the corresponding port.
Without this we will never get RTCP packets from clients in multicast mode.
https://bugzilla.gnome.org/show_bug.cgi?id=732238
2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.4.0 ===
2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.4.0
2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
* gst/rtsp-server/rtsp-media.h:
media: correct misspelled words in description
https://bugzilla.gnome.org/show_bug.cgi?id=733244
=== release 1.3.91 ===
2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.91
2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
* docs/libs/gst-rtsp-server-sections.txt:
docs: update docs
2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-server.c:
server: implement client REMOVE filter
2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: expose _close() method
Expose a previously internal close method to close the client
connection.
2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-pool.c:
session-pool: signal session-removed outside of the lock
Release the lock before emiting the session-removed signal.
2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream.c:
filter: Release lock in filter functions
Release the object lock before calling the filter functions. We need to
keep a cookie to detect when the list changed during the filter
callback. We also keep a hashtable to make sure we only call the filter
function once for each object in case of concurrent modification.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: check if watch is set in handle_teardown()
The unit tests run without a watch
2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
* tests/check/gst/client.c:
client tests: send teardown to cleanup session
2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
* tests/check/gst/rtspserver.c:
server tests: send teardown to cleanup session
2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: keep ref to client for the session removed handler
This extra ref will be dropped when all client sessions have been
removed. A session is removed when a client sends teardown, closes its
endpoint of the TCP connection or the sessions expires.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session.c:
* tests/check/gst/client.c:
client: manage media in session as a last step
Once we manage a media in a session, we can't unmanage it anymore
without destroying it. Therefore, first check everything before we
manage the media, otherwise if something is wrong we have no way to
unmanage the media.
If we created a new session and something went wrong, remove the session
again. Fixes a leak in the unit test.
2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
* examples/test-mp4.c:
* examples/test-ogg.c:
examples: print 'stream ready at url' for mp4 and ogg example
2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
rtsp: fix for MIKEY api change
2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: free watch context only once
The watch context is freed when the source is destroyed. Avoids
a CRITICAL when we try to unref the context twice.
2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: fix build
2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: protect sessions with lock
Protect the list of sessions with the lock.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
Client: keep a ref to the session
Don't just keep a weak ref to the session objects but use a hard ref. We
will be notified when a session is removed from the pool (expired) with
the new session-removed signal.
Don't automatically close the RTSP connection when all the sessions of
a client are removed, a client can continue to operate and it can create
a new session if it wants. If you want to remove the client from the
server, you have to use gst_rtsp_server_client_filter() now.
Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
See https://bugzilla.gnome.org/show_bug.cgi?id=732226
2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
session-pool: add session-removed signal
Add a signal to be notified when a session is removed from the pool.
2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-server.h:
Make rtsp-server.h a single-include header, use it for G-I
https://bugzilla.gnome.org/show_bug.cgi?id=732411
=== release 1.3.90 ===
2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.90
2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: crypto can be NULL
2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
introspection: add missing allow-none annotations
https://bugzilla.gnome.org/show_bug.cgi?id=730952
2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-token.c:
introspection: add (nullable) annotations to return values
https://bugzilla.gnome.org/show_bug.cgi?id=730952
2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
gi: improve annotations
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server.c:
signals: use generic marshal function
Use the generic C marshal function.
Use more explicit type instead of G_TYPE_POINTER
2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-context.h:
context: add type macro
2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
sdp: hide key length defines
They don't have a namespace.
2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.3.3 ===
2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.3
2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
mikey: add different key length parameters
Add encryption and authentication key length parameters to MIKEY. For
the encoders, the key lengths are obtained from the cipher and auth
algorithms set in the caps. For the decoders, they are obtained while
parsing the key management from the client.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
2014-03-16 17:29:48 +0100 Ognyan Tonchev <otonchev@gmail.com>
* tests/check/gst/stream.c:
stream tests: Make sure we get right multicast address from stream
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
2014-06-12 13:49:17 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: ref the context until rtsp watch is alive
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
2014-06-12 13:48:44 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Destroy the rtsp watch after connection close
2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: fix confusing comment
2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-session.c:
rtsp-session: Timeout in header.
Adding the possbilty to always have timout in header.
This is configurabe with setting "timeout-always-visible".
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
2014-05-21 13:23:40 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.3.2 ===
2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* common:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.2
2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From 211fa5f to 1f5d3c3
2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: store TCP ports in transport
Store the TCP ports in the transport when we are doing RTSP over TCP.
This way, we can easily get to the ports from the transport.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-stream.c:
stream: add signals for new RTP/RTCP encoders
New signals to allow the user to configure the dynamically created
encoders.
https://bugzilla.gnome.org/show_bug.cgi?id=730228
2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: Make suspend()/unsuspend() virtual
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
client: fix send-message signal marshaller
Use generic marshalling for the send-message signal. It has
two POINTER arguments, not just one.
https://bugzilla.gnome.org/show_bug.cgi?id=729900
2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
* tests/check/gst/media.c:
tests: add and remove pads only once
In this test we simulate a dynamic pad by watching the caps event.
Because of renegotiation in the base payloader now, this caps is sent
multiple times but we can only deal with 1 invocation, use a variable to
only 'add and remove' the pad once.
2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/rtspserver.c:
tests: add unit test for correct handling of Require headers
https://bugzilla.gnome.org/show_bug.cgi?id=729426
2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
Servers must handle Require headers and must report a failure
if they don't handle any of the Required options, see RFC 2326,
section 12.32: https://tools.ietf.org/html/rfc2326#page-54
https://bugzilla.gnome.org/show_bug.cgi?id=729426
2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.3.1 ===
2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.3.1
2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From bcb1518 to 211fa5f
2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
Update .gitignore
2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/gst/sessionmedia.c:
tests: fix memory leak in sessionmedia unit test
2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: emit a signal before sending a message
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: pass context to send_message
Pass the current context to send_message, we will need it later.
2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: fix typo in comment
2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
media: Do not stop thread twice if default_prepare() fails
2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: set the watch to flushing before going to NULL
First set the watch to flushing so that we unblock any current and
future attempt to send data on the watch, Then set the pipeline to
NULL.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
* gst/rtsp-server/rtsp-session-pool.c:
* tests/check/gst/sessionpool.c:
rtsp-session-pool: Fixes annotation
Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
in the sessionpool test.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: make media_prepare virtual
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
media: stop the thread in more error cases
2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
media: allow NULL as the thread
Use the default context whan passing a NULL thread.
2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: indent cleanup
Coverity was moaning about unreachable code, and I think it was just
confused by { being before the label. We'll see if it pops up again.
Coverity 1197705
2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
client: Add drop-backlog property
When we have too many messages queued for a client (currently hardcoded
to 100) we overflow and drop the messages. Add a drop-backlog property
to control this behaviour. Setting this property to FALSE will retry
to send the messages to the client by waiting for more room in the
backlog.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: support for POST before GET when setting up a tunnel
2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: remove watch of the second client after http tunnel setup
The second client will be freed after the HTTP tunnel has been set up.
Make sure it's RTSP watch is never dispatched again.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
media: Make media_prepare() fail if port allocation fails
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
* tests/check/gst/media.c:
media test: cleanup the thread pool in tests
2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
rtsp-media: Unblock blocked streams in unprepare
The streams will be blocked when a live media is prepared.
The streams should be unblocked in gst_rtsp_media_unprepare.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: release the state lock when going to NULL
Set our state to UNPREPARING and release the state-lock before
setting the pipeline to the NULL state. This way, any pad-added
callback will be able to take the state-lock and check that we are now
unpreparing instead of deadlocking.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: protect status with lock
Make sure we only update the status with the lock.
2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
rtsp: update for MIKEY API changes
2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: parse the mikey response from the client
Parse the mikey response from the client and update the policy for
each SSRC.
2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add method to set crypto info
Make a method to configure the crypto information of a stream.
Set udpsrc in READY instead of PAUSED so that we can configure caps
later.
2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: cleanup error paths
2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: fix docs
2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
* examples/test-video.c:
test: enable SRTP only on RTSPS
We only want to enable SRTP when doing rtsp over TLS so that we can
exchange the keys in a secure way.
2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
* examples/test-video.c:
test: print an error on failure
2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
* configure.ac:
* examples/test-video.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/Makefile.am:
stream: add SRTP support
Install srtp encoder and decoder elements in rtpbin
Add MIKEY in SDP
2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/Makefile.am:
* tests/check/gst/sessionpool.c:
tests: Add unit tests for sessionpool
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/gst/threadpool.c:
tests: Improve code coverage of rtsp-threadpool tests
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/gst/sessionmedia.c:
tests: Improve code coverage for rtsp-session-media
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
gobject-introspection: Add annotations to support language bindings
In addition a few cosmetic changes:
* Adjust the order of arguments
* Fix typo: occured -> occurred
* Fix indentation after Return:-clauses
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Don't mix IPv4 and IPv6 addresses
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: take caps after the session manager
Take the caps for the SDP after they leave the rtpbin so that we can
also get the properties added by rtpbin elements.
2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: release lock while pushing out packets
Keep a cache of the transports and use this to iterate the transport
while pushing packets. This allows us to release the lock early.
See https://bugzilla.gnome.org/show_bug.cgi?id=725898
2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: vmethod for modifying tunnel GET response
Add a vmethod tunnel_http_response where the response to the HTTP GET
for tunneled connections can be modified.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-sdp.c:
sdp: make 1 media line per profile
If we have multiple profiles (AVP or AVPF) for a stream, make one m=
line in the SDP for each profile. The client is then supposed to pick
one of the profiles in the SETUP request. Because the m= lines have the
same pt, the client also knows that only 1 option is possible.
2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
factory: add profile property and pass to media and streams
2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
* examples/test-multicast.c:
* gst/rtsp-server/rtsp-sdp.c:
sdp: pass multicast connection for multicast-only stream
Pass the multicast address of the stream in the connection info in the
SDP so that clients try a multicast connection first.
Only allow multicast connections in the test-multicast example. Also
increase the TTL a little.
2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* .gitignore:
.gitignore: Ignore gcov intermediate files
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: release some locks in error cases
2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
docs: Enable and fix gtk-doc warnings
* Makefile: Enable gtk-doc warnings, like the rest of GStreamer
* addresspool/mediafactory: Add missing annotation colon
* stream: Annotate return value
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From fe1672e to bcb1518
2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 1a07da9 to fe1672e
2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/Makefile.am:
examples: use LDADD for libs instead of LDFLAGS
2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: make sure releases are in .doap file
2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-cgroups.c:
examples: test-cgroups: don't put code with side effects into g_assert()
The g_assert() might get compiled out with the right
compiler/preprocessor flags.
2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/.gitignore:
examples: add cgroup test binary to .gitignore
2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
* examples/test-cgroups.c:
examples: fix cgroup test build
Fixes build failure caused by compiler warning:
test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
.gitignore: ignore temp files created in the course of 'make check'
2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: don't loose frames handling new PLAY request
If client supplied a range check if the range specifies the start point.
If not, then do an accurate seek to the current position. If a start
point was specified do do a key unit seek to make sure the streaming
starts with decodeable frames.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
Revert "media: only flush when setting a new start position"
This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
We need to do the flush in all cases, demuxer block currently for
non-flushing seeks.
2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: only flush when setting a new start position
Only flush the pipeline when we change the start position with
a seek.
See https://bugzilla.gnome.org/show_bug.cgi?id=724611
2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
stream: set ttl-mc before adding the socket
Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
never be set on socket.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-media.c:
media: stop thread if media is already prepared
in gst_rtsp_media_prepare() the thread is not used if media is already
prepared (e.g. media shared) so we want to stop the thread. otherwise, a
leak occurs.
https://bugzilla.gnome.org/show_bug.cgi?id=724182
2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
* Makefile.am:
build: Ship gst-rtsp-server.doap file
2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
* tests/check/gst/rtspserver.c:
tests: Fix another compiler warning with gcc
2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
rtsp-server: Fix lots of compiler warnings with clang
2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
* gst-rtsp-server.doap:
* tests/Makefile.am:
configure: Synchronise with the configure scripts of the other modules
2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
Revert "rtsp-server: support build against last stable release"
This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
Let us require 1.2.3 now, which is going to be released in a few
minutes.
2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
session: improve RTP-Info
Ignore streams that can't generate RTP-Info instead of failing.
Don't return the empty string when all streams are unconfigured but
return NULL so that we don't generate and empty RTP-Info header.
Improve docs a little.
2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
* gst/rtsp-server/rtsp-session-media.c:
Don't free rtpinfo GString when it is NULL
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: only set keyframe flag when modifying start
Only set the keyframe flag when we modify the start position. The
keyframe flag should probably be ignored when no change is requested but
until we can claim this is all documented properly and all demuxer
implement this, avoid setting the flag.
See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-thread-pool.c:
thread-pool: Unref source after mainloop has quit to avoid races in GLib
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: handle NULL seqnum and rtptime arguments
2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-thread-pool.c:
* tests/check/gst/threadpool.c:
thread-pool: Unref reused threads in gst_rtsp_thread_stop()
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: add fallback for missing stats property
Use a fallback when the payloader does not have a stats property
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From f7bc1c3 to 1a07da9
2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: don't leak stats structure
Don't leak the stats structure and deal with NULL stats.
2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Get rtpinfo properties atomically from payloader
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: refactor state change functions and signals
Make functions to set the target state and the pipeline state and emit
the signals from those functions.
2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add signal to notify of pending state changes
2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: support build against last stable release
Until 1.2.3 is out with the new get_type function and we
can require that.
2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
stream: fix compilation
2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add property to configure profiles
2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: let stream check supported transport
Delegate the check if a transport is allowed to the stream.
See https://bugzilla.gnome.org/show_bug.cgi?id=720696
2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add method to check supported transport
Add a method to check if a transport is supported
2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
configure.ac: Only check for gstreamer-check, not check
We include check in gstreamer-check since quite some time now.
2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: return clock-rate from get_rtpinfo
And use it to correct the rtptime to the requested start-time.
See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
session-media: calculate start-time
2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: also return the running-time
Return the running-time in the rtpinfo as well.
2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
session-media: let the session-media make the RTPInfo
Add method to create the RTPInfo for a stream-transport.
Add method to create the RTPInfo for all stream-transports in a
session-media.
Use the session-media RTPInfo code in client. This allows us to refactor
another method to link the TCP callbacks.
2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
mount-points: sort sequence before g_sequence_lookup
* gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
sort sequence if dirty, otherwise lookup will fail.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: rename package from gst-rtsp to gst-rtsp-server
To match git module name and avoid confusion with the
rtsp lib in gst-plugins-base and rtsp plugin in -good.
2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
configure: bump core/base/good requirement to 1.2.0
Bump to released stable version and make implicit
requirements explicit.
2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
* autogen.sh:
* common:
* configure.ac:
Fix broken gettext setup which is not used anyway
2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From dbedaa0 to d48bed3
2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add setup_sdp vmethod
gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
gst_rtsp_media_setup_sdp.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Check return value of sscanf
streamid is only valid if sscanf matched something.
2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Fix iteration
Wouldn't even enter the code block otherwise (i++ was used as the check
and not the postfix).
2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add vmethod to configure media and streams
Implement a vmethod that can be used to configure the media and the
streams based on the current context. Handle the blocksize handling in
the default handler.
See https://bugzilla.gnome.org/show_bug.cgi?id=720667
2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
Make git ignore more unit test binaries
2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-context.h:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-server/rtsp-thread-pool.h:
* gst/rtsp-server/rtsp-token.h:
rtsp-server: add padding to many public structures
Not mini objects though, since they are not subclassable
anyway, nor kept on the stack or inlined in a structure.
2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
media: add new create_rtpbin vmethod
* gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
https://bugzilla.gnome.org/show_bug.cgi?id=719734
2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
* tests/check/gst/media.c:
tests: fix memory leak, free test's thread pool
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream-transport.c:
stream-transport: free url in finalize
2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
media: also do state change in suspended state
2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
media: also handle prepare and range in suspended state
When we are suspended, we are already prepared.
We can get the range in the suspended state.
2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
* tests/check/Makefile.am:
* tests/check/gst/sessionmedia.c:
check: add test for uri in setup
Added unit tests for the new functionality in GstRTSPStreamTransport.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: store setup uri and use in PLAY response
Store the uri used when doing the setup and use that in the PLAY
response.
fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
stream-transport: add method to get/set url
2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: suspend after SDP and unsuspend before PLAYING
Based on patches by Ognyan Tonchev <ognyan@axis.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session.c:
* tests/check/gst/media.c:
* tests/check/gst/mediafactory.c:
media: add suspend modes
Add support for different suspend modes. The stream is suspended right after
producing the SDP and after PAUSE. Different suspend modes are available that
affect the state of the pipeline. NONE leaves the pipeline state unchanged and
is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
state and RESET will bring the pipeline to the NULL state.
A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
this means that the pipeline needs to be prerolled again.
Base on patches by Ognyan Tonchev <ognyan@axis.com>
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: start live streams in blocked state
Start live streams in the blocked state and make them preroll using the
messages. This ensure that no data is played by the sink until we explicitly
unblock the stream right before going to PLAYING.
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: refactor starting and waiting for preroll
Based on patches from Ognyan Tonchev <ognyan@axis.com>
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add API to block streams
Add an API to block on the streams and make it post a message.
Based on patch by Ognyan Tonchev <ognyan@axis.com>
See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
* docs/libs/Makefile.am:
docs: Specify the override file
Even if it's empty (for now) it avoids make distcheck complaining
2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: move default implementations to where they are used
2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: take the right lock in gst_rtsp_media_set_pipeline_state()
We need to take the state_lock when calling this method.
2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
media: handle add-added on non-bins too
Handle dynamic payloaders that are not bins, as used in the unit-test.
2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
rtsp-media/-factory: Fix request pad name comments
These must be escaped for gtk-doc to parse the comments without warnings.
2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
rtsp-media: remove transports if media is in error status
* gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
trying to change to GST_STATE_NULL and media is in error status, we
remove all transports.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: use element metadata to find payloader
Use the element metadata to find the payloader instead of checking
for the base class.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
rtsp-stream: add getter for payload type
* gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
* gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
element and create the stream with this one instead of the dynpay%d
element.
https://bugzilla.gnome.org/show_bug.cgi?id=712396
2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-context.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-token.c:
rtsp-*: Refer to NULL as a constant in comments
Plus one typo fix.
https://bugzilla.gnome.org/show_bug.cgi?id=714988
2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
rtsp-*: Fix type name typos in comments
* rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
* rtsp-auth: Refer to part of constant name as text
* rtsp-auth/-permissions/-token: Refer to Permissions not Permission
* rtsp-session-media: Fix GstRTSPSessionMedia typo
* rtsp-stream: Fix typo when refering to GstBin
https://bugzilla.gnome.org/show_bug.cgi?id=714988
2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* docs/README:
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
docs: Improve documentation
* Include annotation-glossary to quiet gtk-doc
* Rename remaining ClientState -> Context
* Rename object hierarchy file
* Remove stale chapter references
* Add missing function and object references
* Include missing GstRTSPAddressPoolResult
https://bugzilla.gnome.org/show_bug.cgi?id=714988
2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp-server: sprinkle some allow-none annotations for g-i
2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add method to filter transports
Add a method to safely iterate and collect the stream transports
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
rtsp: allow NULL func in filters
Passing a null function make the filters return a list of
refcounted objects.
2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp-server/rtsp-address-pool.c:
* tests/check/gst/addresspool.c:
address-pool: fix address increment
Use a guint instead of guint8 to increment the address. It's still not
completely correct because a guint might not be able to hold the complete
address range, but that's an enhacement for later.
Add unit test to test improved behaviour.
https://bugzilla.gnome.org/show_bug.cgi?id=708237
2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
client: allow absolute path in requests
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: make make_path_from_uri a vmethod
2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/Makefile.am:
* tests/check/gst/stream.c:
stream: Add functions to get rtp and rtcp sockets
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp-server/rtsp-context.c:
* gst/rtsp-server/rtsp-context.h:
context: defing a GType for the context
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-context.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream.c:
Fixed several GIR warnings
2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp-server/rtsp-auth.c:
auth: small typos
2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/Makefile.am:
* tests/check/gst/token.c:
tests: Add unit tests for token
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-token.c:
token: Validate args for gst_rtsp_token_is_allowed
See https://bugzilla.gnome.org/show_bug.cgi?id=710520
2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-token.c:
token: Fix bug when creating empty token
We always want to have a valid GstStructure in the token.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp-server/rtsp-thread-pool.c:
thread-pool: avoid race in shutdown
If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
don't actually stop the mainloop ever. Solve this race by adding an idle source
to the mainloop that calls the _quit. This way we immediately exit the mainloop
if quit was called before we started it.
2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/Makefile.am:
* tests/check/gst/permissions.c:
tests: Add unit tests for permissions
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/gst/mediafactory.c:
tests: Test mediafactory permissions
See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-permissions.c:
permissions: Fix refcounting when adding/removing roles
Previously a role that was removed was unreffed twice, and when
replacing an existing role the replaced role was freed while still being
referenced. Both bugs are now fixed.
See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/gst/media.c:
* tests/check/gst/mediafactory.c:
* tests/check/gst/rtspserver.c:
tests: Check gst_rtsp_url_parse return value
See https://bugzilla.gnome.org/show_bug.cgi?id=710202
2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From 865aa20 to dbedaa0
2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-server.c:
rtsp-server: Fix socket leak
https://bugzilla.gnome.org/show_bug.cgi?id=710088
2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-session-pool.c:
rtsp-session-pool: Make sure session IDs are properly URI-escaped
https://bugzilla.gnome.org/show_bug.cgi?id=643812
2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
* examples/.gitignore:
* examples/test-video.c:
examples: fix compilation when WITH_AUTH is defined
https://bugzilla.gnome.org/show_bug.cgi?id=710228
2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
* .gitignore:
gitignore: Add new test binary
2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
* tests/check/Makefile.am:
* tests/check/gst/threadpool.c:
thread-pool: Add unit test for the thread pools
https://bugzilla.gnome.org/show_bug.cgi?id=710228
2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-thread-pool.c:
thread-pool: Fix thread leak when reusing threads
https://bugzilla.gnome.org/show_bug.cgi?id=709730
2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-server.c:
* tests/check/gst/rtspserver.c:
tests: fixed racy behavior in rtspserver tests
https://bugzilla.gnome.org/show_bug.cgi?id=710078
2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* tests/check/gst/addresspool.c:
tests: Improve address pool unit tests
Add a range with mixed IPV4 and IPV6 addresses to pool.
Get an IPV4 address from an IPV6-only pool.
Get an IPV6 address from an IPV4-only pool.
Reserve a IPV6 address from an IPV4-only pool.
Check for unicast addresses in multicast-only pool.
Check for unicast addresses in uni-/multicast-mixed pool.
https://bugzilla.gnome.org/show_bug.cgi?id=710128
2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: append query string in PAUSE/PLAY/TEARDOWN as well
2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Add query to control path
If the SETUP url contains a query it must be appended to the control
path so that it matches any already created stream in the media. The
query will also be appended to the session media path.
2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: remove old line
2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Correct control comparison
https://bugzilla.gnome.org/show_bug.cgi?id=709176
2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: Check dynamically if the pipeline supports seeking
We should not depend on whether or not the pipeline state change
returned NO_PREROLL or not. A media could dynamically change its
element and switch from seekable to non seekable so it's best to test
the seekable nature of the pipeline dynamically when we try to do a seek.
2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: Return FALSE if seeking is not supported
2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: don't seek accurate by default
Accurate seeking is perhaps a little overkill in the most common situation and
causes some formats (mp3) over slow media to seek extremely slowly.
2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
* tests/check/gst/rtspserver.c:
tests: fix unit test
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Reply 400 if media cannot be constructed
Reply 400 Bad Request instead of 503 Service Unavailable if media
cannot be constructed in SETUP.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Send setup reply once only
If find_media() failed in handle_setup_request() two replies was sent.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 6b03ba7 to 865aa20
2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtsp-server/rtsp-server.c:
server: Emit client-connected signal earlier
Emit client-connected before the client ref is given to a GSource,
otherwise client-connected can be emitted after the client object has
been freed.
2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/addresspool.c:
addresspool: return reason of failure
Let gst_rtsp_address_pool_reserve_address() return the reason why
the address could not be reserved.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
* autogen.sh:
autogen.sh: Sync behaviour with other GStreamer modules
Allows building from outside of tree amongst other things
2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
* common:
Automatic update of common submodule
From b613661 to 6b03ba7
2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 74a6857 to b613661
2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 01a7a46 to 74a6857
2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: Do not read beyond end of path string
If the setup was done without a control url, make sure we don't try to read the
non-existing control string and crash.
2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: Fix RTPInfo header
Refactor the method to make the content_base.
Use the content-base and the control url to construct the RTPInfo
url.
2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: map url to path only in describe
Only map the request url to a path in the DESCRIBE method. The SDP then
contains the base and control urls that should be used to SETUP/PAUSE/
PLAY/TEARDOWN the media.
2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Revert "client: map URL to path in requests"
This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
contains the base and control urls which are used in the SETUP, PLAY,
PAUSE and TEARDOWN requests.
2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: map URL to path in requests
2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-mount-points.h:
mount-points: make vmethod to make path from uri
Make a vmethod to transform an url into a path. The path is then used to lookup
the factory. This makes it possible to also use other bits of the url, such as
the query parameters, to locate the factory.
2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-thread-pool.h:
thread-pool: Add cleanup to wait for the threadpool to finish
Also fix race condition if two threads are asking for the first
thread from the thread pool at once. This would case two internal
GThreadPools to be created.
https://bugzilla.gnome.org/show_bug.cgi?id=707753
2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
client: free threadpool
https://bugzilla.gnome.org/show_bug.cgi?id=707638
2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
* tests/check/gst/mountpoints.c:
mountpoints tests: unref matched factories
https://bugzilla.gnome.org/show_bug.cgi?id=707638
2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
* tests/check/gst/media.c:
media tests: unref thread pool and caps
https://bugzilla.gnome.org/show_bug.cgi?id=707638
2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
auth, media, media-factory: unref permissions
https://bugzilla.gnome.org/show_bug.cgi?id=707638
2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
Makefile: add rule for appsrc example
2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-appsrc.c:
tests: add appsrc example
Add an example on how to use appsrc to feed the server pipeline with data.
2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: remove query part from content-base string
Make sure that after the control url has been resolved, it's
not a part of the query-string.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: don't check url in response
There is no url or method in the response to check
2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Add handle-response signal for when we receive a GET_PARAMETER response
2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
Fix gst_rtsp_server_client_filter, using wrong variable type
2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst/rtsp-server/rtsp-media-factory-uri.c:
rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
For AAC we need to check for framed=true instead of parsed=true.
https://bugzilla.gnome.org/show_bug.cgi?id=701384
2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: optimize pipeline for protocols
When TCP is not an allowed protocol for the stream, avoid creating the
appsrc/appsink/queue and tee elements.
2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: set protocols on streams
2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use protocols supported by stream
2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
media-factory: allow all protocols
2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: configure protocols in new streams
2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add protocols property
2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: send state in "new-state" signal
https://bugzilla.gnome.org/show_bug.cgi?id=705110
2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
* configure.ac:
build: add subdir-objects to AM_INIT_AUTOMAKE
Fixes warnings with automake 1.14
https://bugzilla.gnome.org/show_bug.cgi?id=705350
2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: add method to iterate clients of server
2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Add vmethod for rtsp-media subclass to access rtpbin
2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-client.h:
small documentation fix
2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Do not take range header if range is invalid
2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media.c:
media: add docs for new method
2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Add API to rtsp-media set the pipeline's state
2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
Update current position/duration when gst_rtsp_media_get_range_string is called
2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-cgroups.c:
tests: add some more docs
2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-cgroups.c:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-context.c:
* gst/rtsp-server/rtsp-context.h:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-params.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-thread-pool.h:
* tests/check/gst/client.c:
ClientState -> Context
Rename the clientstate to context and put the code in a separate file.
2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
auth: add support for default token
The default token is used when the user is not authenticated and can be used to
give minimal permissions.
2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* gst/rtsp-server/rtsp-auth.c:
auth: use defines when possible
2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-address-pool.c:
address-pool: improve docs
2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-permissions.c:
permissions: add the role to the copy
2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-permissions.c:
permissions: Also copy the roles
2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-permissions.c:
permissions: Make it build
2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-address-pool.h:
docs: small fixes
2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/client.c:
docs: improve docs
2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
* tests/check/gst/addresspool.c:
* tests/check/gst/rtspserver.c:
address-pool: cleanups
Remove redundant method, improve docs.
2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-token.c:
docs: improve docs
2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-permissions.c:
permissions: implement _remove_role
2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-permissions.c:
permissions: update docs
2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/client.c:
tests: simplify tests
Client settings are now disabled by default so we don't need an auth
module to disable them.
2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
auth: add default authorizations
When no auth module is specified, use our table of defaults to look up the
default value of the check instead of always allowing everything. This was
we can disallow client settings by default.
2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/README:
README: update readme
2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-thread-pool.h:
thread-pool: add more docs
2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-thread-pool.h:
thread-pool: fix race in thread reuse
If we try to reuse a thread right after we made it stop, we end up using a
stopped thread. Catch this case and only reuse threads that are not stopping.
2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: add small debug
2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/client.c:
client: fix test
Add some permissions to media so we can use the auth and enable
client settings.
2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: support pushed context in handle_request
If we already have a pushed state, reuse it and add our own things. This makes
it easier to write tests.
2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
auth: don't auth on methods
Don't authorize on methods anymore but on the resources that we
try to access, this is more flexible.
Move the authorization checks to where they are needed and let the
check return the response on error.
2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-mount-points.c:
mount-points: add some debug
2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/client.c:
tests: almost fix test
2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
auth: let the auth module check client_settings
Let the auth module decide if client settings are allowed for the
current client.
2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-token.c:
* gst/rtsp-server/rtsp-token.h:
token: add method to check boolean permission
2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* examples/test-cgroups.c:
* gst/rtsp-server/rtsp-token.c:
* gst/rtsp-server/rtsp-token.h:
token: simplify token constructor
Use variable arguments to make easier API.
2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* examples/test-cgroups.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add convenience API for factory
2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* examples/test-cgroups.c:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-permissions.h:
permissions: simplify API a little
Avoid passing GstStructure in the add_role method, use varargs instead
to construct the structure behind the scenes. We can then also use the
structure name as the role and simplify some more logic.
2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
auth: fix typo
2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
auth: handle unauthorized response
Move handling of the unauthorized response to the auth module, it can add
the appropriate headers to request authorization for the required method
much better than the client.
2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: allow for sending any message, not only requests
Change the _send_request() method to _send_message() so that we
can both send requests and replies.
2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-server.h:
docs: fix docs
2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-video.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
auth: move TLS handling to auth module
Remove the TLS settings on the server and move it to the auth module because
that is where security related bits go.
2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add state push/pop
2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add connection to state
2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-mount-points.c:
mount-points: fix debug
2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/media.c:
tests: fix media test
2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-thread-pool.c:
thread-pool: we don't require a state
2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: let context ref the server
So that we don't risk losing the server object early anc crash.
2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/client.c:
tests: fix client test
2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/README:
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-token.c:
docs: improve docs
2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
session-pool: make vmethod to create a session
Make a vmethod to create a sessions so that subclasses can create
custom session objects
2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-stream.h:
docs: more updates
2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-thread-pool.h:
docs: update docs
2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
* examples/Makefile.am:
configure: compile cgroup example conditionally
Only compile the cgroup example when we have libcgroup
2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
* examples/Makefile.am:
* examples/test-cgroups.c:
examples: add cgroups example
2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/rtspserver.c:
tests: fix compilation
2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-thread-pool.c:
thread-pool: fix vmethod invocation
2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-thread-pool.h:
thread-pool: store thread type in thread
2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: pass thread from pool to media _prepare
Get a thread from the configured threadpool and pass it to the prepare method of
the media.
2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: Accept a thread in _prepare
Remove out own threadpool handling and use the provided thread and
maincontext for the bus messages and the state changes.
2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: configure client thread pool
2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add method to configure thread pool
2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: use thread pool
Use the thread pool instead of doing our own thing.
2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-thread-pool.c:
* gst/rtsp-server/rtsp-thread-pool.h:
thread-pool: add object to manage threads
Add an object to manage the client and media threads.
2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
auth: debug authorization check
2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: start media pipeline in context
Start the media pipeline in the provided context (or our default one
when NULL). This makes sure that we run the bus thread in this context and that
all media threads are children of this context.
2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
factory: pass permissions to media by default
2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
test: add permissions to auth test
Ass some permissions to the media factory in the test.
2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
auth: simplify auth checks
Remove client from methods, it's now in the state
Perform the check specified by the string, use the information from the
thread local context.
2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add state to current thread
Add the client to the ClientState object.
Place the ClientState on the current thread.
2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: make it possible to set permissions
Make it possible to set permissions on media and media factory objects
2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-permissions.c:
* gst/rtsp-server/rtsp-permissions.h:
permissions: add permissions object
Add a mini object to store permissions based on a role.
2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
auth: add auth checks
Add an enum with auth checks and implement the checks in the auth object.
Perform the checks from the client.
2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.h:
auth: use the token after authentication
After we authenticated a user, keep the Token around in the state.
2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* tests/check/gst/media.c:
media: add optional context for bus messages
Add an optional mainloop to _prepare that will handle the bus messages instead
of always using the shared mainloop.
2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-token.c:
* gst/rtsp-server/rtsp-token.h:
token: add authorization token
Add a simply miniobject that contains the authorizations. The object contains a
GstStructure that hold all authorization fields. When a user is authenticated,
the auth module will create a Token for the user. The token is then used to
check what operations the user is allowed to do and various other configuration
values.
2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
auth: remove auth from media and factory
Remove the auth object from media and factory. We want to have the RTSPClient
authenticate and authorize resources, there is no need to place another auth
manager on the media/factory.
2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.h:
auth: add support for multiple basic auth tokens
Make it possible to add multiple basic authorisation tokens to one authorization
object. Associate with each token an authorization group that will define what
capabilities are allowed.
2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: error out on non-aggregate control
We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: rework setup request a little
Cache the media in DESCRIBE based on the longest matching path with the uri
that we can find in the mount points.
Rework the setup request a little to get the media from the session or from
the longest matching path, this way we can derive the control string as
everything after the path instead of hardcoding it.
Find the stream based on the control string and only open a session when all
this can be done.
2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add method to find a stream by control url
2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add method to check control url of stream
2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
session: use path matching for session media
Use a path string instead of a uri to lookup session media in the sessions. Also
use path matching to find the largest possible path that matches.
2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-mount-points.h:
* tests/check/gst/mountpoints.c:
mount-points: remove useless vmethod
Making lookups in the mount points should not be done with a URL, if there is a
mapping to be done from URL to mount points, we'll need to do it somewhere
else.
2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-mount-points.h:
* tests/check/gst/mountpoints.c:
mount-points: improve mount point searching
Use a GSequence to keep track of the mount points.
Match a URL to the longest matching registered mount point. This should be the
URL to perform aggreagate control and the remainder is the stream specific
control part.
Add some unit tests for this.
2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
* gst/rtsp-server/Makefile.am:
rtsp-server: Allow building of static library
2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/mediafactory.c:
tests: fix compilation
2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-sdp.c:
sdp: get control string from stream
Use the control string as configured in the stream.
2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add methods and property to set control string
2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: cleanups
Rename variables for clarity
Keep media in state when we can
2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add more support for IPv6
Rename _get_address to _get_multicast_address in GstRTSPStream to
make it clear that this function only deals with multicast.
Make it possible to have both an IPv4 and IPv6 multicast address on
a stream. Give the client an IPv4 or IPv6 address depending on the
address it used to connect to the server.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: fix comment
2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: handle failed port allocation
Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
can't allocate any family at all. Also keep track of what port families we
allocated.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: improve docs
2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream-transport.c:
stream-transport: remove old if 0 block
2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
* tests/check/gst/client.c:
tests: fix tests
gst_rtsp_client_get_uri() has been removed
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add method to filter managed sessions
Add a method to filter the sessions managed by this client connection.
See https://bugzilla.gnome.org/show_bug.cgi?id=703016
2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: remove _get_uri() method
Remove the get_uri() method on the client. A client has no uri, the uri
property is an internal property to manage the last cached media for
the client.
2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: fix typo
2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Do not leak the query in default_query_stop
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: don't unlock when conversion fails
Don't unlock the state lock when conversion fails because it was not locked.
2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Add query_position and query_stop vmethods to rtsp-media
2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
Fix typo in property install for rtsp-media's time-provider
2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: clean some variables
Clean some variables and add some guards to _send_request()
2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Add gst_rtsp_client_send_request API
This makes it possible to send arbitrary messages to a client, such as
SET_PARAMETER or GET_PARAMETER
2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add _get_element() method
Add method to get the element used when creating the media.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: fix docs
2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: allow access to the rtp session
https://bugzilla.gnome.org/show_bug.cgi?id=703004
2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
dscp qos support in gst-rtsp-stream
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/rtspserver.c:
tests: fix test
Actually do what the comment says. Also keep the old code around, not sure what
should happen when you get a 454 from a TEARDOWN, does it close the connection?
it currently doesn't.
2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: also watch newly created session
When we newly created a session, start watching it immediately instead of
on the next request.
2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
* tests/check/gst/client.c:
tests: add unit test for new-session
See https://bugzilla.gnome.org/show_bug.cgi?id=701587
2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: emit new-session when new session is created
Only emit new-session when we created a new session for a client, not when a
client picked up a previous session.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: handle asterisk as path in requests
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: handle segment query format mismatch
It's possible that the segment query returns with a different format than what
we asked for, handle this case also.
2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-media.c:
media: use segment stop in collect_media_stats
Use segment stop instead of duration as range end point.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
rtsp-media: Do not leak the element in take_pipeline
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: Make configure_client_transport virtual
This patch makes configure_client_transport virtual. The functionality is
needed to handle some weird clients sending multicast transport settings as url
options.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: Make param_set and param_get virtual
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: convert_range replaces get_range_times
get_range_times worked for handling UTC ranges for seeks, but we also
need to convert back from NPT to the requested unit in
get_range_string. convert_range is now used for both.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
sdp: cleanup sdp info
We don't need to pass the proto, we can more easily check a boolean.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
use 0.0.0.0 or :: for c= line instead of server address
2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
* gst/rtsp-server/rtsp-client.c:
use local address, not remote, in SDP
See https://bugzilla.gnome.org/show_bug.cgi?id=702063
2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 098c0d7 to 01a7a46
2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: possibility to override range time conversion
Make it possible to override the conversion from GstRTSPTimeRange to
GstClockTimes, that is done before seeking on the media
pipeline. Overriding can be useful for UTC ranges, where the default
conversion gives nanoseconds since 1900.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
rtsp-server: Expose the use_client_settings API
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtspstream: handle both ipv4 and ipv6 clients
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-sdp.c:
Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
We already have a way to place extra attributes in the SDP by using a string
property with prefix x- or a- in the caps.
2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-sdp.c:
Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
We already have a way to place extra attributes in the SDP, just make a string
property in the payloader with a- or x- prefix.
2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-sdp.c:
rtsp: place a- and x- properties as attributes
application/x-rtp has properties with a- and x- prefixes that should be
placed as attributes in the SDP for the media instead of being added to the
fmtp.
2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
* examples/test-video.c:
example: add TLS example
2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: add support for TLS
Add methods to set and get a TLS certificate.
Add vmethod to configure a new connection. By default, configure the TLS
certificate in a new connection if needed.
2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: remove accept_client vmethod
This vmethod is not very useful so remove it.
2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: don't crash on NULL GError
2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-session-pool.c:
rtsp-session-pool: corrected session timeout detection
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: improve debug
2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server.c:
server: refactor connection setup
Let the server accept the socket connection and construct a GstRTSPConnection
from it. Remove the code from the client and let the client only deal with
a fully configure GstRTSPConnection object.
We will need this later when the server will configure the connection for
TLS.
2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: keep the transport object alive
Keep the transport object alive while we have it as qdata on the
source.
2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server.c:
rtsp-server: Do not crash on nmapping of server
* generate error when gst_rtsp_connection_accept fails
* do not stop accepting incoming connections because
accepting a client fails
https://bugzilla.gnome.org/show_bug.cgi?id=701072
2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
https://bugzilla.gnome.org/show_bug.cgi?id=700953
2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Parse framerate caps field and set SDP attribute
The SDP attribute and its format is described in RFC4566.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Parse width/height from caps and set SDP attribute
The SDP attribute and its format is described in RFC6064.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-sdp.c:
* tests/check/gst/client.c:
rtsp-sdp: add bandwidth line
https://bugzilla.gnome.org/show_bug.cgi?id=699220
2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 5edcd85 to 098c0d7
2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
* tests/check/gst/media.c:
tests: add dynamic payloader prepare/unprepare check
2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: release lock when removing fakesink
2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: set elements to NULL before removing
When removing a stream, set the elements to NULL first. This avoids
element-is-not-in-NULL-state errors when we dispose the elements.
2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 3cb3d3c to 5edcd85
2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: listen to pad-removed signals
Listen to the pad-removed signal and remove the stream associated with the
removed pad.
Add signal to be notified of the removed pad.
Remove the fakesink in unprepare()
Fix signatures of the signal methods
2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-sdp.c:
tests: add example of reusable pipelines
2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add method to get the srcpad
2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
* tests/check/gst/media.c:
check: add media prepare/unprepare test
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
media: disconnect from signal handlers in unprepare()
We connected to the pad-added and no-more-pads signals in prepare() so
we need to disconnect from them in unprepare().
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
media: don't free streams array
Don't free the streams array in the unprepare() method, they were not
added in prepare().
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
media: don't unref the pipeline in unprepare
Unprepare() should undo what prepare() does. Because the pipeline is
not created in prepare(), we should not unref it in unprepare()
2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-stream.c:
stream: clear session and caps for reuse
Set the session and caps to NULL after unref otherwise we might unref
them again later.
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
client: send out teardown signal before tearing down
The advantage is that in the signal handler you get direct access to
information about what streams are about to get torn down (in the
GstRTSPClientState).
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: expose connection
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From aed87ae to 3cb3d3c
2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
media: add method to get the base_time of the pipeline
Together with a shared clock, this base-time could eventually be sent to
the client so that it can reconstruct the exact running-time of the clock
on the server.
2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
media: add GstNetTimeProvider support
Add a property to let the media provide a GstNetTimeProvider for its clock.
Make methods to get the clock and nettimeprovider
Add a x-gst-clock property to the SDP with the IP and port number of the nettime
provider and also the current time of the clock. This should make it possible
for (GStreamer) clients to slave their clock to the server clock.
2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 04c7a1e to aed87ae
2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: wait for buffering to complete
Wait for buffering to complete before changing the state to the target state.
2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: small cleanup
2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
* tests/check/gst/rtspserver.c:
tests: remove extra unref in test_setup_non_existing_stream
The unref is not needed anymore, teardown runs without it.
https://bugzilla.gnome.org/show_bug.cgi?id=696542
2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
* tests/check/gst/rtspserver.c:
tests: GSocketService cleanup in test_bind_already_in_use
Use g_socket_service_stop so the rtspserver test stops listening for
incoming connections in test_bind_already_in_use.
https://bugzilla.gnome.org/show_bug.cgi?id=696541
2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
Instead use a GWeakRef which is safe to use
This is a known GLib bug, see:
https://bugzilla.gnome.org/show_bug.cgi?id=667145
2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* tests/check/gst/media.c:
* tests/check/gst/rtspserver.c:
rtsp-media/client: Reply to PLAY request with same type of Range
Remember the type of Range from the PLAY request and use the same type for
the reply.
2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* tests/check/gst/client.c:
rtsp-client: expose uri
2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/mediafactory.c:
tests: Hold ref while creating second media
To test if the media aren't shared, make sure we keep the first one while creating a second
otherwise the same memory address may be reused.
2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
* configure.ac:
configure: remove out-of-date comment
2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
* .gitignore:
.gitignore: ignore more build files
2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
* tests/check/Makefile.am:
tests: use right _LIBS variable for gst-plugins-base libs
2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/Makefile.am:
check: add librtp to libs
2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/rtspserver.c:
tests: Add test to check selecting a port the server will send from
2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/rtspserver.c:
tests: Make sure packets are actually received
2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Select unicast address from pool if appropriate
2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-stream.c:
stream: Properties are always there in Gst 1.0
2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/addresspool.c:
tests: Add tests for unicast addresses in pool
2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-address-pool.c:
* tests/check/gst/addresspool.c:
address-pool: Verify that multicast addresses are used for multicast and vice-versa
2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/addresspool.c:
address-pool: Add unicast addresses
2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
* configure.ac:
* gst/rtsp-server/rtsp-server.c:
* tests/check/gst/rtspserver.c:
rtsp-server: Limit the number of threads per server instance
If we exceed the maximum, just round robin the clients over the existing
threads.
2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-server.c:
rtsp-server: No need to store the GMainContext in the client context
2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/rtspserver.c:
tests: Add test for client disconnection
2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/rtspserver.c:
tests: Test client and session timeouts with multiple threads
2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
Document locking and its order
2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/rtspserver.c:
tests: Test that slow DESCRIBE don't block other clients
2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/client.c:
tests: Add tests for client-requested multicast address
2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
* docs/libs/gst-rtsp-server-sections.txt:
docs: Put the various functions in the right sections
2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
docs: Generate docs for GstRTSPAddressPool
2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
client: Check client provided addresses against the address pool
2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
* tests/check/gst/addresspool.c:
address-pool: Add API to request a specific address from the pool
Also add relevant unit tests.
2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/mediafactory.c:
tests: Check the passing around of a RTSPAddressPool
Make sure the RTSPAddressPool is propagated from the MediaFactory all the
way down to the stream.
2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
* tests/check/gst/addresspool.c:
tests: Add more tests for the address pool
2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-address-pool.c:
address-pool: Fix off by one error
When splitting a port range, the port after a skip is not part of range.
2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 2de221c to 04c7a1e
2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
* configure.ac:
configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
AM_CONFIG_HEADER was removed in automake 1.13
https://bugzilla.gnome.org/show_bug.cgi?id=693368
2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From a942293 to 2de221c
2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: make sure the watch exists while sending data
Protect the send_func with a lock. This allows us to wait for sending
to complete before changing the send_func and user_data. We add an
extra ref to the watch to make sure that it remains valid during
sending.
When closing the connection, set the send_func to NULL
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* tests/check/Makefile.am:
tests: use GST_*_1_0 environment variables everywhere
The _1_0 suffixed environment variables override the
non-suffixed ones, so if we're in an environment that
sets the _1_0 suffixed ones, such as jhbuild, we need
to set those to make sure ours actually always get
used.
2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From acb04d9 to a942293
2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: set the client backlog
Set the client backlog to a reasonable default
2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Make the element a constructor parameter
https://bugzilla.gnome.org/show_bug.cgi?id=689594
2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* docs/libs/Makefile.am:
docs: Link with gcov library when gcov is enabled
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: match prepare with unprepare
Really unprepare when there were an equal amount of prepare calls.
2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: media has to be unprepared in finalize
Because unprepare takes away the last ref on the media.
2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
We can't use the refcount to trigger unprepare because it is the unprepare call
that removes the last refcount after all messages are consumed. What we should
probably do is make a prepared refcount and only unprepare when the refcount
reaches 0.
2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: let the source unref the last media ref
the last ref to the media is held by the source so we don't need to add more ref
and unrefs, we simply destroy the media when the source is gone.
2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: improve debug
2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: check state
Make sure we are in the right state when collecting the position and duration.
Only make ourselves PREPARED when we were previously PREPARING.
2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: use g_object_ref/unref for GObjects
2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-client.c:
client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
isn't being used anymore.
2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-media.c:
Fix compiler warning
2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-media-factory-uri.c:
Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-media.h:
small cleanup
2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/media.c:
media: avoid element leak
2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: require an element in media constructor
2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Revert "client: TEARDOWN brings that state to Init again"
This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
The object is already disposed, there is no point in setting the state.
2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: TEARDOWN brings that state to Init again
2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* examples/test-auth.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/media.c:
rtsp: make object details private
Make all object details private
Add methods to access private bits
2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/gst/media.c:
tests: add media tests
2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: check if prepared for some methods
Check that the media object is prepared before doing seek and getting the
current position etc.
Add some g_return checks.
2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/gst/mediafactory.c:
tests: add mediafactory test
2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: improve debug
2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: unref pipeline in finalize to avoid leaking it
2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media.c:
rtsp: use gst_object_unref on GstObjects
2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: require an url
2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-uri.c:
examples: fix include
2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.h:
server: remove unused include
2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/gst/mountpoints.c:
tests: add test for mountpoints
2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: fix factory leak
Keep the factory in the state object only for authorization checks and make
sure we unref it on failure. Also don't keep invalid objects in the state
object.
2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-mount-points.c:
mounts: add g_return_if guards
2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/client.c:
tests: add more tests
2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: improve debug
2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: improve debug and fix leaks
Cleanup the uri and session when there is a bad request.
2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* common:
update common
2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/gst/client.c:
test: add test for session in options request
2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use 454 when session can't be found
We should use 454 when a session can't be found because there was no session
pool configured in the server. This is not a server configuration problem
because the server on which the request is done might not be the same one that
will keep the sessions for us and so it does not need to support sessions.
2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: only free connection when there is one
It's possible that the client doesn't have a connection when we try to free it.
2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/gst/client.c:
tests: add unit test for the client object
2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: small cleanup
2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.h:
client: remove unused include
2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: fix compilation
2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: call destroy without the lock
2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: make the client usable without a socket
Make a method to let the client handle a message and a callback when the client
wants us to send a response message back. This makes it possible to also use the
client object without the sockets, which should make it easier to test.
2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: small cleanup
2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server.c:
client: remove reference to server
We don't need to keep a ref to the server
2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add locking
Also add some g_return_if()
2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: log more errors
2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: fix compilation
2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add generic close-after-send support
Add a property to send_response() to close the connection after the response has
been sent to the client.
2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/README:
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* docs/libs/gst-rtsp-server.types:
* examples/test-auth.c:
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-multicast.c:
* examples/test-multicast2.c:
* examples/test-ogg.c:
* examples/test-readme.c:
* examples/test-sdp.c:
* examples/test-uri.c:
* examples/test-video.c:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media-mapping.h:
* gst/rtsp-server/rtsp-mount-points.c:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
* tests/check/gst/rtspserver.c:
MediaMapping -> MountPoints
Describes better what the object manages.
2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
configure: bump required version of -base
2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: fix seeking
2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: support more Range formats
Use the new -base methods to convert the Range string into a seek start and stop
value.
2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-launch.c:
examples: fix whitespace
2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
test-auth: add example of how to remove sessions
Add an example of the session filter api.
2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-uri.c:
test-uri: remove mapping example
2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-uri.c:
test-uri: fix callback signature
2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
factory: keep ref to factory while media active
While the media from a factory is alive, keep a ref to the factory.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
factory-uri: add some debug
2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: set udp sources to PLAYING
Set the UDP sources to PLAYING and locked state before we add it to the pipeline
so that it doesn't cause our pipeline to produce ASYNC-DONE.
2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
factory-uri: take ref to factory
Take a ref to the factory that we place in our list.
2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/Makefile.am:
* tests/test-reuse.c:
test: add test for server reuse
See https://bugzilla.gnome.org/show_bug.cgi?id=688395
2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-server.c:
server: start and stop multiple times
Stop listening on the RTSP port when the GSource is removed, so clients
can't connect and the server can be started again.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: fix small leak
2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: unref source in finish_unprepare
The source is created in prepare, unref it in finish_unprepare.
See https://bugzilla.gnome.org/show_bug.cgi?id=688707
2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
rtsp-media: remove bus watch before finalizing
* A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
* An extra media ref is added for the bus watch. This extra ref is unreffed by
the GDestroyNotify function.
* gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
* GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
gst_rtsp_media_unprepare before unreffing the media.
This way, the bus watch will be removed before the media is finalized.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: wait until the TEARDOWN response is sent to close the connection
Responses can be sent async so we need to wait until the TEARDOWN response has
been written before we close the connection to the client. This avoids the risk
of writing/polling closed sockets.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: plug socket leak
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 6bb6951 to a72faea
2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
* gst/rtsp-server/rtsp-media-factory-uri.c:
rtsp-server: don't use deprecated API
2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: fix unused-but-set-variable compiler warning
rtsp-client.c:1260:21: error: variable 'protocols' set but not used
2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* TODO:
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-client.c:
rtsp: cleanups
2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
* examples/test-multicast2.c:
examples: add another multicast example
Add an example for how to configure separate multicast ranges for each media
stream.
2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-multicast.c:
test: set shared
2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
stream: use the address managed by the stream
Use the address managed by the stream for multicast. This allows us to have 1
multicast address for each stream.
Because the address is now managed by the stream we don't have to pass it around
anymore.
Set the address pool on the streams.
2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
rtsp: improve debug
2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add signal for new streams
This allows applications to listen for new streams and configure properties on
them, like the address pool.
2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: configure address pool in new streams
2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add methods to deal with address pool
Add methods to get and set the address pool for the stream
Add method to allocate and get the multicast addresses for this stream.
2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: remove MTU property
It is a stream property
2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: set blocksize only on stream
Set the blocksize only on the current stream.
2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: share src and sink sockets
the allocated socket is in the used-socket property, not socket.
2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* tests/check/gst/addresspool.c:
rtsp: make address-pool return an address object
Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
store more info in the structure and allows us to more easily return the address
to the right pool when no longer needed.
Pass the address to the StreamTransport so that we can return it to the pool
when the stream transport is freed or changed.
2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
* examples/test-multicast.c:
examples: add multicast example
Show how to set up the multicast address pool so that media can be
server with multicast.
2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp: use AddressPool
Remove the multicast_group property.
Use the configured addresspool to allocate multicast addresses.
2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
address-pool: add clear method
2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-address-pool.c:
address-pool: small cleanups
2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/check/Makefile.am:
* tests/check/gst/addresspool.c:
tests: add addresspool unit test
2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-address-pool.c:
* gst/rtsp-server/rtsp-address-pool.h:
address-pool: add object to manage multicast addresses
Make an object that can manage a rage of multicast addresses and ports.
2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: set default max-threads property
2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: wait for concurrent _prepare
If a prepare is busy, wait for the result.
2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: add lock around message handler
We don't want to dispatch messages while we are still processing the result of
the state change.
2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add lock to protect state changes
2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: add locking
2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
stream-transport: add keep-alive method
2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
stream-transport: add method to handle RTP/RTCP
Call new methods instead of poking into the structures directly.
2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
session-media: add locking
2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
session: add locking
2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: free old socket
2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media-mapping.h:
mapping: add locking
2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: add locking
2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
auth: add locking
2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: add max-thread property
2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: use a threadpool for the mainloops
2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: rename method
gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
don't really create the client from the socket, we use the socket for the
client.
2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server.c:
server: rework maincontext handling in clients
Make a separate method to attach a client to a MainContext.
Let the server decide in what GMainContext the client will operate and give this
context to the client in attach. Then the server can later decide to use a
separate thread for each client or just use the mainthread.
2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
session: move session header code in session object
2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
* COPYING:
* COPYING.LIB:
* examples/test-auth.c:
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-readme.c:
* examples/test-sdp.c:
* examples/test-uri.c:
* examples/test-video.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media-mapping.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-params.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
* tests/check/gst/rtspserver.c:
* tests/test-cleanup.c:
Fix FSF address
2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session.c:
rtsp-server: added annotations to indicate type of ownership transfer of return values
https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
* configure.ac:
No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
* Makefile.am:
* bindings/Makefile.am:
* bindings/vala/Makefile.am:
* bindings/vala/gst-rtsp-server-0.10.deps:
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.deps:
* bindings/vala/packages/gst-rtsp-server-0.10.files:
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
* bindings/vala/packages/gst-rtsp-server-0.10.namespace:
* configure.ac:
bindings: remove vala bindings
They'll be reunited with the other GStreamer bindings
https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
rtsp: only create transport when needed
Only create the StreamTransport when configured.
2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: small cleanup
2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
rtsp: refactor configuration of transport
Move the configuration of the transport to a place where it makes
more sense.
2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: refactor transport parsing
2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: refuse to change the MTU on shared media
If we change the MTU of chared media, it changes for all clients.
We don't want to set the MTU to something large for clients that
stream over UDP.
2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-mp4.c:
* gst/rtsp-server/rtsp-media.c:
small fixes to docs and debug
2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-stream.c:
stream: transports must already have been removed
2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
stream: improve join and leave of the pipeline
simplify code
Do the cleanup properly
Add some docs
2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: move unprepare below default implementation
Makes it easier to find the default implementation
2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: signal unprepared when we actually finish
2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: no need to unlock, unprepare does that when needed
2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.h:
docs: update docs
2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-mapping.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp: fix MTU setting
Fix setting of the MTU. There is no need for a vmethod.
2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/README:
docs: update docs
2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump version number after refactoring
2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-session-media.c:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp: massive refactoring
Make GObjects from the remaining simple structures.
Remove GstRTSPSessionStream, it's not needed.
Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
a GstRTSPStream should be transported to a client.
Rename GstRTSPMediaFactory::get_element -> create_element because that
more accurately describes what it does.
Make nice methods instead of poking in the structures.
Move some methods inside the relevant object source code.
Use GPtrArray to store objects instead of plain arrays, it is more
natural and allows us to more easily clean up.
Move the allocation of udp ports to the Stream object. The Stream object
contains the elements needed to stream the media to a client.
Improve the prepare and unprepare methods. Unprepare should now undo
everything prepare did. Improve also async unprepare when doing EOS on
shutdown. Make sure we always unprepare correctly.
2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Unref server address clients connected to
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-server.c:
rtsp-server: don't ref server socket if it is NULL
Fixes test_bind_already_in_use unit test again after commit 6a497440.
https://bugzilla.gnome.org/show_bug.cgi?id=686644
2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
* tests/check/Makefile.am:
tests: Add libgio link dependency
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media-mapping.h:
rtsp-media-mapping: rename find_media vfunc to find_factory
The virtual method and class method should have the same name
so it is correctly represented in GIR file
https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
rtsp-server: fixed comments and GIR annotations
https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-media-mapping.c:
media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-server.c:
rtsp-server: allow binding on port 0 (binds on a random port)
2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
rtsp-server: add bound-port property
bound-port can be used to retrieve the port number when the server is bound on
port 0, which binds on a random port.
2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
rtsp-media-factory: make ::get_element overridable by GI bindings
The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
as the invoker for ::get_element(), making it overridable by GI generated
bindings.
2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-media-factory-uri.c:
rtsp-media-factory-uri: don't autoplug parsers in a loop
Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
h264parse forever.
2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/Makefile.am:
Explicitly link against gio. Fix link error on mac.
2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
* gst/rtsp-server/rtsp-session.c:
session: add ttl to the transport header in SETUP
See https://bugzilla.gnome.org/show_bug.cgi?id=685561
2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media.c:
client: Use client transport settings for multicast if allowed.
This patch makes it possible for the client to send transport settings for
multicast (destination && ttl). Client settings must be explicitly allowed or
the server will use its own settings.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 6c0b52c to 6bb6951
2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: do not destroy the rtsp watch
Don't destroy the client watch while dispatching. The rtsp watch is
automatically destroyed after the rtsp watch function closed() has
been called.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 4f962f7 to 6c0b52c
2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-media.c:
media: fix check for seekability
2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use more GIO
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: remove obsolete includes
2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
* gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
be available in "on_new_ssrc". The transports are added in
gst_rtsp_media_set_state when going to PLAYING state. However,
"on_new_ssrc" might be called before this happens.
https://bugzilla.gnome.org/show_bug.cgi?id=683304
2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: add signals for rtsp requests (fixes #683287)
2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
add new-session signal to rtsp-client (fixes #683058)
2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 668acee to 4f962f7
2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-server.c:
* tests/check/gst/rtspserver.c:
rtsp-server: fixed segfault in gst_rtsp_server_create_socket
Do not assume that *error is set in g_socket_address_enumerator_next.
Added test_bind_already_in_use unit-test.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
* common:
Automatic update of common submodule
From 94ccf4c to 668acee
2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
rtsp-client: make create_sdp virtual method
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 98e386f to 94ccf4c
2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: fix docs
2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
rtsp-server: use an existing socket to establish HTTP tunnel
Make it possible to transfer a socket from an HTTP server to be used as
an RTSP over HTTP tunnel.
2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp: Handle the blocksize parameter
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
* tests/check/Makefile.am:
* tests/check/gst/rtspserver.c:
Have unit test get header from source dir, not installed dir
This makes compilation of unit tests work in a build directory other
than the source directory.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: update for gst_element_make_from_uri() changes
2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
* configure.ac:
* tests/Makefile.am:
* tests/check/Makefile.am:
* tests/check/gst/rtspserver.c:
rtsp: add unit test
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: don't collect media stats when going to NULL
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: don't leak transports
2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: free transport on no_stream in SETUP handler
2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: changed session media iteration
In client_unlink_session: now don't iterate in session->medias
list where items are removed by gst_rtsp_session_release_media.
Instead, repeatedly remove the first item.
2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
GstRTSPSessionMedia is not a GObject type. When the
GstRTSPSession is freed, it will free the media.
2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-media-factory.c:
factory: plug pad leak in collect_streams
In gst_rtsp_media_factory_collect_streams: unref the srcpad that
was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
will take one reference, and the other reference will otherwise
give a memory leak.
2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
* configure.ac:
configure: suppress some warnings when debug is disabled
Warnings about unused variables should be suppressed if core has the
debug system disabled.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/Makefile.am:
docs: fix build in uninstalled setup
Include gst-plugins-base libs properly.
2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
* docs/libs/gst-rtsp-server.types:
docs: include headers defining rtsp-server object types
Fixes compiler warnings during docs build.
https://bugzilla.gnome.org/show_bug.cgi?id=676824
2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
* configure.ac:
configure: Add warning flags for compiler when configuring
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* common:
Automatic update of common submodule
From 03a0e57 to 98e386f
2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* common:
Automatic update of common submodule
From 1fab359 to 03a0e57
2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
* gst/rtsp-server/rtsp-client.c:
client: fix GSocketAddress leak in gst_rtsp_client_accept
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* common:
Automatic update of common submodule
From f1b5a96 to 1fab359
2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 92b7266 to f1b5a96
2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From ec1c4a8 to 92b7266
2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 3429ba6 to ec1c4a8
2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-server.c:
rtsp: fix compiler warnings
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From dc70203 to 3429ba6
2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
rtsp-server: port to new thread API
2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 6db25be to dc70203
2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
rtsp-server: Fix compilation and compiler warnings
2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* autogen.sh:
* configure.ac:
* gst/rtsp-server/Makefile.am:
configure: Modernize autotools setup a bit
Also we now only create tar.bz2 and tar.xz tarballs.
2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 464fe15 to 6db25be
2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 7fda524 to 464fe15
2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
* docs/libs/Makefile.am:
* docs/version.entities.in:
* gst-rtsp.spec.in:
* gst/rtsp-server/Makefile.am:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp-server.pc.in:
* tests/Makefile.am:
rtsp-server: Update versioning
2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
Merge remote-tracking branch 'origin/0.10'
Conflicts:
gst/rtsp-server/rtsp-session-pool.c
2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/rtsp-server/rtsp-session-pool.c:
rtsp-server: Don't use deprecated GLib API
2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Replace master with 0.11
2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* docs/README:
A couple minor typo fixes
2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: fix state of the appqueue
2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
factory: use videoconvert
2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
factory: change to new style caps
2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-pool.c:
rtsp-server: port to GIO
Port to GIO
2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
configure: fix build
2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/README:
docs: fix for gst_rtsp_server_set_port() -> _set_service()
https://bugzilla.gnome.org/show_bug.cgi?id=666548
2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* examples/Makefile.am:
First rule of gst-rtsp-server club: don't talk about gst-phonon
2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
* pkgconfig/gstreamer-rtsp-server.pc.in:
pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
For consistency with all other modules.
2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: update for new map API
2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
* bindings/Makefile.am:
* bindings/python/Makefile.am:
* bindings/python/arg-types.py:
* bindings/python/codegen/Makefile.am:
* bindings/python/codegen/__init__.py:
* bindings/python/codegen/argtypes.py:
* bindings/python/codegen/code-coverage.py:
* bindings/python/codegen/codegen.py:
* bindings/python/codegen/definitions.py:
* bindings/python/codegen/defsparser.py:
* bindings/python/codegen/docextract.py:
* bindings/python/codegen/docgen.py:
* bindings/python/codegen/fileprefix.override:
* bindings/python/codegen/fileprefixmodule.c:
* bindings/python/codegen/h2def.py:
* bindings/python/codegen/mergedefs.py:
* bindings/python/codegen/mkskel.py:
* bindings/python/codegen/override.py:
* bindings/python/codegen/reversewrapper.py:
* bindings/python/codegen/scmexpr.py:
* bindings/python/rtspserver-types.defs:
* bindings/python/rtspserver.defs:
* bindings/python/rtspserver.override:
* bindings/python/rtspservermodule.c:
* bindings/python/test.py:
* configure.ac:
python: remove pygst-based python bindings
pygi is the future, apparently.
2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
* common:
Automatic update of common submodule
From c463bc0 to 7fda524
2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 2a59016 to c463bc0
2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 0807187 to 2a59016
2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 11f0cd5 to 0807187
2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-auth.c:
example: update for new caps
2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-video.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp-server: port some more to 0.11
Fix caps.
Remove bufferlist stuff
Update for new API.
Add queue before appsink now that preroll-queue-len is gone.
Update for request pad changes.
2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add a seekable boolean
Maintain the seekable state with a new variable instead of reusing the
is_live variable.
2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
* gst/rtsp-server/rtsp-media.c:
Disallow seek in live media
2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
* gst/rtsp-server/rtsp-server.c:
#ifdef statements for windows socket creation were missing
2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From a39eb83 to 11f0cd5
2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 605cd9a to a39eb83
2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use method to access property
2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add protocols property
Add a property to configure the allowed protocols in the media created from the
factory.
2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add media-configure signal
Add signal to allow the application to configure the media after it was created
from the factory.
2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use method to access property
2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add protocols property
Add a property to configure the allowed protocols in the media created from the
factory.
2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add media-configure signal
Add signal to allow the application to configure the media after it was created
from the factory.
2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use media multicast group
2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
retab some .h
2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.h:
sdp: copy and free the server ip address
Copy and free the server ip address to make memory management easier later.
2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: configure multicast in media
2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add property for multicast group
Add a property to configure the multicast group in the media.
Based on patches from Marc Leeman and Robert Krakora.
2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add property for multicast group
Add a property to configure the multicast group in the media factory.
Based on patches from Marc Leeman and Robert Krakora.
2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: do configuration of transport in one place
Move the configuration of the transport destination address to where we also
configure the other bits.
2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use media multicast group
2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
retab some .h
2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-sdp.h:
sdp: copy and free the server ip address
Copy and free the server ip address to make memory management easier later.
2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: configure multicast in media
2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add property for multicast group
Add a property to configure the multicast group in the media.
Based on patches from Marc Leeman and Robert Krakora.
2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add property for multicast group
Add a property to configure the multicast group in the media factory.
Based on patches from Marc Leeman and Robert Krakora.
2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: do configuration of transport in one place
Move the configuration of the transport destination address to where we also
configure the other bits.
2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
* gst/rtsp-server/rtsp-client.c:
client: destroy pipeline on client disconnect with no prior TEARDOWN.
The problem occurs when the client abruptly closes the connection without
issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
server is where the pipeline gets torn down. Since this handler is not called,
the pipeline remains and is up and running. Subsequent clients get their own
pipelines and if the do not issue TEARDOWNs then those pipelines will also
remain up and running. This is a resource leak.
2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
For example, it can be used to retrieve source elements like appsrc, in a more
convenient way than subclassing get_element.
2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
* gst/rtsp-server/rtsp-server.c:
rtsp-server: hold on to reference while using object
2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: use new api
2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
configure: use unstable api
2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
* gst/rtsp-server/rtsp-client.c:
client: fix reference counting
2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
fix compiler warnings about unused variables
2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
* examples/test-launch.c:
* examples/test-readme.c:
* examples/test-uri.c:
* examples/test-video.c:
examples: tell rtsp uri when ready
2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
* common:
Automatic update of common submodule
From 69b981f to 605cd9a
2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: update for buffer API change
2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
Makefile.am: 0.10 => @GST_MAJORMINOR@
2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/rtsp-server/.gitignore:
.gitignore: 0.10 => 0.11
2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
Makefile.am: 0.10 => @GST_MAJORMINOR@
2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 9e5bbd5 to 69b981f
2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From fd35073 to 9e5bbd5
2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 46dfcea to fd35073
2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media.c:
media: port to new caps API
2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
* bindings/vala/gst-rtsp-server-0.10.vapi:
Updated Vala bindings.
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
Add a signal for newly connected clients.
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
* bindings/python/rtspserver.override:
python: override gst_rtsp_media_mapping_add_factory to fix refcounting
2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-funnel.c:
* gst/rtsp-server/rtsp-funnel.h:
* gst/rtsp-server/rtsp-media.c:
rtsp-server: port to 0.11
2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* common:
add common
2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
Conflicts:
common
configure.ac
2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From c3cafe1 to 46dfcea
2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
* bindings/python/Makefile.am:
* bindings/python/rtspserver.defs:
python bindings: wrap GstRTSPMediaFactoryClass vfuncs
2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
* bindings/python/arg-types.py:
python bindings: add GstRTSPUrlParam
Needed to implement MediaFactory virtual proxies
2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
* bindings/python/arg-types.py:
python bindings: fix returning GstRTSPUrl types
2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
* bindings/python/arg-types.py:
python bindings: add arg type for GstRTSPUrl
2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
* bindings/python/rtspserver.defs:
python bindings: fix the definition of MediaFactory.collect_stream
2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 1ccbe09 to c3cafe1
2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 193b717 to 1ccbe09
2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From b77e2bf to 193b717
2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* Makefile.am:
build: Include lcov.mak to allow test coverage report generation
2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From d8814b6 to b77e2bf
2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* common:
Automatic update of common submodule
From 6aaa286 to d8814b6
2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 6aec6b9 to 6aaa286
2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
* autogen.sh:
autogen: wingo signed comment
2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
* gst/rtsp-server/rtsp-session-pool.c:
session: use full charset for RTSP session ID
As specified in RFC 2326 section 3.4 use full valid charset to make guessing
session ID more difficult.
https://bugzilla.gnome.org/show_bug.cgi?id=643812
2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
rtsp-server: Don't install the funnel header
2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* common:
Automatic update of common submodule
From 1de7f6a to 6aec6b9
2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: require core/base 0.10.31
Needed at least for gst_plugin_feature_rank_compare_func().
2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From f94d739 to 1de7f6a
2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: remove more unused code
2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: remove duplicate filtering
Remove the duplicate filtering code now that we have a released -good version.
Give a warning instead.
2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
media: fix default buffer size
2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add property to configure the buffer-size
Add a property to configure the kernel UDP buffer size.
2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add property to configure kernel buffer sizes
Add a property to configure the kernel UDP buffer size.
2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: set PYGOBJECT_REQ before using it
https://bugzilla.gnome.org/show_bug.cgi?id=640641
2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/Makefile.am:
docs: recursive into sub-directories on 'make upload'
2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/version.entities.in:
docs: mention full version these docs are for, not just major-minor
2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
back to development
=== release 0.10.8 ===
2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
release 0.10.8
2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
rtsp-server: clarify docs a little
2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: init debug category before starting thread
2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
auth: add realm to make it more spec compliant
2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: add locking
2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-video.c:
example: improve example docs a little
2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: ensure the watch has a ref to the server
2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: simpify channel function
2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: simplify management of channel and source
We don't need to keep around the channel and source objects. Let the mainloop
and the source manage the source and channel respectively.
2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* Makefile.am:
* configure.ac:
build tests
2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* tests/.gitignore:
* tests/Makefile.am:
* tests/test-cleanup.c:
tests: add tests directory and cleanup test
2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
server: improve debugging in various objects
2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: chain up to the parent finalize
2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
* bindings/python/rtspserver-types.defs:
* bindings/python/rtspserver.defs:
* bindings/python/rtspserver.override:
* bindings/python/test.py:
gst-rtsp-server: update python bindings
2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use the response from the clientstate
Create the response object only once and store in the client state.
Make all methods use the state response,
2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: use signal to keep track of clients
Keep track of all the clients that the server creates and remove them when they
fire the 'closed' signal.
2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: emit signal when closing
2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-auth.c:
* examples/test-video.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
media: enable per factory authorisations
Allow for adding a GstRTSPAuth on the factory and media level and check
permissions when accessing the factory.
Add hints to the auth methods for future more fine grained authorisation.
Add example application for per factory authentication.
2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-params.h:
rtsp-server: Pass ClientState structure arround
Pass the collected information for the ongoing request in a GstRTSPClientState
structure that we can then pass around to simplify the method arguments. This
will also be handy when we implement logging functionality.
2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: add methods to configure authorisation
2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: unref auth in finalize
2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: unref auth in finalize
2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* docs/libs/gst-rtsp-server.types:
docs: add more docs
2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: separate create and accept
Create separate create and accept methods so that subclasses can create custom
client object.
Configure the server in the client object and prepare for keeping track of
connected clients.
2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: add support for setting the server.
Add support for keeping a ref to the server that started this client
connection.
2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
auth: fix memleak and add some docs
Fix a memleak of the basic auth token.
Add docs for the helper function
2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
client: delegate setup of auth to the manager
Delegate the configuration of the authentication tokens to the manager object
when configured.
2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-video.c:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
auth: add authentication object
Add an object that can check the authorization of requests.
Implement basic authentication.
Add example authentication to test-video
2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: move includes back
the includes are needed for sockaddr_in.
2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
rtsp: move network includes where they are needed
2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
* gst/rtsp-server/rtsp-media.h:
rtsp-media.h: Minor corrections in comments.
Fixes #638944
2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From e572c87 to f94d739
2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* .gitignore:
* docs/.gitignore:
* docs/libs/.gitignore:
* examples/.gitignore:
* gst/rtsp-server/.gitignore:
gitignore: updates
2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* docs/libs/Makefile.am:
docs: We don't build ps/pdf for API reference docs
2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From ccbaa85 to e572c87
2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* common:
Automatic update of common submodule
From 46445ad to ccbaa85
2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-funnel.c:
* gst/rtsp-server/rtsp-funnel.h:
* gst/rtsp-server/rtsp-media.c:
funnel: rename fsfunnel to rtspfunnel
Rename the funnel to avoid conflicts with the farsight one.
2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/fs-funnel.c:
* gst/rtsp-server/fs-funnel.h:
* gst/rtsp-server/rtsp-media.c:
rtsp-media: add and use fsfunnel
Add a copy of fsfunnel to the build because input-selector removed the (broken)
select-all property that we need.
2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
gobject-introspection: use PKG_CONFIG_PATH specified at configure time
Use PKG_CONFIG_PATH specified at configure time (if any) as well
for the g-ir-compiler, rather than just assuming the env var has
been set.
2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* .gitignore:
* Makefile.am:
* configure.ac:
* m4/Makefile.am:
* m4/codeset.m4:
build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
* gst/rtsp-server/Makefile.am:
gobject-introspection: fix g-i build for uninstalled setup
Requires gst-plugins-base git (> 0.10.31.2).
2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-uri.c:
examples: add some more options and comments
2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
factory-uri: use right property type
2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
factory-uri: attempt to configure buffer-lists
Attempt to configure buffer lists in the payloader for improved performance.
2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: attempt to configure bigger UDP buffers
Attempt to configure bigger udp kernel send buffers to avoid overflowing the
send buffers with high bitrate streams.
2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
* gst/rtsp-server/rtsp-client.c:
client: use the socket length from getsockname
Use the length returned by getsockname to perform the getnameinfo call because
the size can depend on the socket type and platform.
Fixes #638723
2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
docs: add uri factory to the docs
2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.h:
docs: improve docs
2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp-server: add support for buffer lists
Add support for sending bufferlists received from appsink.
Fixes #635832
2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
media: make method to retrieve the play range
Make a method to retrieve the playback range so that we can conditionally create
a different range for the SDP and the PLAY requests.
2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add signal to notify of state changes
2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.h:
client: cleanup headers
2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: fix typo
2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory-uri.h:
factory-uri: add support for gstpay
Add an option to prefer gstpay over decoder + raw payloader.
2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory-uri.h:
factory-uri: rework the autoplugger.
Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
before payloaders.
2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
factory-uri: use better factory filter
Make better payloader filter based on autoplug rank and RTP use case.
2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* common:
Automatic update of common submodule
From 169462a to 46445ad
2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: set SO_REUSEADDR before bind
Set the SO_REUSEADDR _before_ bind() to make it actually work.
2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: emit prepared signal when prepared
Make a 'prepared' signal and emit it when we successfully prepared the element.
This signal can be used to configure the media object after it has been prepared
for streaming.
2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
* common:
Automatic update of common submodule
From 011bcc8 to 169462a
2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
python an optional dependency
* configure.ac: Move up valgrind and g-i checks. Make the python
dependency optional, as it was before.
2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' into 0.11
Conflicts:
common
configure.ac
2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: update range when active clients changed
When we changed the number of active clients, update the current range
information because we want the second client connecting to a shared resource
continue from where the stream currently.
2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory-uri.h:
factory-uri: add colorspace and fix pt
Rework the way we pass data to the autoplugger.
When we have raw caps, plug a converter element to make pluggin to raw
payloaders more successful.
Make sure all dynamically plugged payloaders have a unique payload types.
2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
* examples/test-uri.c:
example: add example of the uri factory
2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-media-factory-uri.c:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-server.h:
factory-uri: add a factory to stream any URI
Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
when we have one.
2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: ignore spurious ASYNC_DONE messages
When we are dynamically adding pads, the addition of the udpsrc elements will
trigger an ASYNC_DONE. We have to ignore this because we only want to react to
the real ASYNC_DONE when everything is prerolled.
2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
media-factory: make lock macro
2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-server: Remove unused variable and dead assignment
2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-readme.c:
* examples/test-sdp.c:
* examples/test-video.c:
examples: Run gst-indent
2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
rtsp-server: Run gst-indent
Since it wasn't using the upstream common previously, there was no
indentation check before commiting.
2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
* gst/rtsp-server/rtsp-media-mapping.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp-server: Some more doc fixups
2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* Makefile.am:
Makefile: Add cruft-cleaning support
2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* Makefile.am:
* configure.ac:
* docs/Makefile.am:
* docs/libs/Makefile.am:
* docs/libs/gst-rtsp-server-docs.sgml:
* docs/libs/gst-rtsp-server-sections.txt:
* docs/libs/gst-rtsp-server.types:
* docs/version.entities.in:
docs: Add gtk-doc build system
2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
Makefile.am: Use standard GIR make behaviour
2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
* autogen.sh:
* configure.ac:
autogen/configure: Bring more in sync to standard gst module behaviour
2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: warn and fail when gstrtpbin is not found
2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
configure: open 0.11 branch
2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
* .gitmodules:
* common:
Add common submodule
2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
* common/ChangeLog:
* common/Makefile.am:
* common/c-to-xml.py:
* common/check.mak:
* common/coverage/coverage-report-entry.pl:
* common/coverage/coverage-report.pl:
* common/coverage/coverage-report.xsl:
* common/coverage/lcov.mak:
* common/gettext.patch:
* common/glib-gen.mak:
* common/gst-autogen.sh:
* common/gst-xmlinspect.py:
* common/gst.supp:
* common/gstdoc-scangobj:
* common/gtk-doc-plugins.mak:
* common/gtk-doc.mak:
* common/m4/.gitignore:
* common/m4/Makefile.am:
* common/m4/README:
* common/m4/as-ac-expand.m4:
* common/m4/as-auto-alt.m4:
* common/m4/as-compiler-flag.m4:
* common/m4/as-compiler.m4:
* common/m4/as-docbook.m4:
* common/m4/as-libtool-tags.m4:
* common/m4/as-libtool.m4:
* common/m4/as-python.m4:
* common/m4/as-scrub-include.m4:
* common/m4/as-version.m4:
* common/m4/ax_create_stdint_h.m4:
* common/m4/check.m4:
* common/m4/glib-gettext.m4:
* common/m4/gst-arch.m4:
* common/m4/gst-args.m4:
* common/m4/gst-check.m4:
* common/m4/gst-debuginfo.m4:
* common/m4/gst-default.m4:
* common/m4/gst-doc.m4:
* common/m4/gst-error.m4:
* common/m4/gst-feature.m4:
* common/m4/gst-function.m4:
* common/m4/gst-gettext.m4:
* common/m4/gst-glib2.m4:
* common/m4/gst-libxml2.m4:
* common/m4/gst-plugindir.m4:
* common/m4/gst-valgrind.m4:
* common/m4/gtk-doc.m4:
* common/m4/introspection.m4:
* common/m4/pkg.m4:
* common/mangle-tmpl.py:
* common/plugins.xsl:
* common/po.mak:
* common/release.mak:
* common/scangobj-merge.py:
* common/upload.mak:
common: Remove static version
2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
* common/m4/introspection.m4:
Update introspection.m4 to match usage
2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* README:
README: update
Remove old stuff from the README
2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
back to development
=== release 0.10.7 ===
2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
release 0.10.7
2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-ogg.c:
test-ogg: remove parsers
Remove the parsers, they are not needed anymore as oggdemux now outputs normal
buffers with timestamps. Using the parsers also seems to break things.
2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
Updated Vala bindings
2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* common/m4/introspection.m4:
* configure.ac:
* gst/rtsp-server/Makefile.am:
Added initial gobject-introspection support
2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: don't use host for shared hash key
When we generate the key to share made between connections, don't include the
host used to connect so that we can share media even if between clients that
connected with localhost and ones with the ip address.
2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* bindings/vala/Makefile.am:
build: fix distcheck
2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
Update Vala bindings
2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* bindings/vala/Makefile.am:
* configure.ac:
Fix configure checks and installation location for Vala bindings
Fixes bug #628676.
2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
back to development
=== release 0.10.6 ===
2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
configure: release 0.10.6
2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: help the compiler a little
2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session.c:
media: cleanup media transport before freeing
Cleanup the media transport data before freeing. In particular, remove the qdata
from the rtpsource object.
2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media-factory: add eos-shutdown property
Add an eos-shutdown property that will send an EOS to the pipeline before
shutting it down. This allows for nice cleanup in case of a muxer.
Fixes #625597
2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: use multiudpsink send-duplicates when we can
If we have a new enough multiudpsink with the send-duplicates property, use this
instead of doing our own filtering. Our custom filtering code should eventually
be removed when we can depend on a released -good.
2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: don't leak destinations
Refactor and cleanup the destinations array when the stream is destroyed.
2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: don't add udp addresses multiple times
Keep track of the udp addresses we added to udpsink and never add the same udp
destination twice. This avoids duplicate packets when using multicast.
2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: disable use of SO_LINGER
SO_LINGER cause the client to fail to receive a TEARDOWN message because the
server close()s the connection.
2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: use 5 second linger period in SO_LINGER
Wait 5 seconds before clearing the send buffers and reseting the connection with
the client when we do a close. This should be enough time to get the message to
the client.
See #622757
2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
* gst/rtsp-server/rtsp-server.c:
server: use SO_LINGER
SO_LINGER on the socket will make sure that any pending data on the socket is
flushed ASAP and that the socket connection is reset. This makes sure that the
socket can be reused immediately.
Fixes 622757
2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/README:
README: add blurb about shared media factories
2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
* gst/rtsp-server/rtsp-media.c:
Add stdlib.h for atoi()
2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* bindings/python/Makefile.am:
* bindings/vala/Makefile.am:
build: distcheck fixes
Fix 'make distcheck', somewhat (it still fails because it tries to
install files into /usr/share/vala/vapi/ irrespective of the
configured prefix).
2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump core/base requirements to released version
Makes things less confusing for people.
2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: fail if GStreamer core/base requirements are not met
2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: improve client cleanups
Make sure the session does not timeout when using TCP. We need to do this
because quicktime player does not send RTCP for some reason in tunneled
mode.
Refactor some cleanup code.
Fixes #612915
2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
session: add support for prevent session timeouts
Add an atomix counter to prevent session timeouts when we are, for example,
streaming over TCP.
2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: fix unlink on session timeouts
When our session times out, make sure we unlink all streams in this
session.
Remove the tunnelid when closing the connection.
2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
session: small cleanups
2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: handle lost_tunnel callbacks
Handle lost_tunnel callbacks and use it to store the tunnelid back into the
hashtable so that we can reuse it for when the client reopens the POST
socket.
Close the connection after a TEARDOWN.
Make sure or watchid is cleared when the watch is removed.
Fixes #612915
2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
rtsp-server: add more support for multicast
2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: allow configuration of allowed lower transport
2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
* gst/rtsp-server/rtsp-server.c:
rtsp: keep track of server ip and ipv6
Keep track of how the client connected to the server and setup the udp ports
with the same protocol.
Copy the server ip address in the SDP so that clients can send RTCP back to
us.
2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
session: indent
2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use right size for malloc
2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
server: comment ipv6 server listening address
2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: allow for ipv6 sockets
2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
server: rework server part
Allow setting a bind address, make sure we can deal with ipv6.
Remove the port property and change with the service property.
2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.h:
media: update comments a little
2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: make content-base better
Use the URI formatting functions to make a content-base. Also make sure that
there is a trailing / at the end.
2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: guard against invalid paths
2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-video.c:
test: catch server bind errors
2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-media.c:
rtspmedia: emit "unprepared" if _prepare fails.
Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
media object is removed from its factory's cache.
2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: collect media position when seek completes
2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
* gst/rtsp-server/rtsp-client.c:
client: call unlink_streams in client finalize
Fixes #599027
2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: limit the time to wait to something huge
Avoid waiting forever but limit the timeout to 20 seconds.
2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-sdp.c:
sdp: reindent and check for prepared status
2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session.c:
media: avoid doing _get_state() for state changes
When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
until the media is prerolled or in error. This avoids doing a blocking call of
gst_element_get_state() that can cause lockups when there is an error.
Fixes #611899
2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: reindent
2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: better error handling
Improve the error handling a bit.
2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: rework transport parsing
Rework the transport parsing code so that we can ignore transports we don't
support instead of just picking the first one we can parse.
Configure a (for now hardcoded) destination for multicast transports.
2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: set multicast sink parameters
Disable loop and automatic multicast join on the udpsink elements.
Add some more debug info.
Reset some state variables in the right place.
Use the right port numbers for multicast.
2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
session: handle transport setup correctly
Handle UDP, MCAST and TCP transport negotiation more correctly.
Store the server session SSRC in the transport.
2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: implement error_full
Implement error_full to avoid some segfaults when the rtspconnection calls it.
See #608245
2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/README:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-server.c:
docs: update docs and comments
2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
* gst/rtsp-server/rtsp-sdp.c:
sdp: make server work better when behind a proxy
2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-client.c:
client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
Use GStreamer's debugging subsystem
2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media-factory.c:
server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
back to development
=== release 0.10.5 ===
2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
release 0.10.5
2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
configure: bump required versions
2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
* gst/rtsp-server/rtsp-client.c:
client: call weak-unref on client->sessions from finalize
Fixes bug #596305
2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media.c:
media: Fixed crasher where caps got unref'ed too often
2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* configure.ac:
* pkgconfig/.gitignore:
* pkgconfig/Makefile.am:
* pkgconfig/gst-rtsp-server-uninstalled.pc.in:
Added pkg-config file to use gst-rtsp-server uninstalled
2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: add some docs
2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp: Use gst_rtsp_watch_send_message().
Use gst_rtsp_watch_send_message() since the old API which used
gst_rtsp_watch_queue_message() has been deprecated.
2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
back to development
=== release 0.10.4 ===
2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
Release 0.10.4
2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp: allocate channels in TCP mode
When the client does not provide us with channels in TCP mode, allocate channels
ourselves.
2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: don't crash when tunnelid is missing
When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
don't crash but return an error response to the client.
Fixes #589489
2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
bindings: update vala bindings with new method
2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
sessionpool: add function to filter sessions
Add generic function to retrieve/remove sessions.
2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* configure.ac:
configure: bump core/base requirements to release
2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: fix indentation
2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media.c:
Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media.c:
set state and remove elements of media in for loop
2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
Added gst_rtsp_media_remove_elements function to Vala bindings
2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Added gst_rtsp_media_remove_elements function
2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
* gst/rtsp-server/rtsp-media.c:
Don't use name for gstrtpbin so we can add multiple instances to the pipeline
2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
Updated Vala bindings
2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Added vmethod unprepare to GstRTSPMedia
The default implementation sets the state of the pipeline to GST_STATE_NULL
2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
Made collect_streams function public
2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
Added vmethod create_pipeline to GstRTSPMediaFactory
The pipeline is created in this method and the GstRTSPMedia's element is added to it
2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: use g_source_destroy()
We need to use g_source_destroy() because we might have added the source to a
different main context than the default one.
2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-params.c:
* gst/rtsp-server/rtsp-params.h:
rtsp: prepare for handling GET/SET_PARAMETER
Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
is a body now.
Fix return codes of handlers.
2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: don't leak session pads
2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: clean up the messages a bit
2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-sdp.c:
sdp: warn and skip streams without media
2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
vala: Fixed typo in header file of RTSPMediaStream
2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: fix message
Fix a debug message
Make dumping RTCP stats configurable
2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: be less verbose and leak less
2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: don't leak the destination address
2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
rtsp: use RTCP to keep the session alive
Use the RTCP rtcp-from stats field to find the associated session and use this
to keep the session alive.
2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
session: add 5sec to the real session timeout
Allow the session to live 5sec longer before really timing out. This should give
clients some extra time to keep the session active.
2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: replay OK to GET/SET_PARAMETER
Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
so that we return OK for those requests.
2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: keep track of active transports
Keep track of which transport is active to avoid closing the connection too
soon.
Remove the destination transport also when going to NULL.
Print some stats about the SDES and other RTCP messages we receive from the
clients.
2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-sdp.c:
example: add SDP relay example
2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: also count active TCP connections
2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp: add support for dynamic elements
Add support for dynamic elements.
Don't set live pipelines back to paused.
2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-sdp.c:
sdp: don't add encoding name when absent in caps
2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: warn when we can't do RTP-Info
2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
factory: factor out the stream construction
2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: only add RTP-Info when we have the info
Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
depayloader.
2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
back to development
=== release 0.10.3 ===
2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
release: 0.10.3
- Fixes a bug where it put the wrong verion in pkgconfig
- Link RTP and RTCP sources
2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: link the RTP udpsrc to the session manager
Link the RTP udpsrc and the appsrc to the session manager so that they don't
shut down when the client sends a packet to open firewalls.
2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* pkgconfig/gst-rtsp-server.pc.in:
Don't use hard-coded version number in pkg-config file
2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
back to development
=== release 0.10.2 ===
2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
release 0.10.2
2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* .gitignore:
* common/m4/.gitignore:
* examples/.gitignore:
* pkgconfig/.gitignore:
add some .gitignore files
2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: seek to key frames
2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
media: emit the unprepared signal by id
Emit the unprepared signal by id instead of name and set the media as
reused.
2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-media.c:
Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-server.c:
Added finalize function to GstRTPSPServer to unref session pool and media mapping
2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
Updated vala bindings
2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
server: use appsink and appsrc with the API
Use the appsink/appsrc API instead of the signals for higher
performance.
2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-ogg.c:
tests: set the payload type correctly
2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
factory: connect to the unprepare signal
Connect to the unprepare signal for non-reusable media so that we can remove
them from the cache.
2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: add signal to notify of unprepare
2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
media: more work on making the media shared
Add a reusable flag to medias, indicating that they can be reused after a state
change to NULL.
Small cleanups.
2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-readme.c:
examples: mark the example as shared for testing
2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
client: support shared media
Always perform the state actions even if the target state of the pipeline is
already correct, we still want to add/remove the transports when we are dealing
with shared media.
Keep a counter of the number of active transports for a media so that we can use
this to perform a state change when needed.
Perform a state change of the pipeline only when the first transport was added
or when there are no active transports.
2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
client: fix refcounting crasher
Don't need to remove the weak refs in the finalize methods, they are already
removed in the dispose.
Don't register the callback with a DestroyNofity.
2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Fix rtsp client refcount management in TCP mode.
Don't unref a client ref we never had. Fixes an unref
of an already-free client object after a client
teardown request for me.
2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
docs: fix typo in API docs
2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
More seeking fixes.
Keep the udp sources in playing even if we go to paused. unlock the sources when
we shut down.
Add some more debug info.
Only seek when we need to.
Keep track of the position when we go to paused.
2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Add beginnings of seeking.
Parse the Range header and perform a seek on the pipeline for the requested
position. It's disabled currently until I figure out what's going wrong.
2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
allow pause requests for now.
--
2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Remove weak ref on the session in teardown
We need to remove our weakref from the session when we do a teardown because
else we close the TCP connection prematurely.
2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-session-pool.c:
Do some more session cleanup
Make session timeout kill the TCP connection that currently watches the
session.
Remove the client timeout property.
2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Add TCP transports
Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
connection.
2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
* examples/test-launch.c:
Add example server that takes launch lines
Add an example server that streams any -launch line.
2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-readme.c:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Add support for live streams
Add support for live streams and ranges
Start on handling TCP data transfer.
2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
Free the pipeline before other things
---
2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Only free the pending tunnel if there is one
--
2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media.c:
rtsp-server: Add support for tunneling
Add support for tunneling over HTTP.
Use new connection methods to retrieve the url.
Dispatch messages based on the message type instead of blindly
assuming it's always a request.
Keep track of the watch id so that we can remove it later.
Set the media pipeline to NULL before unreffing the pipeline.
2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Fix for channel -> watch rename in gstreamer
Rename the RTSPChannel to RTSPWatch and remove an unused variable.
2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Use ASYNC RTSP io
Use the async RTSP channels instead of spawning a new thread for each client.
If a sessionid is specified in a request, fail if we don't have the session.
2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
Add better debug info
Add some better debug info.
2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/test-video.c:
Time out sessions
Add support for session timeouts in the example.
2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
Pass GTimeVal around for performance reasons
Get the current time only once and pass it around so that sessions don't have to
get the current time anymore.
Add experimental support for a GSource that dispatches when the session needs to
be cleaned up.
2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Add better support for session timeouts
Add a method to request the number of milliseconds when a session will timeout.
2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Add suport for RTP manager monitoring
Add the first stage in monitoring the rtp manager.
Make sure we don't update the state to something we don't want.
2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Add support for session keepalive
Get and update the session timeout for all requests. get the session as early as
possible.
2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Handle media bus messages
Handle media bus messages in a custom mainloop and dispatch them to the
RTSPMedia objects. Let the default implementation handle some common messages.
2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
Some more session timeout handling
Move the session header setting code to a central place so that we always add
the timeout parameter too.
Handle timeouts by running the session cleanup code.
Stop media before cleaning up.
2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Add timeout property
Add a timeout property ot the client and make the other properties into GObject
properties.
2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-pool.c:
Use getters and setters in property code
Use the getters and setters for the timeout property instead of locking
ourselves.
2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Add more timeout stuff
Add method to check if a session is expired.
Add method to perform cleanup on a session pool.
2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Add beginnings of session timeouts and limits
Add the timeout value to the Session header for unusual timeout values.
Allow us to configure a limit to the amount of active sessions in a pool. Set a
limit on the amount of retry we do after a sessionid collision.
Add properties to the sessionid and the timeout of a session. Keep track of
creation time and last access time for sessions.
2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Cleanup of sessions and more
Fix the refcounting of media and sessions in the client. Properly clean up the
session data when the client performs a teardown.
Add Server header to responses.
Allow for multiple uri setups in one session.
Add Range header to the PLAY response and add the range attribute to the SDP
message.
Fix the session pool remove method, it used the wrong key in the hashtable. Also
give the ownership of the sessionid to the session object.
2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
Rename a variable
Rename the 'server_port' variable to simply 'port'.
2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Rework the way we handle transports for streams
Make the media accept an array of transports for the streams that we have
configured for the play/pause requests.
Implement server states for a client and its media.
Require 0.10.22.1 (git HEAD) of gstreamer.
2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
Drop const from functions dealing with urls
Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
have the right const in them.
2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
Fix various leaks
Fix some leaks.
2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
More cleanups
Don't keep a reference to the GstRTSPMedia in the stream.
Free more things when freeing the GstRTSPMedia.
2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/README:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
More docs and small cleanups
Add some more docs and update the README
Cleanup some method names.
Remove an unneeded idx field in the GstRTSPMediaStream
2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* docs/README:
* examples/Makefile.am:
* examples/test-readme.c:
Add a README and more example code
Add a README file that contains a small introduction on how to use the server
along with the example code explained in the readme.
2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-server.c:
Fix some leaks and change default port
Fix some memory leaks by setting the udpsrc elements to the unlocked state after
we finished the initial preroll. If we keep them locked, setting the pipeline to
NULL will not stop and clean up the sources correctly.
Change the default RTSP port to 8554 aka the official alternative RTSP port.
2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Cleanups to the session object
Remove some unneeded variables in the session state of a stream such as the
owner media and the server transport.
Get the configuration of a media stream in a session based on the media_stream
in the original object instead of our cached index.
Free more data in the finalize method.
2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Cleanups and reuse media from DESCRIBE
Handle thread create errors.
Rename some internal methods to better match what they actually do.
Handle misconfiguration of session_pool and media_mapping gracefully.
Cache the DESCRIBE media and uri in the client connection and reuse them when
we receive a SETUP request in the same connection for the same uri.
Cleanup the client connection object.
2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Add shared properties to media and factory
Add the shared property to media.
Implement some simple caching in the factory depending on if the media is shared
or not.
2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Add a little comment
Add some comment about the content-base header.
2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-video.c:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Reorganize things, prepare for media sharing
Added various other test server examples
Move the SDP message generation to a separate helper.
Refactor common code for finding the session.
Add content-base for realplayer compatibility
Clean up request uris before processing for better vlc compatibility.
Move prerolling and pipeline construction to the RTSPMedia object.
Use multiudpsink for future pipeline reuse.
2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
Back to development
Back to 0.10.1.1
=== release 0.10.1 ===
2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* configure.ac:
Make 0.10.1 release
Release 0.10.1
2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* bindings/vala/Makefile.am:
Fix make dist
Add more directories and files to the dist.
2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/python/Makefile.am:
* bindings/python/rtspserver.override:
Fixed compile error of python bindings
2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
Marked values as nullable accordingly
2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.excludes:
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
Updated Vala bindings
2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media-mapping.h:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
Cleanups and doc updates
Add some more documentation and do some minor cleanups here and there.
2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
More improvements
Rename GstRTSPMediaBin to GstRTSPMedia
Parse the request url into a GstRTSPUri object and pass this object to the
various handlers and methods that require the uri.
2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/main.c:
Update example
Add some more docs and remove some old code from the example.
2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
Handle state change failures better
Handle state change failures better when changing the state of the pipeline to
determine the SDP.
2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
Make element creation more extendible
Add get_element vmethod to the default MediaFactory so that subclasses can just
override that method and still use the default logic for making a MediaBin from
that.
2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/main.c:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media-mapping.c:
* gst/rtsp-server/rtsp-media-mapping.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
Make the server handle arbitrary pipelines
Make GstMediaFactory an object that can instantiate GstMediaBin objects.
The GstMediaBin object has a handle to a bin with elements and to a list of
GstMediaStream objects that this bin produces.
Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
with methods to register and remove those mappings.
Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
used by the server instance.
Modify the example application so that it shows how to create custom pipelines
attached to a specific mount point.
Various misc cleanps.
2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
Allow setting a custom media factory for a server
2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Allow setting a custom media factory for a client.
2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
Add Makefile entry for the media factory
2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
Add media factory to map urls to media pipeline objects.
2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
Add comments. Remove unused field
2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
Allow custom session pools to override the session id allocation algorithms Add some comments.
2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session.h:
Add some comments.
2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Move the connection code in one place Add some comments
2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
Make vmethod to create and accept new clients. Add some docs.
2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
Name the parameters more appropriately.
2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-session-pool.c:
Do some more cleanup of the session pool.
2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-client.c:
Check if return value of gst_rtsp_session_get_media is not NULL
2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/Makefile.am:
Install rtsp-session and rtsp-session-pool headers
2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* .gitignore:
* Makefile.am:
* acinclude.m4:
* bindings/python/Makefile.am:
* bindings/python/arg-types.py:
* bindings/python/codegen/Makefile.am:
* bindings/python/codegen/__init__.py:
* bindings/python/codegen/argtypes.py:
* bindings/python/codegen/code-coverage.py:
* bindings/python/codegen/codegen.py:
* bindings/python/codegen/definitions.py:
* bindings/python/codegen/defsparser.py:
* bindings/python/codegen/docextract.py:
* bindings/python/codegen/docgen.py:
* bindings/python/codegen/fileprefix.override:
* bindings/python/codegen/fileprefixmodule.c:
* bindings/python/codegen/h2def.py:
* bindings/python/codegen/mergedefs.py:
* bindings/python/codegen/mkskel.py:
* bindings/python/codegen/override.py:
* bindings/python/codegen/reversewrapper.py:
* bindings/python/codegen/scmexpr.py:
* bindings/python/rtspserver-types.defs:
* bindings/python/rtspserver.defs:
* bindings/python/rtspserver.override:
* bindings/python/rtspservermodule.c:
* configure.ac:
Add python bindings.
2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* bindings/Makefile.am:
* configure.ac:
Don't go into python dir when requirements for python bindings are missing
2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* bindings/Makefile.am:
* bindings/vala/Makefile.am:
* configure.ac:
Install Vala bindings if vala is available
2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server-0.10.deps:
* bindings/vala/gst-rtsp-server-0.10.vapi:
* bindings/vala/packages/gst-rtsp-server-0.10.deps:
* bindings/vala/packages/gst-rtsp-server-0.10.excludes:
* bindings/vala/packages/gst-rtsp-server-0.10.files:
* bindings/vala/packages/gst-rtsp-server-0.10.gi:
* bindings/vala/packages/gst-rtsp-server-0.10.metadata:
* bindings/vala/packages/gst-rtsp-server-0.10.namespace:
Regenerated Vala bindings
2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* bindings/vala/gst-rtsp-server.vapi:
* bindings/vala/packages/gst-rtsp-server.metadata:
Fixed typo in included headers for vala bindings
2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* Makefile.am:
* configure.ac:
* pkgconfig/Makefile.am:
* pkgconfig/gst-rtsp-server.pc.in:
Added pkgconfig file
2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
* bindings/vala/gst-rtsp-server.vapi:
* bindings/vala/packages/gst-rtsp-server.excludes:
* bindings/vala/packages/gst-rtsp-server.gi:
* bindings/vala/packages/gst-rtsp-server.metadata:
Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
* bindings/vala/gst-rtsp-server.vapi:
* bindings/vala/packages/gst-rtsp-server.deps:
* bindings/vala/packages/gst-rtsp-server.files:
* bindings/vala/packages/gst-rtsp-server.gi:
* bindings/vala/packages/gst-rtsp-server.metadata:
* bindings/vala/packages/gst-rtsp-server.namespace:
Added Vala bindings
2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
* gst/rtsp-server/rtsp-session.c:
Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
* examples/Makefile.am:
* gst/rtsp-server/Makefile.am:
Put GStreamer version in library name
2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
* gst/rtsp-server/Makefile.am:
Fix some issues to pass distcheck
2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtsp-server/rtsp-server.c:
Added port property to GstRTSPServer class.
2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* Makefile.am:
* autogen.sh:
* configure.ac:
* examples/Makefile.am:
* examples/main.c:
* gst/Makefile.am:
* gst/rtsp-server/Makefile.am:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-server.c:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-pool.c:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
* src/Makefile.am:
Split in library and example program
2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
* src/rtsp-client.h:
Removed obsolete variable
2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
* src/rtsp-client.c:
* src/rtsp-client.h:
Removed pipeline variable GstRTSPClient, because it's only used in one function
2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* src/rtsp-media.c:
Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
* src/rtsp-session.c:
Initialize some more vars.
2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
* src/rtsp-session.c:
Initialize variable to avoid compiler warning.
2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
* .gitignore:
Add a reasonable generic .gitignore
Reference in a new issue View git blame Copy permalink