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Author SHA1 Message Date
Wim Taymans
bcc5ac5298 rtsp: dispatch when initial buffer has data 
When we have data in the inital buffer, dispath the read function to read it
even if the socket has no data to read.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702652
 2013-06-21 11:50:33 +02:00
Wim Taymans
ad6c16fdfc rtsp: manage writer child source better 
Only add the write child source when we have something to write or else
we will dispatch forever without doing anything.
 2013-06-20 17:28:46 +02:00
Jonas Holmberg
82e5ec553b audioencoder: unref before memset 
Unref allocator and input_caps in encoder context before memsetting the
context.
 2013-06-19 13:56:28 +02:00
Edward Hervey
420dacb2d5 xmptag: More efficient GSList usage 
Instead of constantly appending (which gets more and more expensive), just
prepend to the list (O(1)) and reverse the list before usage.

https://bugzilla.gnome.org/show_bug.cgi?id=702545
 2013-06-19 12:01:44 +02:00
Branko Subasic
4dd5c5b808 rtpbuffer: add gst_rtp_buffer_get_payload_bytes 
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.

The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.

https://bugzilla.gnome.org/show_bug.cgi?id=698562
 2013-06-18 11:23:40 +02:00
Ognyan Tonchev
f240d34c7e audiobasesrc: add 2 missing gst_buffer_unmap () calls 
There are 2 missing calls to gst_buffer_unmap () in the error handling in
create ().

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702467
 2013-06-17 16:34:26 +02:00
Sreerenj Balachandran
8a2dbeb1ef playsink: Fix the block diagram of deinterlace bin. 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702465
 2013-06-17 16:28:44 +02:00
Brendan Long
b5f7a621bb playbin: Emit {audio,text,video}-changed signals when pads are removed 
https://bugzilla.gnome.org/show_bug.cgi?id=702195
 2013-06-14 14:23:11 +02:00
Sebastian Dröge
eb09e561c4 videoconvert: Fix leaking of the chroma resample helper objects 2013-06-11 15:24:05 +02:00
Sreerenj Balachandran
eabd2a5c7b tests: add more unit test for playbin 
Add unit test for autoplugging of video_decoder/video_sink combination
based on capsfeatures.
 2013-06-11 15:17:37 +02:00
Sebastian Dröge
567be29db2 rtspconnection: Make sure to set a sensible default port for the GSocketConnection 
Otherwise it will connect to port 0 if no port is given in the URI.

https://bugzilla.gnome.org/show_bug.cgi?id=701798
 2013-06-10 15:31:38 +02:00
Sebastian Dröge
15375861b5 adder: Reject segments that have a different rate than the output segment 
adder does no rate conversion.
 2013-06-09 19:20:20 +02:00
Sebastian Dröge
4465741222 playbin: When activating a fixed sink, proxy error messages too 
If activating a fixed sink fails, everything will fail later anyway
and we can just error out early.
 2013-06-08 23:51:13 +02:00
Sebastian Dröge
44352deadb playbin: Improve autoplugging of decoder/sink combinations by trying to activate the sink 
And if that fails don't bother autoplugging that sink. Also gives
us more accurate sink caps.
 2013-06-08 23:34:53 +02:00
Sebastian Dröge
ff8839d97b playbin: Proxy the playbin context to the sinks 2013-06-08 23:22:54 +02:00
Sebastian Dröge
77af24c493 playbin: Proxy sink messages if we activate a sink in playbin already 
This makes sure the application gets any context related messages and
can do whatever is required to a) get the sink a context or b) share
the context with other elements in the pipeline.

The proxying is necessary because the sink is not a child element of
playbin, but instead will at a later point be a child of some bin
inside playsink.

https://bugzilla.gnome.org/show_bug.cgi?id=700967
 2013-06-08 23:22:54 +02:00
Sebastian Dröge
c49531f1b7 decodebin: Let serialize queries before caps events through 
Otherwise we're going to deadlock forever because no autoplugging
happens without having caps, but caps can never be send because
we're blocking.

Serialized queries before caps should never be sent unless really
necessary.
 2013-06-06 15:57:49 +02:00
Sebastian Dröge
15bbf42088 Back to development 2013-06-05 18:36:40 +02:00
Sebastian Dröge
ff5d3313d4 Release 1.1.1 2013-06-05 18:31:27 +02:00
Sebastian Dröge
597856bc8c Update .po files 2013-06-05 16:20:38 +02:00
Sebastian Dröge
c28d7b61a5 Automatic update of common submodule 
From 098c0d7 to 01a7a46
 2013-06-05 15:14:43 +02:00
Sebastian Dröge
bd62595a75 videodecoder: Change GST_WARNING to a GST_DEBUG 
It's completely normal for some decoders to queue 50-60 frames without
it causing any problems, e.g. RPi.
 2013-06-04 17:49:55 +02:00
Sebastian Dröge
c06377b385 audioencoder: Remove private copy of gst_audio_info_is_equal() 
And improve the public one a bit based on it.
 2013-06-01 09:06:22 +02:00
Brendan Long
63961242df rtspconnection: remove functions added in GLib 2.34 
g_pollable_stream_read and g_pollable_stream_write were added in GLib 2.34,
but Ubuntu 12.04 and Debian Wheezy still use GLib 2.32.

Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=701316
 2013-05-31 14:12:10 +02:00
Sebastian Dröge
cc5a476fc6 adder: Add GstChildProxy interface for the sinkpads 
This allows to set the sinkpad properties more easily.

Next step: Implement proper synchronization in adder, almost done!
 2013-05-30 23:56:52 +02:00
Sebastian Dröge
349174b48e adder: Hold object lock in setcaps a bit longer to prevent race conditions 2013-05-30 23:56:52 +02:00
Sebastian Dröge
dde8d8919d adder: Simplify segment event handling 
We don't care about upstream segments but generate our own. This
makes the code more similar to videomixer again.
 2013-05-30 23:56:52 +02:00
Sebastian Dröge
2953810ba5 adder: Use gst_audio_info_is_equal() to check if we get the same caps 2013-05-30 23:56:52 +02:00
Sebastian Dröge
5065e76b1c audio: Add gst_audio_info_is_equal() 2013-05-30 23:56:52 +02:00
Sebastian Dröge
0693b78e86 adder: Don't calls gst_pad_set_caps() on sinkpads 
It doesn't make much sense and the CAPS query handling
on the sinkpads should handle this.
 2013-05-30 23:56:52 +02:00
Sebastian Dröge
5e63a0f674 adder: Set GAP flag on silence buffers we created 2013-05-30 23:56:52 +02:00
Sebastian Dröge
667b856cbc adder: Remove caching of the processing function 
The compiler will generate a hashtable from the switch-case, and
we need to call functions explicitely for the volume!=1.0 cases
anyway.
 2013-05-30 23:56:52 +02:00
Sebastian Dröge
a76ade2d78 adder: Add support for per-stream volumes 2013-05-30 23:56:52 +02:00
Sebastian Dröge
addb07bc58 adder: Add optimized orc code for F64 processing 2013-05-30 23:56:52 +02:00
Sebastian Dröge
308c8fef68 adder: The output buffer must be readable and writable 2013-05-30 23:56:51 +02:00
Sebastian Dröge
509246cbac adder: Add support for muting individual pads 2013-05-30 23:56:51 +02:00
Sebastian Dröge
3974379a7f adder: Sync pad properties with the GstController 2013-05-30 23:56:51 +02:00
Sebastian Dröge
561ce54b27 adder: Add custom GstPad subclass to hold additional data and properties 
This will later allow to set per-stream volumes and mute status.
 2013-05-30 23:56:51 +02:00
Wim Taymans
0b933ff87b rtsp: add method to get the TLS connection 2013-05-30 17:31:13 +02:00
Wim Taymans
c0f13c2513 rtsp: let the sockets be reffed by the connection 
Don't add an extra ref to the sockets but use that of the connection.
Keep the connection around as an IOStream.
 2013-05-30 13:14:46 +02:00
Wim Taymans
2fc85d3980 rtsp: Cleanup the error path 
Make sure the watch is removed when we close the read socket because of
an error.
 2013-05-30 10:50:42 +02:00
Wim Taymans
ad5632586a rtsp: cleanup the watch reset function 2013-05-30 10:45:42 +02:00
Wim Taymans
07babdd68a rtsp: check if the streams are still active 
Don't try to read/write from an inactive stream. When we, for example,
transfer the second connection in tunneling mode, we are not interested anymore
on read/write activity on the old connection.
 2013-05-30 10:30:09 +02:00
Wim Taymans
d09028b4c3 rtsp: use child sources instead of using the sockets 
Use the source of the pollable input/output streams instead of
accessing the sockets directly.
 2013-05-30 07:36:52 +02:00
Wim Taymans
4ada677095 rtsp: fix input/output streams for tunneling 2013-05-30 07:35:18 +02:00
Wim Taymans
4f660c388c rtsp: don't use sockets for blocking 
Use the blocking and non-blocking API of the input/output streams instead
of polling the sockets directly. This also allows us to simplify some
code.
 2013-05-30 07:35:18 +02:00
Wim Taymans
909e119a23 rtsp: add TLS support 
Add flag to select TLS in the transport.
Enable TLS on the socketclient when we use a TLS uri.
 2013-05-30 07:35:14 +02:00
Wim Taymans
057bbae6c5 rtspconnection: use the input/output stream of clientconnection 
Don't use the raw sockets for RTSP communication but use the IOStream.
This is needed if we are going to use TLS later.
 2013-05-30 07:20:51 +02:00
Wim Taymans
2d41ee370c rtsp: set sockets non-blocking 2013-05-30 07:20:51 +02:00
Wim Taymans
a42a7be5df rtsp: use GSocketClient for making connections 
Use the GSocketClient API for making connections with the server. This removes a
bit of code and gives us the ability to do TLS later.
 2013-05-30 07:20:51 +02:00