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3564 commits

Author SHA1 Message Date
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
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
ff5d3313d4 Release 1.1.1 2013-06-05 18:31:27 +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
5065e76b1c audio: Add gst_audio_info_is_equal() 2013-05-30 23:56:52 +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
Wim Taymans
15f3c995aa Revert "rtspconnection: Use a GSocketAddressNumerator to resolve the addresses" 
This reverts commit 15a0bb0a10.

We should be using GSocketClient
 2013-05-30 07:20:51 +02:00
Sebastian Dröge
15a0bb0a10 rtspconnection: Use a GSocketAddressNumerator to resolve the addresses 
Instead of just trying the first possible resolution we're trying all
resolutions until one works.
 2013-05-27 14:53:48 +02:00
Wim Taymans
97784b1563 video-format: fix NV16 unpack 
We can just use the NV12 functions, the only difference is the
vertical subsampling.
 2013-05-27 11:53:27 +02:00
Wim Taymans
73190bcf79 video-chroma: add interlaced flag 2013-05-27 11:25:09 +02:00
Wim Taymans
0c60f0daa4 video-chroma: add chroma resampler 
Add functions to up/downsample chroma in horizontal and vertical
directions. These functions work in-placeand are meant to be used on the
input/output of the pack/unpack functions.
 2013-05-27 11:05:07 +02:00
Wim Taymans
2924365020 video: don't perform subsampling while packing 
Don't perform subsampling when packing but let this be done by a
separate subsampling step.
 2013-05-27 11:05:06 +02:00
Wim Taymans
b5de0552a5 video: move chroma functions to separate file 2013-05-27 11:05:06 +02:00
Wim Taymans
38317e3f09 videometa: fix docs 2013-05-27 11:05:06 +02:00
Sebastian Dröge
c5e9df4b51 videoencoder: Don't require an output state to be set before allocating output buffers 2013-05-25 16:08:06 +02:00
Sebastian Dröge
b8c6413a8e audio: Always provide a buffer in gst_audio_(enc|dec)oder_allocate_output_buffer() 
We have no way of tell the caller of the exact error (e.g. if we're flushing),
so will have to wait until the caller uses API that returns a GstFlowReturn,
for example when pushing this buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=700006
 2013-05-24 16:54:46 +02:00
Sebastian Dröge
0c2c909497 video: Always provide a buffer in gst_video_(enc|dec)oder_allocate_output_buffer() 
We have no way of tell the caller of the exact error (e.g. if we're flushing),
so will have to wait until the caller uses API that returns a GstFlowReturn,
for example when pushing this buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=700006
 2013-05-24 16:54:46 +02:00
Alexander Schrab
a049b102da alsasrc: Make using driver timestamps possible 
https://bugzilla.gnome.org/show_bug.cgi?id=699744
 2013-05-20 11:25:17 +02:00
Benjamin Gaignard
e90e2bb822 dmabuf: Make sure that memory is unmapped before releasing it 
Be sure that memory is unmapped before releasing it.

https://bugzilla.gnome.org/show_bug.cgi?id=700411
 2013-05-17 09:50:23 +02:00
Tim-Philipp Müller
612e20d4f6 video: make mask arguments to gst_video_format_from_masks() unsigned 
These should really be unsigned.
 2013-05-16 11:35:58 +01:00
Benjamin Gaignard
5da2bd3216 video: fix gst_video_format_from_masks() for little endian masks with alpha 
Need to byte-order swap the alpha mask as well in this case.

https://bugzilla.gnome.org/show_bug.cgi?id=700413
 2013-05-16 10:36:20 +01:00
Michael Olbrich
ced858fa65 dmabuf: set the initial memory size to the full size 
https://bugzilla.gnome.org/show_bug.cgi?id=700427
 2013-05-16 11:17:57 +02:00
Tim-Philipp Müller
77405b97ed video: update disted orc backup files to fix build without liborc 
https://bugzilla.gnome.org/show_bug.cgi?id=700400
 2013-05-15 18:20:50 +01:00
Arnaud Vrac
af24e23880 video: add NV16 format 
This format is usually used by hardware video decoders for 4:2:2 sampling

https://bugzilla.gnome.org/show_bug.cgi?id=700377
 2013-05-15 13:46:46 +02:00
Sebastian Dröge
be154ee9d6 audio-info: Always pass NULL as position parameter to gst_audio_info_set_format() 
https://bugzilla.gnome.org/show_bug.cgi?id=700259
 2013-05-15 09:26:56 +02:00
Nicolas Dufresne
94b7ae7767 rtpbasepayload: Delay segment event after caps 
https://bugzilla.gnome.org/show_bug.cgi?id=700222
 2013-05-14 09:50:22 +02:00
Sebastian Dröge
b401f447d2 audio-info: For more than 64 channels don't allow a channel layout 
More than 64 channels have all channels unpositioned.

https://bugzilla.gnome.org/show_bug.cgi?id=700259
 2013-05-14 09:34:21 +02:00
Mathieu Duponchelle
6f233f67ef videodecoder: don't set the list to NULL after taking its address 2013-05-10 09:22:07 +02:00
Sebastian Dröge
2cc6a62b31 videoencoder: Make sure to push any pre-caps events before the caps are set 2013-05-09 16:05:59 +02:00
Sebastian Dröge
82f1572205 videodecoder: Make sure to not push any post-caps events before we have caps 
and that we push pre-caps events before we push caps, even if we don't
have a GstVideoFrame yet.
 2013-05-09 16:05:59 +02:00
Sebastian Dröge
ba8e7062a4 Revert "videodecoder: If a frame is to be dropped, don't update timestamps" 
This reverts commit c9c5cd8eef.
 2013-05-09 10:37:06 +02:00
Sebastian Dröge
c9c5cd8eef videodecoder: If a frame is to be dropped, don't update timestamps 2013-05-09 08:54:45 +02:00
Sebastian Dröge
351405d8a0 audio: Make sure to push pre-caps events before the caps event 2013-05-08 15:56:34 +02:00
Sebastian Dröge
3e4aec6e7b video: Make sure to push pre-caps events before the caps event 
https://bugzilla.gnome.org/show_bug.cgi?id=699894
 2013-05-08 15:50:34 +02:00