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3562 commits

Author SHA1 Message Date
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
Julien Moutte
6f9006c9b9 riff: Manually calculate bitrate of ADPCM streams 
Some ADPCM encoding tools like Oxelon generate WAV files with
wrong format header declaring an invalid bitrate.

As wavparse uses the average bitrate to calculate timestamps
and duration the decoder can be confused by receiving timestamps
completely out of sync with the decoded samples.

ADPCM is a CBR audio codec so we can calculate the average bitrate
instead of trusting the format header.

https://bugzilla.gnome.org/show_bug.cgi?id=636245
 2013-05-07 15:01:05 +02:00
Tim-Philipp Müller
1a83c7a36a riff: wma can have more than 6 channels 
Some versions anyway.
 2013-05-03 16:24:55 +01:00