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Author SHA1 Message Date
Wim Taymans
8428423c04 qtdemux: handle SEGMENT query 2013-06-20 11:31:22 +02:00
Kishore Arepalli
5b32891ae1 avidemux: duration query returns zero for DV video in avi 
https://bugzilla.gnome.org/show_bug.cgi?id=702625
 2013-06-19 11:17:22 +02:00
Sebastian Dröge
b001da2926 qtdemux: Disable usage of allocation queries 
This can only reliably work if demuxers have a
separate streaming thread per srcpad. This should be
done in a demuxer base class, which integrates parts
of multiqueue

https://bugzilla.gnome.org/show_bug.cgi?id=701856
 2013-06-19 11:07:48 +02:00
Alex Ashley
46a137c810 Avoid skipping moov atoms for fragmented MP4 files. 
bug #700505

Following a representation change that causes a resolution change,
the video decoder fails to decode correctly. Dashdemux detects the
representation change and pushes a new caps event and an
initialization segment (a new moov atom) to the downstream qtdemux,
but it doesn't handle this new moov yet, it will only parse the
first one it receives.

This commit changes qtdemux to accept a new moov in a dash bitstream
switching scenario.
 2013-06-19 01:44:22 -03:00
Thiago Santos
384e8f6c34 qtdemux: send stream-start only once for each stream 
Do not send stream start again when reconfiguring a pad for new caps.
That is common for adaptive streams
 2013-06-19 00:55:30 -03:00
Jens Georg
745be945ce rtpmp2tdepay: accept mislabelled streams from GStreamer 0.10 as well 
The mp2t payloader in 0.10 mislabelled the streams as MP2T-ES
instead of MP2T, so accept that as well for compatibility reasons.

https://bugzilla.gnome.org/show_bug.cgi?id=702457
 2013-06-17 15:39:17 +01:00
Wim Taymans
d9bc48edc9 rtspsrc: manage element state ourselves 
Lock the state of the all our elements and manage their states
outselves. Because we are working async, we can't rely on the state
change function to set the state at the right time or to return the
right return value from the state change function.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702046
 2013-06-16 05:40:13 +02:00
Bruno Gonzalez
e89a48616b matroskademux: Don't unlock stream lock without locking it first 
https://bugzilla.gnome.org/show_bug.cgi?id=702167
 2013-06-14 14:10:13 +02:00
Wim Taymans
51c9f7989f rtpsession: Use the right hashtable to calculate bandwidth 
Don't use an unused hashtable to iterate source to calculate bandwidth.
Remove unused code.
 2013-06-13 16:02:19 +02:00
Sebastian Dröge
01cc493944 Revert "videomixer: When all sinkpads are eos, update output segment stop and forward it" 
This reverts commit 2d3910fc79.

It's not solving any problem and instead causes code to fall apart.

https://bugzilla.gnome.org/show_bug.cgi?id=701519
 2013-06-12 18:25:59 +02:00
Tim-Philipp Müller
213cd2777b matroskademux: mark subtitle streams as sparse in stream-start event 
And also mark the streams that should be selected by default if
marked so in the headers.

https://bugzilla.gnome.org/show_bug.cgi?id=600648
 2013-06-12 15:31:22 +01:00
Stefan Sauer
39c4c5f251 audiopanorama: add prebuilt files 2013-06-11 22:14:33 +02:00
Stefan Sauer
349a60e164 audiopanorama: cleanup of transform() 
Only map input if we are reading it. Cleanup the logging and the comments a bit.
 2013-06-11 21:48:18 +02:00
Stefan Sauer
1dc06932a2 audiopanorama: use orc to speedup processing 
Use special variants for the case when we don't change the panorama (pan=0.0).
Simplify the processing functions by passing the panorama value directy instead
of the instance. Use orc for clearing buffers too.
 2013-06-11 21:48:18 +02:00
Mathieu Duponchelle
6e23f1fec4 videomixer: check last end_time after conversion to running segment 
The last end_time was saved after conversion, so the comparison
had to be made after conversion for it to make sense.

https://bugzilla.gnome.org/show_bug.cgi?id=701385
 2013-06-11 21:03:35 +02:00
Mathieu Duponchelle
4243714301 videomixer: add mix->segment.start to output_end_time 
When the segment start is not 0, this created a situation where
the output_end_time is inferior to output_start_time, and the duration
of the next buffer ended up underflowing.

https://bugzilla.gnome.org/show_bug.cgi?id=701385
 2013-06-11 21:03:03 +02:00
Sebastian Dröge
e2b46a776f matroskademux: Send stream headers after the segment event 
https://bugzilla.gnome.org/show_bug.cgi?id=700799
 2013-06-11 13:54:53 +02:00
Sebastian Dröge
adc9f0bd10 qtdemux: Do allocation query after exposing all pads and no-more-pads 
Also configure video streams as early as possible.

Related https://bugzilla.gnome.org/show_bug.cgi?id=701856
but not fixing that.
 2013-06-11 12:27:19 +02:00
Sebastian Dröge
ab275b62a8 flvdemux: Don't forward CAPS events from upstream 
Just use the default pad event handler.

https://bugzilla.gnome.org/show_bug.cgi?id=701976
 2013-06-11 12:27:19 +02:00
Stefan Sauer
4ef27eb0f9 audiopanorama: move the enum to the header and use instead of gint 
Move the enum for the processing method to the header so that we can use the
type for the instance struct.
 2013-06-09 20:39:48 +02:00
Sebastian Dröge
1ba08e331c wavenc: Link with libgstbase for GstByteWriter 2013-06-07 15:15:15 +02:00
Sebastian Dröge
db1c2a28a6 wavparse: Push stream-start event in pull mode before anything else 2013-06-07 13:27:07 +02:00
Sebastian Dröge
048866f1b1 Release 1.1.1 2013-06-05 18:31:40 +02:00
Sebastian Dröge
ea75b890dc wavenc: Fix taglist ref handling that made the unit test fail 2013-06-05 15:50:04 +02:00
Wim Taymans
0d27829a6b udpsink: avoid leaking the host 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701586
 2013-06-05 12:14:01 +02:00
Thiago Santos
7c12435f9b qtdemux: make sure taglist is writable before adding tags 
Avoids assertions
 2013-06-02 15:37:06 -03:00
Thiago Santos
78dfdee2aa qtdemux: effectively skip tracks that weren't listed on the 1st moov 
Without this, stream is NULL and the code will try to access it, leading
to segfaults.
 2013-06-02 13:06:15 -03:00
Thiago Santos
70fca21c28 qtdemux: skip redundant check 
!got_moov is already checked the line above
 2013-06-02 13:06:15 -03:00
Stefan Sauer
bcf1bba689 level: remove unused variables in instance struct 2013-06-01 21:34:37 +02:00
Anton Belka
db29522a43 wavenc: add tags & toc support 
Write tags as LIST INFO chunk. Format the toc as cue + LIST adtl chunk. Remove
old #ifdef'ed code.
 2013-06-01 21:34:37 +02:00
Wim Taymans
1f0600ee6f Revert "rtph264pay: Restructuring to allow for adding optional caps" 
This reverts commit 61666898cf.

This commit changes what the set_sps_pps() function does, not it doesn't
set caps anymore (and should have been renamed). The main problem is that
not all call sites are updated and thus leak the string.
 2013-05-31 15:18:48 +02:00
Wim Taymans
1516c14881 Revert "rtph264pay/depay: Add frame dimensions a payloaded caps" 
This reverts commit 3dca756a5d.

The H264 RTP spec has no attributes for width and height.
 2013-05-31 15:11:12 +02:00
Wim Taymans
b79d217396 Revert "rtph264pay/depay: Add optional framerate caps for use in SDP" 
This reverts commit d8825e2a5c.

There is no framerate attribute in the h264 RTP spec.
 2013-05-31 15:09:51 +02:00
Wim Taymans
190b3d6688 Revert "rtpjpegpay/depay: Replace framesize caps with width/height" 
This reverts commit 0075d111b4.

Extra application/x-rtp are SDP fields, which are strings.
 2013-05-31 15:08:16 +02:00
Wim Taymans
f870cef8bc Revert "rtpjpegpay/depay: Replace framerate caps field with fraction" 
This reverts commit 9fd25a810b.

We deal with sdp attributes in application/sdp, which are always strings.
 2013-05-31 15:05:51 +02:00
Wim Taymans
25082a50b9 rtspsrc: add extra TLS url protocols 
We also support TLS protocols now.
 2013-05-31 12:34:22 +02:00
Sebastian Dröge
e2e1d1a158 videomixer: Add FIXME comment about the DURATION query from adder 
Currently the code just takes with maximum upstream duration, which
is wrong. It should be the maximum upstream duration in running time.
 2013-05-30 23:56:38 +02:00
Mathieu Duponchelle
5223868caa videomixer: Set a reference to mix->current_caps as the QUERY_CAPS result. 2013-05-30 15:36:48 -04:00
Stefan Sauer
6feaf69bec level: misc cleanups 
Fix some oudated comments. Sort out some confusion of interval_frames and num_frames.
 2013-05-30 17:38:55 +02:00
Stefan Sauer
52282b5faa level: fix discontinuities in timestamps 2013-05-28 19:09:12 +02:00
Wim Taymans
80850df711 rtspsrc: create and push stream-start in TCP mode 2013-05-28 15:45:49 +02:00
Wim Taymans
4fc1f3088b rtspsrc: remove some obsolete code 
It is not needed to do a state change from the _play() function on
ourselves. The state change function already did that and we don't want to
interfere with that (or use hacks to avoid interference).
 2013-05-28 15:10:07 +02:00
Wim Taymans
e6f850996b rtspsrc: set RTCP caps on the RTCP pads 2013-05-28 12:26:25 +02:00
Wim Taymans
63f0ecbbe7 rtpsession: send stream-start and segment events 
Also send stream-start and segment event on the RTCP pad.
We don't need to send anything on the sync_src pad because we
already forwarded all incomming events.
 2013-05-28 12:26:25 +02:00
Wim Taymans
779bcc093c rtspsrc: add signal to handle server requests 
Add a signal to be notified of a server request. The signal handler can then
construct the response message for the server.

See https://bugzilla.gnome.org/show_bug.cgi?id=632207
 2013-05-28 12:26:24 +02:00
Nicolas Dufresne
cd30a81ee3 videomixer: Maintain z-order when new pad are added 
https://bugzilla.gnome.org/show_bug.cgi?id=701109
 2013-05-27 22:43:25 -04:00
Thibault Saunier
7a3df1ab31 videomixer: Always handle flush_stop_pending atomically 
It is not protected with the COLLECT_PADS_STREAM_LOCK anymore
 2013-05-25 12:20:08 -04:00
Thibault Saunier
608bd3e2db videomixer: Do not take COLLECT_PADS_STREAM_LOCK when unnecessary 
Collectpad takes the lock itself when receiving serialized events
and we should not take it for not serialized ones
 2013-05-25 11:03:31 -04:00
Sebastian Dröge
1b5a8ac41c flxdec: Properly skip non-frame chunks 2013-05-24 19:34:05 +02:00
Sebastian Dröge
ae3ee32f42 flxdec: Flush data from adapter after reading it 
Otherwise we're going in an infinite loop, reading the same data
over and over again.
 2013-05-24 19:31:14 +02:00
Andoni Morales Alastruey
a62af107ae goom2k1: fix more duplicated symbols 2013-05-24 09:29:23 +02:00
Sebastian Rasmussen
9fd25a810b rtpjpegpay/depay: Replace framerate caps field with fraction 
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
 2013-05-23 21:05:49 +02:00
Sebastian Rasmussen
0075d111b4 rtpjpegpay/depay: Replace framesize caps with width/height 
The previous implementation had the formatting of SDP attributes happen
in each RTP payloader, now instead the constituent values are propagated
as caps fields. This allows for applications to do SDP offer/answer
based on caps negotiation.

Keep parsing a-framerate, x-framerate and x-dimensions in rtpjpegdepay
to be backwards compatible with previous payloaders.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
 2013-05-23 21:05:43 +02:00
Sebastian Rasmussen
d8825e2a5c rtph264pay/depay: Add optional framerate caps for use in SDP 
This allows for applications to format SDP attributes and still do SDP
offer/answer based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
 2013-05-23 21:04:17 +02:00
Sebastian Rasmussen
3dca756a5d rtph264pay/depay: Add frame dimensions a payloaded caps 
This allows for applications to format SDP attributes and still do SDP
offer/answer based on caps negotiation.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
 2013-05-23 21:04:11 +02:00
Sebastian Rasmussen
61666898cf rtph264pay: Restructuring to allow for adding optional caps 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749
 2013-05-23 21:04:00 +02:00
Sebastian Dröge
e26b8c2832 (dyn|multi)udpsink: Add properties to specify the bind address and port 
By default we use the any addresses and a random port for binding the socket.
 2013-05-23 18:42:09 +02:00
Sebastian Dröge
5b79b8ff3c (dyn|multi)udpsink: Bind socket before using it 
https://bugzilla.gnome.org/show_bug.cgi?id=700878
 2013-05-23 18:05:07 +02:00
Sebastian Dröge
1ed7f7a6a8 (multi)udpsink: Add missing getters for socket-v6 and used-socket-v6 properties 2013-05-23 17:26:31 +02:00
Nicolas Dufresne
d8c5e31657 videomixer: Don't hold stream-lock while pushing non-serialized events 
https://bugzilla.gnome.org/show_bug.cgi?id=700868
 2013-05-23 09:20:04 -04:00
Nicolas Dufresne
a7e0f251ca videomixer: Don't hold object lock while sending events 
https://bugzilla.gnome.org/show_bug.cgi?id=700868
 2013-05-23 09:20:04 -04:00
Sebastian Dröge
ecc6c607ff deinterlace: The return value of gst_pad_set_caps() is not relevant anymore 
Caps can fail to be set because the pad is not linked yet for example.
 2013-05-22 17:34:07 +02:00
David Schleef
318cd39c3e qtdemux: Add error if file has playready drm 2013-05-21 18:21:49 -07:00
Thibault Saunier
18ef4f18d0 videomixer: Send a reconfigure event upstream if sinkpad caps are not usable 
https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2013-05-21 12:15:36 -04:00
Alexander Schrab
a1df050356 mulawdec: Handle NULL buffers in handle_frame 
https://bugzilla.gnome.org/show_bug.cgi?id=698894
 2013-05-21 15:18:04 +02:00
Sebastian Rasmussen
2361567bae rtpjpegpay/depay: Add framesize caps for use in SDP 
The format of the value adheres to RFC6064 and it is meant to be parsed
and included in the SDP sent by gst-rtsp-server to its clients.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
 2013-05-21 09:09:03 +02:00
Sebastian Rasmussen
919eed0787 rtpjpegpay: Add optional framerate caps for use in SDP 
The format of the value adheres to RFC4566 and it is meant to be parsed
and included in the SDP sent by gst-rtsp-server to its clients.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748
 2013-05-21 09:08:21 +02:00
Mathieu Duponchelle
2d3910fc79 videomixer: When all sinkpads are eos, update output segment stop and forward it 
https://bugzilla.gnome.org/show_bug.cgi?id=699793
 2013-05-20 21:06:56 +02:00
Mathieu Duponchelle
521c9a7b5d videomixer: Don't reset the output segment on flush stop 
Only init it when getting from READY to PAUSED, and change it on seek events.

https://bugzilla.gnome.org/show_bug.cgi?id=699793
 2013-05-20 21:03:03 +02:00
Thibault Saunier
86b106091c videomixer: Send caps event from the streaming thread 
This way we avoid races in caps negotiation and we make sure
that the caps are sent after stream-start.

https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2013-05-19 09:28:04 -04:00
Thibault Saunier
718f9004d0 videomixer: Do not send flush_stop when receiving a seek 
There is no reason to send a flush-stop when receiving a seek event.
In the case of a flushing seek, we could eventually want to, but in
the code path were we check if the seek is "flushing", we have the
following comment that makes sense:

"we can't send FLUSH_STOP here since upstream could start pushing data
after we unlock mix->collect.
We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
forwarding the seek upstream or from gst_videomixer_collected,
whichever happens first."

https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2013-05-19 09:28:04 -04:00
Thibault Saunier
85b6852deb videomixer2: Protect flush_stop_pending with the collectpad stream lock 
And make sure to expect a flush-stop after a flush-start

https://bugzilla.gnome.org/show_bug.cgi?id=684237
 2013-05-19 09:28:04 -04:00
Michael Olbrich
d1c56376d6 rtpmp4apay: clear config buffer before using it 
This is necessary because parts of the memory are only modified with "|="

https://bugzilla.gnome.org/show_bug.cgi?id=700514
 2013-05-18 10:57:56 +01:00
Thiago Santos
55caa99ccd qtdemux: Do not expect EOS after a segment event if upstream is mss 
In case qtdemux is handling a mss stream, do not mark the stream to wait
for EOS after a segment. Even if it seems to be the last one according to
the current streams information.

MSS handling is different here because there is another demuxer driving
the pipeline
 2013-05-16 16:50:49 -03:00
Thiago Santos
5517e352ab qtdemux: only set channels and rate if qtdemux knows it 
Setting both of those to 0 is pointless and means that qtdemux
doesn't know the real value. Avoid setting it in this case.
 2013-05-16 16:50:49 -03:00
Arnaud Vrac
6edcc564ba qtdemux: set alac caps using info from codec buffer 
The samplerate field in the STSD atom is not right for some ALAC files
(usually when audio is 96kHz/24bits), so the audio caps must be
extracted from the codec data.

https://bugzilla.gnome.org/show_bug.cgi?id=700382
 2013-05-15 18:42:11 +01:00
Arnaud Vrac
8ed611cdbc avidemux: do not push discont buffers if they aren't discont 
https://bugzilla.gnome.org/show_bug.cgi?id=682110
 2013-05-15 13:16:11 +01:00
Joshua M. Doe
837dcfb363 videocrop: Add support for GRAY16_LE/GRAY16_BE 
https://bugzilla.gnome.org/show_bug.cgi?id=700331
 2013-05-15 09:29:30 +02:00
Sebastian Dröge
41e1af3751 rgvolume: Send all events through the proxypads instead of just sending to the target 
Otherwise the sticky events are missing on the proxypads.
 2013-05-14 17:29:58 +02:00
Sebastian Dröge
4fdbf88a65 matroskaparse: Make sure to send a segment event before dataflow 2013-05-14 13:52:18 +02:00
Sebastian Dröge
5c8bb90262 deinterlace: Improve handling of min/max buffer numbers of the buffer pool 2013-05-14 09:45:12 +02:00
Matej Knopp
30c00f4fb7 deinterlace: set caps for buffer pool config 2013-05-14 09:38:24 +02:00
Olivier Crête
4f0fdabf10 multifilesink: Let the base class do get_times 
This will make sync=TRUE work, the default is still sync=FALSE
 2013-05-13 13:34:22 -04:00
Nicolas Dufresne
f67c227878 interleave: Send stream-start before caps event 2013-05-13 15:37:38 +02:00
Nicolas Dufresne
04c9f43567 rtpmux: Send stream-start before caps 2013-05-13 15:37:05 +02:00
Sebastian Dröge
6dee7d3a06 icydemux: Fix sticky event handling 2013-05-13 15:19:25 +02:00
Sebastian Dröge
9ac456bd43 flvmux: Push sticky events in the right order 2013-05-13 15:06:03 +02:00
Sebastian Dröge
0ab23ef5c9 deinterleave: Fix sticky event handling 2013-05-13 14:54:35 +02:00
Sebastian Dröge
c94fbf6206 deinterleave: Code style fixes 2013-05-13 13:55:44 +02:00
Sebastian Dröge
f28ab45f3e rtpgstpay: First let baseclass handle events, then put them into the stream 
Fixes handling of sticky events.

https://bugzilla.gnome.org/show_bug.cgi?id=700213
 2013-05-13 13:44:35 +02:00
Tim-Philipp Müller
8359b6bff1 multipartdemux: fix example pipeline 
Need jpegparse.
 2013-05-10 14:01:14 +01:00
Nicolas Dufresne
0b737fba0d shapewipe: Can't map twice the same buffer for writing 
I took the opportunity to simplify that code a bit. We now use
gst_buffer_make_writable() to make the buffer writable and map twice the
same buffer, with first map being read/write, and second read only. This
get rid of the critical:

GStreamer-CRITICAL **: gst_structure_set_name: assertion `IS_MUTABLE

https://bugzilla.gnome.org/show_bug.cgi?id=700044
 2013-05-10 09:27:02 +02:00
Nicolas Dufresne
13a5d0304d shapewipe: Ensure caps are writable 
The exist one case where that we endup with original caps in ret, in which
case we are not guaratied to have writable caps. Simply ensure this is the
caps are writable before entering the loop.

https://bugzilla.gnome.org/show_bug.cgi?id=700044
 2013-05-10 09:26:07 +02:00
Nicolas Dufresne
59c2f459de shapewipe: Fix sample pipeline in documentation 
https://bugzilla.gnome.org/show_bug.cgi?id=700044
 2013-05-10 09:26:00 +02:00
Sebastian Dröge
3110b7cc31 Revert "videomixer2: Take into account new segments" 
This reverts commit 84ae670ab4.

Actually this is not how it is supposed to work. videomixer
creates a [0,-1] segment and then puts frames of the different
streams there based on their running times in their own segments.
 2013-05-09 16:26:19 +02:00
Mathieu Duponchelle
84ae670ab4 videomixer2: Take into account new segments 
Also forward the event downstream on the next opportunity.

https://bugzilla.gnome.org/show_bug.cgi?id=699793
 2013-05-09 16:18:54 +02:00
Tim-Philipp Müller
643450c9b8 Revert "gstrtspsrc: set buffer-size for multicast buffers" 
This reverts commit 2481e95d03.

This is already done five lines above, it was added a year
ago in commit 561b131e.
 2013-05-09 09:09:59 +01:00
Nicolas Dufresne
2d53229a86 audiowsinclimit: Frequence property renamed cutoff 
Updating the documentation to reflect this change.

See: https://bugzilla.gnome.org/show_bug.cgi?id=699964
 2013-05-09 08:46:04 +02:00
Aha Unsworth
2481e95d03 gstrtspsrc: set buffer-size for multicast buffers 
For receiving video data via RTSP when the video is sent via
multicast there is no way to specify the udpsrc buffer-size.

On windows the native network buffer is not large and with video
i-frames being huge the buffer is to small and you get i-frame corruption,
it looks terrible, and there is no (easy) way to set the udpsrc buffer-size.

https://bugs.freedesktop.org/show_bug.cgi?id=52264
 2013-05-08 16:57:53 -03:00
Sebastian Dröge
1588cda9a1 videomixer2: Send stream-start before caps event 
https://bugzilla.gnome.org/show_bug.cgi?id=699895
 2013-05-08 16:02:46 +02:00
Thiago Santos
a0e934e72e qtdemux: push new caps events when caps change 
Whenever the demuxer has a new caps on a stream, it should set the
new_caps variable to true and a new caps event will be pushed before
the next buffer
 2013-05-07 19:29:17 -03:00
Thiago Santos
725faab590 qtdemux: do not push discont buffers if they aren't discont 
qtdemux takes its buffers from a GstAdapter. Those buffers are created
from the larger buffer that it obtained from upstream and they carry
the same flags, including DISCONT if it is set. In these cases, all
buffers that qtdemux is going to push would be marked as DISCONT.

This scenario can make parsers/decoders flush on every buffer leading
to no decoding at all hapenning. This patch prevents this by unsetting
the flag if it shouldn't be set.
 2013-05-07 19:29:17 -03:00
Thiago Santos
4d073beeee qtdemux: some code cleanup for mss handling code 
* Explicitly init variables for fragmented formats at init
* Do not use GstClockTime type if the variable isn't a timestamp
* Fix a style/readability issue at an if block
* Group 2 mss mode conditional blocks together to improve readability

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:29:17 -03:00
Thiago Santos
d1b91c755c qtdemux: avoid storing non-time newsegments to push later 
This can confuse downstream when they get a byte segment after receiving
the natural time segment from qtdemux that it sends when starting to
push buffers. This is specially the case with parsers that try to
convert the position from byte to time format and might miss the
correct position for playback to start.
 2013-05-07 19:29:17 -03:00
Thiago Santos
895525b5cb qtdemux: avoid setting fields to non-writable caps 2013-05-07 19:29:17 -03:00
Wim Taymans
544d926732 qtdemux: don't send so many segment events 
Only send one segment event in the beginning of the stream, not
after each moov and moof atom.

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:29:17 -03:00
Wim Taymans
d9cd4fcc17 qtdemux: place incomming timestamps on output 
Place the incomming timestamp (if any) directly onto the outgoing buffers
and interpollate other timestamps.

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:29:17 -03:00
Thiago Santos
cca2f555d1 qtdemux: improve reset of internal status 
Reset different variables on state changes to ready and when
handling a flush-stop. For handling flush stops we should check
if there is an upstream adaptive demuxer driving the pipeline as this
means that qtdemux will get a new moov atom. For 'standard' isomedia
streams this isn't true and qtdemux should keep the previous moov
information around.

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:29:17 -03:00
Thiago Santos
6c69e59677 qtdemux: prepare qtdemux to accept multiple dash moovs in a row 
Whenever dashdemux switches bitrates it sends a new moov with the
new stream configuration. qtdemux should now handle this by splitting
the exposing and configuration of streams into separate functions. When
the stream is new it is configured and exposed, when it is a new bitrate
of an existing stream it is only reconfigured.

Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:25:30 -03:00
Andre Moreira Magalhaes (andrunko)
2a7d3d1598 qtdemux: Move FLUSH_STOP/PAUSED_TO_READY handling to a reset method. 
Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:18:03 -03:00
Louis-Francis Ratté-Boulianne
d499b461da qtdemux: Remove old pads when exposing streams and other general fixes. 
Conflicts:
	gst/isomp4/qtdemux.c
 2013-05-07 19:18:03 -03:00
Thiago Santos
a3c19eeea1 qtdemux: handle mss streams 
smoothstreaming streams should be handled as a special kind of
fragmented isomedia. In MSS the fragments will not contain a
'moov' atom with the media descriptions, this has to be extracted
from the caps.

Additionally, there should be another demuxer upstream that is likely
going to be the one to answer/act on queries and events, so qtdemux has
to forward those upstream.
 2013-05-07 19:18:03 -03:00
Sebastian Rasmussen
9532b04947 rtpgstpay: fix invalid memory access in event handler 
First process event in payloader, then hand it to the
base class which takes ownership of the event.

https://bugzilla.gnome.org/show_bug.cgi?id=699637
 2013-05-04 10:49:23 +01:00
Tim-Philipp Müller
68ac392e8f ac3parse, dcaparse: check buffer size before trimming 
and unref old buffer as soon as possible.
 2013-05-04 10:08:47 +01:00
Andoni Morales Alastruey
3462282b83 dcaparse: add support for "audio/x-private1-dts" 2013-05-03 13:44:23 +02:00
Andoni Morales Alastruey
4531381541 ac3parse: add support for "audio/x-private1-ac3" 2013-05-03 13:44:23 +02:00
Andoni Morales Alastruey
4a78a77e65 rtp: fix duplicated symbols with libvpx 2013-05-02 14:03:33 +02:00
Andoni Morales Alastruey
584fdbad84 goom2k1: fix duplicated symbols with goom 2013-05-02 14:03:26 +02:00
Sebastian Dröge
ae05ed4f05 rtph264pay: If the adapter is empty on EOS don't try to map its content 
https://bugzilla.gnome.org/show_bug.cgi?id=699314
 2013-05-01 15:49:45 +02:00
Ognyan Tonchev
0584d5c4c9 matroskademux: add stream-format=raw to aac caps 
https://bugzilla.gnome.org/show_bug.cgi?id=699303
 2013-05-01 15:47:15 +02:00
Tim-Philipp Müller
7ccb387e85 udp: log WARNING debug message if UDP multicast is likely to be broken 2013-04-27 11:25:12 +01:00
Tim-Philipp Müller
4273eccace udpsrc: add includes to get socklen_t defined on Windows 
https://bugzilla.gnome.org/show_bug.cgi?id=692400
 2013-04-27 11:16:54 +01:00
Yury Delendik
4bc06859d1 qtdemux: add support for VP6F VP6 flash codec 
https://bugzilla.gnome.org/show_bug.cgi?id=699010
 2013-04-27 09:39:45 +01:00
Edward Hervey
3e5ad52c0d monoscope: Fix debug statement 2013-04-26 12:16:49 +02:00
Alexander Schrab
3ec9673dfc mulawdec: change base class to GstAudioDecoder 
https://bugzilla.gnome.org/show_bug.cgi?id=698894
 2013-04-26 08:46:34 +02:00
Mathieu Duponchelle
6b153ce385 videomixer: send stream-start event. 2013-04-25 16:09:34 -03:00
Wim Taymans
1df2e623b5 docs: add some pay/depayloaders 
See https://bugzilla.gnome.org/show_bug.cgi?id=551631
 2013-04-25 14:05:55 +02:00
Sebastian Dröge
fb0384fa0d mulaw: Some minor memleak fixes and cleanup 2013-04-25 12:44:15 +02:00
Alexander Schrab
f0edb5fb70 mulawenc: change to gstaudioencoder base, added bitrate tags 2013-04-25 12:36:15 +02:00
Sebastian Dröge
b1af93f791 (multi)udpsink: Use separate sockets for IPv4 and IPv6 
https://bugzilla.gnome.org/show_bug.cgi?id=534243
 2013-04-25 12:12:23 +02:00
Sebastian Dröge
0b552150ce dynudpsink: Use separate sockets for IPv4 and IPv6 
https://bugzilla.gnome.org/show_bug.cgi?id=534243
 2013-04-25 12:09:27 +02:00
Sebastian Dröge
ed8ea46424 udp: Don't include removed gstudp.h in noinst_HEADERS 2013-04-25 10:43:56 +02:00
Sebastian Dröge
afb284e3a9 udp: Remove unused enum type 2013-04-25 09:16:14 +02:00
Sebastian Dröge
a957457cc1 udp: Use the generic marshaller instead of generating marshallers 2013-04-25 09:13:51 +02:00
Sebastian Dröge
07d3363436 udpsrc: Rename instance variable from host to multi_group 
This is more consistent as it's used for the multicast-group property.
 2013-04-25 09:07:41 +02:00
Sebastian Dröge
427673d283 udpsrc: Add bind-address property 
This is equivalent to multicast-group currently for backwards compatibility.
In 2.0 this should be handled separately, the former only being the multicast
group and the latter always being the address the socket is bound to, even if
a multicast group is given.
 2013-04-25 09:05:12 +02:00
Wim Taymans
5ba3fd3c63 vrawdepay: return output buffer from process 
Return the output buffer from the process function instead of pushing
it ourselves. This way, the subclass can actually deal with the return
value of the push.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693727
 2013-04-24 16:24:25 +02:00
Wim Taymans
eac9efb92e rtp: a marker bit should translate to RESYNC 
A marker bit on an audio packet does not mean a DISCONT (in the GStreamer sense
of missing data) but it means that the packet is the end of a talkspurt and thus
a good opportunity to resync to the clock. Use the RESYNC buffer flag to note
this.
Real discontinuities are marked with DISCONT still when the seqnum has a GAP or
when the input buffer has the DISCONT flag set.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=627204
 2013-04-24 15:42:45 +02:00
Sebastian Dröge
fdb667ae00 rtpjpegdepay: Drop frame if it's less than 2 bytes large 
https://bugzilla.gnome.org/show_bug.cgi?id=677560
 2013-04-22 10:19:29 +02:00
Sreerenj Balachandran
504360fe36 autodetect: use _plugin_feature_rank_compare API instead of duplicating the code. 2013-04-18 14:00:06 +02:00
Olivier Crête
24bb263d54 videomixer: Don't unref query, we don't own it 
Fixes double-unref bug. Bug found by Youness Alaoui
 2013-04-16 19:29:48 -04:00
Sebastian Dröge
b0b0557c48 gst: Add better support for static plugins 2013-04-15 15:54:11 +02:00
Andoni Morales Alastruey
2ea9a66dd5 goom2k1: fix duplicated symbol with goom 2013-04-15 08:43:05 +02:00
Wim Taymans
9d7519f66e rtp: register tag image types 
The rtpgstdepay needs the type to be available in order to deserialize the
event.
 2013-04-12 16:18:42 +01:00
Wim Taymans
b1f4587d75 rtpgstdepay: handle event parse failures better 2013-04-12 16:18:42 +01:00
Anton Belka
b959d827be wavenc: add TOC setter support 2013-04-12 14:35:47 +02:00
Stefan Sauer
f4577ff492 wavenc: small cleanups for toc handling 
Don't add empty labl/note chunks. Always pass instance as the first param. Add more logging.
 2013-04-12 14:35:47 +02:00
Sebastian Dröge
b17750ed9e rtspsrc: Proxy the ntp-sync property of rtpbin 2013-04-12 12:58:50 +02:00
Sebastian Dröge
53dae1585e rtspsrc: Give the manager always the name "manager" 
This allows to use the GstChildProxy interface to adjust
properties on it.
 2013-04-12 12:51:05 +02:00
Anton Belka
bda2703e88 wavenc: add 'note' chunk support 2013-04-11 20:47:18 +02:00
Wim Taymans
f8013487c9 rtspsrc: add support for NetClientClock 
When the server suggests a GstNetTimeProvider in the SDP, set up a
GstNetClientClock that slaves to the remote clock and suggest this clock in
provide_clock.
 2013-04-11 15:00:05 +01:00
Wim Taymans
f96aa414e1 udpsink: avoid alloc and free in render function 
Avoid doing alloc and free in the render function for each buffer. Instead,
allocate the needed arrays in _init and use those.
 2013-04-11 14:57:11 +01:00
Stefan Sauer
48b9919e31 waveparse: remove superfluous g_list_first() calls 
The variables already point to the start of the list.
 2013-04-10 14:25:24 +02:00
Andreas Fenkart
20d3ec8810 rtpsbcdepay: fix sbc frame length calculation for mono and stereo modes 
https://bugzilla.gnome.org/show_bug.cgi?id=697463
 2013-04-09 23:17:57 +01:00
Anton Belka
5ae92ce770 wavparse: add 'note' chunk support 
Add 'note' chunk support in TOC as GST_TAG_COMMENT

https://bugzilla.gnome.org/show_bug.cgi?id=696549
 2013-04-09 22:58:27 +02:00
David Schleef
a55ccff854 qtdemux: check value inside enda to set endianness 2013-04-09 13:30:17 -07:00
Wim Taymans
ece73b786a icydemux: avoid copy when we can 2013-04-09 17:34:12 +02:00
Wim Taymans
91a3afc4dc gstpay: use bufferlist to avoid memcpy 2013-04-09 16:53:31 +02:00
Wim Taymans
3d7d757521 udpsink: improve debug 2013-04-09 16:53:31 +02:00
Alexander Schrab
79d5a7d03c wavparse: error out if we receive eos before any valid data 
https://bugzilla.gnome.org/show_bug.cgi?id=696684
 2013-04-09 00:27:31 +01:00
Matej Knopp
67c2219687 deinterlace: force deinterlacing in "interlaced" mode 
https://bugzilla.gnome.org/show_bug.cgi?id=697467
 2013-04-07 20:48:21 +01:00
Nicola Murino
c41c16424d rtpsbcdepay: fix printf format compiler warnings 
https://bugzilla.gnome.org/show_bug.cgi?id=697343
 2013-04-05 13:50:19 +01:00
Stefan Sauer
b79f667ef4 level: resync on discont 
Drop pending data on discont and start a new cycle with a new base timestamp.
Cleanup some variables.
 2013-04-04 22:49:49 +02:00
Olivier Crête
f8831c0cd2 rtpsbcdepay: Rank as secondary 
This way, it will be selected by decodebin
Bug reported by andreas.fenkart@streamunlimited.com

https://bugzilla.gnome.org/show_bug.cgi?id=697227
 2013-04-03 18:25:36 -04:00
Stefan Sauer
2e56032031 level: subdivide buffers for sample accurate interval handling 
Previously we would skip level message when processing buffers > the requested
interval. Also the message frequency would contain quite some jitter due to only
considering them at the end of buffers.

Cleanup the tests while we're at it.
 2013-04-03 21:40:17 +02:00
Stefan Sauer
b062171dda spectrum: remove old since comment 2013-04-03 20:30:08 +02:00
Sebastian Dröge
d80ff8e7f3 rtspsrc: Proxy the multicast-iface property of udpsrc 2013-04-03 17:53:13 +02:00
Olivier Crête
6f3734c305 rtpssrcdemux: Only forward stick events while holding the sinkpad stream lock 
Otherwise we get a race where if the RTCP packet comes in first and while
it is added the pads, the segment event arrives on the RTP stream, the event
may be lost completely and never forwarded.
 2013-04-02 23:42:42 -04:00
Olivier Crête
76679f9ae9 rtpssrcdemux: No need to explicitely forward the caps 
They are forwarded with the other events
 2013-04-02 23:42:41 -04:00
Olivier Crête
4ad8693f3c rtpssrcdemux: Remove unused GstSegment 2013-04-02 23:42:41 -04:00
Olivier Crête
7293b0eff7 rtpssrcdemux: Simplify event forwarding 
Use the gst_pad_forward() mechanic, this way we won't miss pads that are
added while we are pushing
 2013-04-02 23:42:41 -04:00
Olivier Crête
f4c3aef13a rtpssrcdemux: Don't cross the internal links 
We had the wrong condition to check for the internal links, so RTP and RTCP
pads got crossed!
 2013-04-02 23:42:41 -04:00
Tim-Philipp Müller
078ff16abe matroskademux: fix some debug messages 2013-04-03 00:49:37 +01:00
Arnaud Vrac
00b46b4744 matroskademux: handle TrueHD audio codec id 
https://bugzilla.gnome.org/show_bug.cgi?id=697113
 2013-04-02 22:47:54 +01:00
Wim Taymans
ac2bcfa833 theorapay: add delta-unit to output frames 2013-03-31 19:14:04 +02:00
Matej Knopp
5686512b77 qtmux: use timestamp delta as duration if possible 
https://bugzilla.gnome.org/show_bug.cgi?id=696437
 2013-03-30 15:18:45 -07:00
Josep Torra
509631f60b rtp: fixes debug message printf related compiler warnings in SBC depayloader 2013-03-30 09:44:41 +01:00
Arun Raghavan
87bdad4bfc rtp: Add an rtpsbcdepay element 
Pretty straightforward - takes SBC encapsulated in RTP, depayloads, and
pushes out SBC buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-03-28 17:22:33 +00:00
Tim-Philipp Müller
477cc51fe7 rtp: fix SBC payloader 
Init RTP buffer on stack correctly, so mapping it works
without criticals and the payloader actually works.
 2013-03-27 22:18:34 +00:00
David Schleef
53f8b05b08 Use %03u for format in gst_pad_create_stream_id_printf() 2013-03-25 18:57:08 -07:00
Sebastian Dröge
56062768af capssetter: Prevent unneeded caps copying and allocation 2013-03-25 10:12:03 +01:00
Dirk Van Haerenborgh
766c5b22ed capssetter: Pass any or filter caps upstream 
capsetter accepts anything and just forwards different caps,
as such it should return ANY caps on the sinkpad.

https://bugzilla.gnome.org/show_bug.cgi?id=693005
 2013-03-25 10:11:32 +01:00
Tim-Philipp Müller
35769f7c5d wavparse: expose CUE sheet items as tracks not chapter entries in TOC 
https://bugzilla.gnome.org/show_bug.cgi?id=677306
 2013-03-24 17:55:55 +00:00
Tim-Philipp Müller
163a7afa1a wavenc: add some example pipelines 2013-03-23 12:59:26 +00:00
Anton Belka
e808173483 wavenc: add TOC support 
https://bugzilla.gnome.org/show_bug.cgi?id=680998
 2013-03-23 12:55:08 +00:00
Matej Knopp
f29e62c131 qtdemux: make empty subtitle buffer recognition more robust 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-23 11:24:23 +00:00
David Schleef
c0443a17c4 qtmux: Fix test regression with one buffer streams 2013-03-22 15:14:15 -07:00
David Schleef
5bd2864101 qtdemux: split large raw audio samples 
In order to deal with a file that has samples that are 24 seconds
long.  Seeking still doesn't work with such files.
 2013-03-22 14:14:05 -07:00
David Schleef
364433c105 qtmux: Remove documentation for dts-method 2013-03-22 14:14:04 -07:00
David Schleef
6571e388be qtmux: deprecate dts-method property 2013-03-22 14:14:04 -07:00
David Schleef
ee56a7cb99 qtmux: Fix problems causing bad durations in file 
- Fix up out-of-order incoming DTS values.
- Fix duration of initial sample.
 2013-03-22 14:14:04 -07:00
David Schleef
816e186029 qtmux: fix all timestamps once first_ts is determined 2013-03-22 14:14:04 -07:00
David Schleef
258c40c6dd qtmux: Use PTS/DTS from incoming buffers 
Remove old DTS guessing code.
 2013-03-22 14:14:04 -07:00
Nicola Murino
709f05234f qtmux: expose mulaw caps 
https://bugzilla.gnome.org/show_bug.cgi?id=696052
 2013-03-22 20:08:06 +00:00
Rodolfo Schulz de Lima
874808fd2c qtdemux: fix sample leak when processing private qt tags 
https://bugzilla.gnome.org/show_bug.cgi?id=696355
 2013-03-22 08:47:17 +00:00
Matej Knopp
d8ac666137 qtmux: set stream language code from tag 
https://bugzilla.gnome.org/show_bug.cgi?id=696358
 2013-03-22 08:40:26 +00:00
Matej Knopp
49d9050e9a qtdemux: send GAP events for subtitle streams 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:37 +00:00
Matej Knopp
516a0b8acb qtdemux: ignore empty subtitle buffers 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:34 +00:00
Matej Knopp
f494635126 qtdemux: recognize SBTL subtype for subtitles 
https://bugzilla.gnome.org/show_bug.cgi?id=696244
 2013-03-21 10:03:14 +00:00
Anton Belka
0f97b6f978 flacparse: add support for the toc-select event 
Select tracks from the CUE sheet by sending a toc-select
event based on the uid in the TOC.

https://bugzilla.gnome.org/show_bug.cgi?id=540891
 2013-03-21 00:38:48 +00:00
Michael Smith
b85c5f236b mp4mux: in faststart mode, don't output up to 4 kB of garbage at the end. 2013-03-19 18:09:31 -07:00
Tim-Philipp Müller
5240b7453c sbcparse: pack multiple frames into one output buffer 
Don't output a single buffer for every tiny SBC frame
 2013-03-20 00:35:17 +00:00
Kishore Arepalli
288e05c99d deinterlace: fix infinite loop on EOS with non-default methods or fields 
Fixes problem of infinite loop in gst_deinterlace_reset_history.
Last field in the history was never deinterlaced because idx becomes negative.

Happens e.g. with method=scalerbob fields=bottom or
method=greedyl fields=top

https://bugzilla.gnome.org/show_bug.cgi?id=695644
https://bugzilla.gnome.org/show_bug.cgi?id=693173
 2013-03-17 14:47:26 +00:00
Tim-Philipp Müller
dfde4179e8 avimux: change raw video caps order so that GRAY8 is last 
People like colours.

https://bugzilla.gnome.org/show_bug.cgi?id=695543
 2013-03-12 00:16:18 +00:00
Ognyan Tonchev
3f8ad30cee rtph264pay: Don't use upstream caps with peer_query_caps () 
Calling gst_pad_peer_query_caps () on the src pad with the caps
upstream can produce as a filter from gst_rtp_h264_pay_getcaps ()
is wrong and makes caps negotiation fail if upstream caps are not
NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=695629
 2013-03-11 16:55:13 -04:00
Dirk Van Haerenborgh
065bdf5925 avimux: support raw BGR 
https://bugzilla.gnome.org/show_bug.cgi?id=695543
 2013-03-11 14:51:00 +01:00
Dirk Van Haerenborgh
d7743cf7c6 avidemux: support raw video with negative height 
https://bugzilla.gnome.org/show_bug.cgi?id=695541
 2013-03-11 14:23:46 +01:00
Tim-Philipp Müller
694dbcc5a0 dtmf: move dtmf plugin from -bad to -good 
https://bugzilla.gnome.org/show_bug.cgi?id=687416
 2013-03-09 01:18:30 +00:00
Tim-Philipp Müller
a4c5aa38ec Merge branch 'dtmf-moved-from-bad' 
https://bugzilla.gnome.org/show_bug.cgi?id=687416
 2013-03-09 00:30:38 +00:00
Sebastian Dröge
539126c097 matroska: Include config.h, it's needed for _stdint.h 2013-03-03 11:59:31 +01:00
Sebastian Dröge
1810786083 flacparse: Fix (wrong) use of uninitialized variable compiler warning 2013-03-03 11:53:04 +01:00
Tim-Philipp Müller
677bfecc6f qtdemux: add variant field to H.263 caps 
avdec_h263 won't get plugged otherwise.
 2013-03-02 13:59:52 +00:00
Arnaud Vrac
1cff6427f1 qtdemux: skip disabled tracks 
ISO/IEC 14496-12 specifies disabled tracks should be completely
ignored, so just do it.

Avoids deadlock during prerolling for some files.

Also prevents 'chapter' subtitle tracks from showing up.

https://bugzilla.gnome.org/show_bug.cgi?id=693993
https://bugzilla.gnome.org/show_bug.cgi?id=628790
 2013-03-02 13:54:23 +00:00
Stefan Sauer
15a81baea5 spectrum: remove the since doc-comment from 0.10 2013-02-28 09:43:12 +01:00
Stefan Sauer
b62cb3edcd level: add a "post-messages" property and deprecate "message" 
In spectrum this was changed from 0.10 to 1.0, lets do this here too.
 2013-02-28 09:43:12 +01:00
Olivier Crête
df5ca83baf rtpmp4gdepay: streamtype is not put by all RTSP server, not make it optional 
Specific case here is Wowza 3.5.0
 2013-02-26 14:19:10 -05:00
Thomas Vander Stichele
df8f5f2f83 level: put back deprecation warnings 2013-02-25 00:35:58 +01:00
Thomas Vander Stichele
52b7aab711 level: send last message on EOS 2013-02-25 00:19:22 +01:00
Mark Nauwelaerts
56e2767c20 avidemux: push mode: handle some more 0-size buffer cases 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684944
 2013-02-24 19:28:07 +01:00
Tim-Philipp Müller
8004ae0369 matroskamux: fix up example pipeline in docs 2013-02-23 18:50:52 +00:00
Paul HENRYS
10802cae73 rtpsession: Fix wrong code organisation in case of collision 
change_ssrc field of RTPSession should be set before calling
rtp_session_schedule_bye_locked () as this function will call reconsider function
that will wake up rtcp_thread which will call rtp_session_on_timeout () that will
check change_ssrc to change the ssrc.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184
 2013-02-22 09:28:07 +02:00
Jean-François Fortin Tam
f5cb19e287 alpha: improve descriptions of chroma keying-related properties and enums 
https://bugzilla.gnome.org/show_bug.cgi?id=694374
 2013-02-22 00:09:56 +00:00
Youness Alaoui
a65fd146f8 alpha: Do not override the method with custom r/g/b values 
Depending on the order g_object_set() calls aare made, the
target r/g/b settings will override the method if set to
green/blue. Change that so we do not use the target-r/g/b values
unless the method is set to custom.

https://bugzilla.gnome.org/show_bug.cgi?id=694374
 2013-02-22 00:04:51 +00:00
Ognyan Tonchev
42d8b96f2d auparse: do not leak src_caps 
https://bugzilla.gnome.org/show_bug.cgi?id=694275
 2013-02-21 19:31:59 +00:00
Wim Taymans
a61055809f rtpsession: only delay RTCP when we are a sender 
Only delay the RTCP thread when we are a sender, which we can know because we
have a send_rtp_src pad. Otherwise we might delay the RTCP thread if we
are only a receiver and then there is no code path that wakes up the
RTCP thread and we end up without RTCP packets.
 2013-02-20 21:07:41 +02:00
Tim-Philipp Müller
5b19be933b qtdemux: fix up dodgy code that tries to fix up a broken moov atom 
After gst_buffer_new_and_alloc() gst_buffer_copy_into() will likely
append to the already-existing memory instead of filling it.
 2013-02-18 20:04:05 +00:00
Tim-Philipp Müller
34b81f7c93 qtdemux: fix potential crash on short MOOV atom 
Don't unmap short MOOV atom buffer twice, which happened
in the case where we don't fix up the MOOV atom.

Fixes crashes when thumbnailing partial mp4 file where
the MOOV atom is still incomplete.

https://bugzilla.gnome.org/show_bug.cgi?id=694010
 2013-02-18 16:35:08 +00:00
Stefan Sauer
99f84b8c4c audiopanorama: remove channel-mask from caps 
The channel-mask is only needed for channels>2 which we don't do.
 2013-02-15 21:30:15 +01:00
Tim-Philipp Müller
01c6512d5f udpsrc: use g_socket_set_option() to set buffer size with newer GLib versions 
So we have to worry less about portability.

https://bugzilla.gnome.org/show_bug.cgi?id=692400
 2013-02-15 14:11:36 +00:00
Sebastian Dröge
a7ddbc03fe rtp-payloading: Fix unit test caps and AMR depayloader sink template caps 
Fields were missing from the actual caps, or too many fields
existed in the template caps.
 2013-02-13 12:02:46 +01:00
Michael Smith
e3430b0d07 qtdemux: extract codec_data for ProRes 2013-02-12 13:19:53 -08:00
Tim 'mithro' Ansell
c499a81848 avimux: Fixing buffer leak in gst_avi_mux_do_buffer 
gst_avi_mux_do_buffer was leaking data from gst_collect_pads_pop.
 2013-02-12 10:09:05 +01:00
Mark Nauwelaerts
bf81dce432 avidemux: correct duration for audio VBR buffers in pull mode 2013-02-10 15:10:32 +01:00
Mark Nauwelaerts
f0645b79c5 avidemux: proper position reporting and push mode timestamping 
... and align current_total semantics in push and pull mode,
which tracks bytes for CBR and blocks for VBR.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691481
 2013-02-08 21:41:55 +01:00
Wim Taymans
2d5319c1fa rtpsession: delay RTCP until first RTP packet 
Delay sending the first RTCP packet until we have sent the first RTP packet.
Otherwise we will send out a Receiver Report instead of a sender report.

See https://bugzilla.gnome.org/show_bug.cgi?id=691400
 2013-02-08 17:05:27 +01:00
Wim Taymans
2971ed44ee rtpsession: remove dead code 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355
 2013-02-07 15:06:40 +01:00
Paul HENRYS
0e91c949d8 rtpptdemux: forward sticky events and then set caps 
When a new src pad is added, first forward the sticky events and then
set the caps on the src pad

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786
 2013-02-07 14:38:20 +01:00
Markovtsev Vadim
7cebe2fc41 rtpjitterbuffer: improve debug output 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688935
 2013-02-07 14:32:26 +01:00
Wim Taymans
978cc9f538 rtpbin: rework cleanup of streams 
Move the work of cleaning up the client streams in the free_stream
function. This allows us to properly clean up the client streams when we
remove an RTP stream as well.

Based on patch by Sujay <sdatar@cisco.com>

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156
 2013-02-07 13:02:34 +01:00
Tim 'mithro' Ansell
3a5d17e852 videomixer2: avoid caps leak 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693307
 2013-02-07 11:40:35 +01:00
Wim Taymans
c3077012c0 jitterbuffer: do skew estimation only for new timestamps 
Only run the skew estimation code when we have a new RTP timestamp. If we have
the same RTP timestamp, we simply use the previous estimation. This works
because the new observation with the same RTP timestamp has to have a bigger
receiver time and is thus not going to influence the estimation except for
causing more jitter.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023
 2013-02-06 17:15:11 +01:00
Wim Taymans
640de61740 rtspsrc: only EOS when our source sends BYE 
Only EOS when we receive a BYE event from the SSRC of our stream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453
 2013-02-06 14:01:16 +01:00
Wim Taymans
0540492ab2 rtspsrc: save the stream SSRC 
Conflicts:
	gst/rtsp/gstrtspsrc.c
 2013-02-06 14:00:56 +01:00
Wim Taymans
c8fb1c720c rtspsrc: flush connection when stopping 
When we stop, we can flush all pending commands so that we can stop and
join the task.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924
 2013-02-06 13:18:18 +01:00
Stefan Sauer
96f8775a0d spectrum: remove outdates readme 
Lets remove the readme from pre-0.1.0 that is completely irrelevant now.
 2013-02-05 22:02:13 +01:00
Stefan Sauer
86ae581928 audiopanorama: add more debug logging 2013-02-05 18:51:27 +01:00
Rico Tzschichholz
682e49a752 audioparsers: fix typo in noinst_headers 2013-02-04 18:38:41 +00:00
Stefan Sauer
1f1fe47cb6 audiopanorama: further port to 1.0 
Transformcaps is not called with caps containing single structures anymore. Also add missing filter handling. Still does not negotiate though.
 2013-02-04 11:08:23 +01:00
Stefan Sauer
d187b96ee2 audiopanorama: fix caps 
We don't turn float into 32bit pcm. Looks like a typo from updating the caps.
 2013-02-03 22:45:52 +01:00
Olivier Crête
fe3e535853 level: Add missing coma between formats 2013-02-03 13:14:50 +01:00
Matthew Waters
b9151a9c28 videomixer: fix eos timestamp check 
fixes hang in videotestsrc num-buffers=20 ! videomixer ! fakesink

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692935
 2013-01-31 16:45:38 +01:00
Dirk Van Haerenborgh
18ff57d6b3 avimux: add support for raw monochrome 8-bit video 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692932
 2013-01-31 13:00:17 +01:00
Wim Taymans
747447d298 rtpsession: avoid '...is used uninitialized' 2013-01-29 10:32:51 +01:00
Youness Alaoui
f6a00ad6e9 qtdemux: set interleaved layout correctly for LPCM audio 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:44:01 +00:00
Youness Alaoui
a76524ea08 qtdemux: add support for LPCM fourcc (uncompressed audio in Quicktime7) 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:57 +00:00
Youness Alaoui
69b814546a qtdemux: print all debug for sound sample description v2 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:49 +00:00
Youness Alaoui
92ff8a9b09 qtdemux: sound sample description v2 doesn't override samples_per_packet 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:42 +00:00
Youness Alaoui
ee3d9cbd98 qtdemux: pass stsd data to qtdemux_audio_caps() 
We will need that later for LPCM format support. Disable
QDM2 parsing of stsd data which dead code before as well
because data was always NULL.

https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:38 +00:00
Youness Alaoui
6d3ff78575 qtdemux: add len check for sound sample descriptions v1 and v2 
https://bugzilla.gnome.org/show_bug.cgi?id=663458
 2013-01-28 23:43:28 +00:00
Tim-Philipp Müller
629772f735 rtpmanager: use C89-style comments 2013-01-28 23:07:34 +00:00
Olivier Crête
451217c437 gstrtpsession: Fix double-declared variable 2013-01-28 18:06:15 -05:00
Olivier Crête
7300d489fe rtp: Fix compilation errors in previous patches 2013-01-28 17:58:20 -05:00
Haakon Sporsheim
86c13ceae6 rtpsession: Ensure MT safe event handling and plug event leak. 
https://bugzilla.gnome.org/show_bug.cgi?id=667826
 2013-01-28 17:44:31 -05:00
Idar Tollefsen
268c998a32 rtpsession: mt-safe event-push 
By taking a ref of the sink-pad under lock, it won't dissappear
while the push is taking place

https://bugzilla.gnome.org/show_bug.cgi?id=667816
 2013-01-28 17:34:50 -05:00
Pascal Buhler
f459fe2673 rtpssrcdemux: Safely push on pads that might be removed due to a RTCP BYE 
https://bugzilla.gnome.org/show_bug.cgi?id=667815
 2013-01-28 17:01:27 -05:00
Tim-Philipp Müller
721dd1ab26 sbcparse: init some variables to avoid bogus compiler warnings 2013-01-28 11:58:50 +00:00
Wim Taymans
4397c8ffbf rtpdepay: remove payload type restrictions 
Remove the pt restrictions for all the depayloaders that have an
encoding-name. We can use this to autoplug decoders.
Remove the encoding-name for all the payloaders with a fixed payload
type.
We now either have an encoding-name or a pt in the sinkpad caps of
a depayloader.

See https://bugzilla.gnome.org/show_bug.cgi?id=639292
 2013-01-28 12:41:04 +01:00
Marc Leeman
bab2f3c92b rtp: remove payload requirements from selected depayloaders 
encoding name is required in the caps and is a better fit for autoplugging than
the pt value. Hardware manufacturers have a bad habit of skimming through RFCs
and in this case; use unassigned numbers for encoders instead of dynamic
numbers.

In essence, this patch will add support for a lot of Bosch hardware encoders
without breaking autoplugging.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292
 2013-01-28 12:23:41 +01:00
Mark Nauwelaerts
a1a579afeb qtdemux: push mode: only parse moov 1 once 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691570
 2013-01-27 12:54:20 +01:00
Tim-Philipp Müller
47fccbe635 rtpdtmfsrc: fix compiler warning 
gstrtpdtmfsrc.c: In function 'gst_dtmf_src_prepare_message.isra.1':
gstrtpdtmfsrc.c:669:3: error: 's' may be used uninitialized in this function
 2013-01-26 22:58:29 +00:00
Olivier Crête
db5c3f4048 rtpdtmfdepay: Fix missing work in doc 2013-01-25 21:06:05 -05:00
Olivier Crête
92f9a9d9ff rtpdtmfsrc: Post the messages after the clock wait 
This way, the messages will be closer in time to when the packets are sent out
 2013-01-25 20:45:43 -05:00
Olivier Crête
0d316b4f43 rtpdtmfsrc: Only set the duration when starting to send 
The duration depends on the clock rate, which could change due to renegotiation
 2013-01-25 20:45:43 -05:00
Olivier Crête
90497aa3cd rtpdtmfsrc: remove "ssrc" from caps 
ssrc is uint and we don't have a uint range type
 2013-01-25 20:45:43 -05:00
Tim-Philipp Müller
d62019fff2 qtmux: set language to 'undefined' instead of English by default 2013-01-24 21:08:51 +00:00
Mark Nauwelaerts
0777a600e3 audioparsers: sbc: fix bogus compiler warning 
gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame':
gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i
 2013-01-22 19:26:09 +01:00
Thijs Vermeir
16128f0234 autoparsers: use appropriate printf format for gsize 2013-01-16 14:32:56 +01:00
Tim-Philipp Müller
9455a3aee1 rtpsbcpay: update some fields in the caps to their new name 
and to match the parser. "mode" got renamed to "channel-mode"
and "allocation" to "allocation-method".
 2013-01-16 10:19:36 +00:00
Tim-Philipp Müller
9f7a949773 audioparsers: add SBC audio parser 
From-scratch rewrite, the bluez one was useless and broken.

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-01-15 17:45:30 +00:00
Tim-Philipp Müller
39ef892938 rtp: import rtpsbcpay from bluez and port to 1.0 
Compiles, but not tested yet (sbc elements still need to be ported).

https://bugzilla.gnome.org/show_bug.cgi?id=690582
 2013-01-10 12:43:50 +00:00
Olivier Crête
c6dea5d09c dtmf/spandsp: Move dtmfdetect to use libspandsp 
Remove our copy of the tone_detect.c file and use the original
from libspandsp. Also move the element to the spandsp plugin.
 2013-01-09 20:05:16 -05:00
Marcel Holtmann
4196feb659 rtpsbcpay: Remove workaround for compiler warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
fe79c60d74 rtpsbcpay: Add pragma based workaround for GStreamer warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
08e95e7249 rtpsbcpay: Update copyright information 2013-01-10 00:15:36 +00:00
Marcel Holtmann
7fa03c0076 rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin 2013-01-10 00:15:35 +00:00
Marcel Holtmann
27a6b0abfe rtpsbcpay: Update copyright information 2013-01-10 00:15:35 +00:00
Marcel Holtmann
f890079aae rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup) 2013-01-10 00:15:35 +00:00
Johan Hedberg
7d4f846112 rtpsbcpay: More coding style fixes 2013-01-10 00:15:35 +00:00
Luiz Augusto von Dentz
151ad9b28d rtpsbcpay: Remove possible extra memcpy for gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
69c8374b7c rtpsbcpay: Fix bug sending empty packages and remove a buffer copy. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
7b3e4356ea rtpsbcpay: Fix runtime warnings of gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
f74f061f3b rtpsbcpay: Update gstreamer plugin to use new sbc API. 2013-01-10 00:13:14 +00:00
Marcel Holtmann
b9be04f07b rtpsbcpay: Update copyright information 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
687400ecf4 rtpsbcpay: Fixes gstreamer caps and code cleanup. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
a4f9624261 rtpsbcpay: Fix gtreamer payloader sending fragmented frames. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
41e2f4f544 rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
96971cd323 rtpsbcpay: Make a2dpsink to act like a bin and split the payloader. 2013-01-10 00:13:14 +00:00
Wim Taymans
72402cc649 rtp: small improvements 2013-01-08 16:27:42 +01:00
Wim Taymans
af055d9574 jitterbuffer: refactor handle sync code 
Move the code that combines the last SR packet and the current jitterbuffer sync
values into a sync structure, into its own function. We want to reuse this bit
later.
 2013-01-07 15:50:33 +01:00
Wim Taymans
87f7d6b9bf rtp: include downstream latency in SR calculations 
When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
 2013-01-07 15:45:10 +01:00