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7614 commits

Author SHA1 Message Date
Wim Taymans
fde438791e rtpjitterbuffer: small debug improvement 2013-09-30 12:30:23 +02:00
Wim Taymans
6e7d547be4 rtpjitterbuffer: reset skew does not reset clock-rate 
Don't reset the clock-rate when we reset the skew correction algorithm.
Reset the skew correction algorithm when we change the clock-rate.
 2013-09-30 11:53:08 +02:00
Wim Taymans
03d520eb69 rtpjitterbuffer: pause timer when PAUSED 
Also pause the timer when we go to the PAUSED state. It is possible that we
don't have a clock or base-time in PAUSED to perform the timeouts.
 2013-09-30 11:16:32 +02:00
Wim Taymans
4a31aec513 rtpjitterbuffer: improve debug 2013-09-30 11:15:25 +02:00
Hans Månsson
041946423a mp4mux: Do not require framerate in peer video caps 
Remove the framerate restriction on the caps.

Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708864
 2013-09-28 13:02:11 +02:00
Wim Taymans
8c5ce0dbdc rtspsrc: also go into the loop function after connect 
When we have opened the stream, go into the loop function so that we can
receive messages from the server.
 2013-09-27 15:08:31 +02:00
Matej Knopp
40c0586c17 matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8 
https://bugzilla.gnome.org/show_bug.cgi?id=707933
 2013-09-27 14:38:19 +02:00
Wim Taymans
d4b4b4e924 rtpjitterbuffer: don't calculate skew without rtptime 
Skip trying to calculate the skew when we don't have an rtptime.
It causes problems when lost packet events are placed in the jitterbuffer.
 2013-09-26 16:21:33 +02:00
Wim Taymans
6095e2e859 rtspsrc: disable checks when linking pads 
We know the pad links will work (and we don't check the return value
anyway).
 2013-09-25 17:42:02 +02:00
Wim Taymans
2efd58fc84 rtpbin: avoid some pad link checks 
Link pads without checks, we know it will work.
 2013-09-25 17:38:31 +02:00
Sebastian Dröge
4a91a93d4e qtmux: Don't error out if downstream is not seekable for non-fragmented variants 
Doing so would be a regression over 1.0 and breaks the unit test.
However the result will be most likely unusable, so let's post
a warning message on the bus.
 2013-09-25 13:25:34 +02:00
Wim Taymans
97f4674655 rtpjitterbuffer: calculate some stats 2013-09-25 10:50:05 +02:00
Wim Taymans
b1d29483bb rtpjitterbuffer: move send_lost_event function 
Move the send_lost_event function to the do_lost_event handling, there is no
need to have a separate function.
 2013-09-25 10:50:05 +02:00
Thiago Santos
dc02d91c14 qtdemux: add code to parse creation time earlier than 1970 
Use g_date_time seconds manipulation to allow to cover the quicktime
spec for creation_time. It uses seconds since 1904.

Both paths could be done using the generic approach of seconds since
1904 with GDateTime handling, but the first path using seconds from
1970 should be more commonly found and avoids a few objects creation and
ref/unref, so keep it there for performance.

Additionally, the code for handling seconds since 1970 changed from >
to >= because having 0 seconds since 1970 is also a valid case for that
path to handle.

https://bugzilla.gnome.org/show_bug.cgi?id=707975
 2013-09-24 15:16:54 -07:00
Matej Knopp
a1a493dae4 matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily 
https://bugzilla.gnome.org/show_bug.cgi?id=708505
 2013-09-24 15:12:44 -07:00
Wim Taymans
adf5d96044 rtpmanager: update docs 2013-09-23 16:34:15 +02:00
Wim Taymans
e5019de80d docs: update docs with 1.0 element names 2013-09-23 15:36:47 +02:00
Wim Taymans
8ce674da87 rtpjitterbuffer: always store lost event in jitterbuffer 
Always prepare a lost event in the jitterbuffer, it is to wake up and make the
pushing thread continue. We drop the event when we are not supposed to push lost
events downstream.
 2013-09-23 14:45:27 +02:00
Wim Taymans
9f3345fcc2 rtpjitterbuffer: schedule lost event differently 
Schedule the lost event by placing it inside the jitterbuffer with the seqnum
that was lost so that the pushing thread can interleave and push it properly.
 2013-09-23 14:45:27 +02:00
Wim Taymans
ae389aeb0c rtpjitterbuffer: remove list debug 2013-09-23 14:45:26 +02:00
Wim Taymans
28641e3145 rtpjitterbuffer: add type to the item 
So that the upper layer can know what data is contained in the item.
 2013-09-23 14:45:26 +02:00
Wim Taymans
479c7642fd rtpjitterbuffer: fix flush 
Pass function to flush to properly free the queue items.
 2013-09-23 14:45:25 +02:00
Wim Taymans
0cc887eb98 rtpjitterbuffer: append seqnum -1 packets 2013-09-23 14:45:25 +02:00
Wim Taymans
39a2ba669d rtpjitterbuffer: use structure to hold packet information 
Make the jitterbuffer operate on a structure containing all the packet
information. This avoids mapping the buffer multiple times just to get the RTP
information. It will also make it possible to store other miniobjects such as
events later.
 2013-09-23 14:45:25 +02:00
Wim Taymans
1760817005 rtpjitterbuffer: update expected timer when possible 
When we receive a packet and we have some missing packets, we can update their
estimated arrival times based on the timestamp difference.
 2013-09-23 14:45:25 +02:00
Wim Taymans
fdc1ed1680 rtpjitterbuffer: fix order of timeout events 
Improve the order of the timeout events, if there are timers with the same
timeout, we want to trigger the lowest seqnum first. For this we need to loop
over the complete array of timers to find the best one before triggering the
timeout.
 2013-09-23 14:45:25 +02:00
Wim Taymans
0b1a7edfea rtpjitterbuffer: send lost event before signaling next buffer 
First send the lost event, then update the next_seqnum counter and then
send the signal to the pushing thread that it can retry to push a buffer. This
avoids pushing out buffers before the lost event is pushed.
 2013-09-23 14:45:25 +02:00
Wim Taymans
5051f51f0a jitterbuffer: configure clock-rate on jitterbuffer 
Add a get and setter to configure the clock-rate in the jitterbuffer instead of
passing it as an argument to the insert method.
 2013-09-23 14:45:24 +02:00
Wim Taymans
3c421e7e48 rtpjitterbuffer: add option to reset retransmission timers 2013-09-23 14:45:24 +02:00
Wim Taymans
6f4deab298 rtpjitterbuffer: stop the timer thread 
The timeout code could release the lock so we need to check if we are allowed to
wait for the clock some more.
 2013-09-23 14:45:24 +02:00
Wim Taymans
cba4e6a707 rtpjitterbuffer: unlock only once 2013-09-23 14:45:24 +02:00
Wim Taymans
5dc207948c rtpjitterbuffer: improve flush and shutdown 
There is no need to unschedule the timer in flush-start, flush-stop will remove
the timers and unschedule.
Unschedule the current timer before attempting to join the timer thread.
 2013-09-23 14:45:23 +02:00
Wim Taymans
a512cc2d3c rtpjitterbuffer: set correct expected time 
When we already have a timer for a packet, skip it but don't forget to adjust
the dts to the expected dts of the next packet.
 2013-09-23 14:45:23 +02:00
Wim Taymans
517ea0f4d9 jitterbuffer: improve debug 2013-09-23 14:45:23 +02:00
Wim Taymans
c395bf62dd alpha: use POFFSET instead of OFFSET 
Use the more correct POFFSET macro to get the offset of a component in its
plane. The offset macro gives the offset of the component relative to the start
of the frame.
 2013-09-23 14:45:23 +02:00
Sebastian Dröge
94ad6724ba goom: Fix MMX assembly compilation with clang 
clang does not want or need a clobber list for emms:
error: clobbers must be last on the x87 stack

Patch taken from the FreeBSD ports, provided by
Dan McGregor <dan.mcgregor@usask.ca>
 2013-09-21 18:48:19 +02:00
Sebastian Dröge
d8841b4832 matroska-demux: Make sure that subtitle buffers are \0-terminated 
https://bugzilla.gnome.org/show_bug.cgi?id=707933
 2013-09-20 10:22:40 +02:00
Andoni Morales Alastruey
cfefdaebb6 qtmux: handle issues correctly when downstream is not seekable 
The streamable property only make sense for fragmented formats.
For regular MP4, when downstream is not seekable we can't rewrite
the headers, so qtmux can only work with fast-start=TRUE, where
the headers are written finishing the file.
For fragmented MP4, when streamable is not seekable and the streamable
property is FALSE, we must enforce streamable=TRUE warning the user
about this change

https://bugzilla.gnome.org/show_bug.cgi?id=707242
 2013-09-20 10:09:48 +02:00
Andoni Morales Alastruey
9ae5082204 qtmux: make "streamable" TRUE as default 
The most common use case for fragmented MP4 (Dash and Smooth Streaming)
is producing streamable content (even for VOD). streamable=FALSE would only
be used to generate fragmented MP4 with and index of MOOF's that could
be reproduced without a playlist/manifest
https://bugzilla.gnome.org/show_bug.cgi?id=707242
 2013-09-20 10:09:48 +02:00
Andoni Morales Alastruey
5732684e18 qtmux: deprecate the streamable property for non-fragmented MP4 
The streamable property only makes sense for fragmented MP4.
https://bugzilla.gnome.org/show_bug.cgi?id=707242
 2013-09-20 10:09:48 +02:00
Wim Taymans
926e2fa93b alpha: don't assume planar formats have just 1 block 
Don't assume planar formats have just one memory block with the data but use the
macros to access the right memory block where a component can be found.
 2013-09-19 16:50:44 +02:00
Wim Taymans
fd6c57cf10 rtpjitterbuffer: keep delay as a separate variable in timer 
Keep a separate delay in the timer so that we still know the original timestamp
of the packet that this timer refers to. We can then place the correct
running-time in the Retransmission event.
 2013-09-19 14:32:48 +02:00
Wim Taymans
d34184dd03 rtpjitterbuffer: fix writability of properties 2013-09-19 14:32:48 +02:00
Wim Taymans
6bb2626498 rtpjitterbuffer: reevaluate the current timer after timeout 
When we trigger the timeout logic of a timer, reevaluate it because it is
possible that it still has the lowest timeout.
 2013-09-18 16:33:40 +02:00
Wim Taymans
8d021b6ede rtpjitterbuffer: don't update time when unscheduled 
Don't try to estimate the current time when we got unscheduled.
 2013-09-18 16:31:26 +02:00
Wim Taymans
65606a25bf rtpjitterbuffer: init packet spacing on first buffer 
Already init the packet spacing variables on the first buffer so that we can
calculate the spacing on the second buffer already.
 2013-09-18 16:29:37 +02:00
Wim Taymans
f2efdf28f5 rtpjitterbuffer: push the lost event from the timer thread 
Instead of pushing the lost event from the chain function, schedule a timeout
that will push the lost event from the timer thread. This avoid blocking the
upstream thread while we push and sync the event.
 2013-09-18 14:57:09 +02:00
Wim Taymans
5d5fc03e04 rtpjitterbuffer: round gap duration to multiple of duration 
Make sure the gap duration in the lost event is a multiple of the packet
duration.
Enable another test.
 2013-09-18 14:12:47 +02:00
Wim Taymans
6e4a051d40 rtpjitterbuffer: keep track of duration 
Keep track of the estimated duration of missing packets and use it in the lost
event.
Enable another unit test
 2013-09-18 12:29:38 +02:00
Wim Taymans
ac3bb3acf6 rtpjitterbuffer: handle large gaps with one lost event 
When we have a large number of missing packets, generate one lost event for all
the packets that have no chance of being pushed out in time.
Fix and activate unit test for large gaps.
 2013-09-18 11:59:28 +02:00
Wim Taymans
26402e1c21 rtpjitterbuffer: refactor lost event sending 
Also make sure we only increment the expected seqnum and last
output timestamp.
 2013-09-18 11:57:06 +02:00
Wim Taymans
f49981a597 jitterbuffer: refactor timeout triggers 2013-09-17 23:29:56 +02:00
Wim Taymans
047021c443 jitterbuffer: simplify the timeout code 
Keep track of the current time in the timeout loop.
Loop over all timers and trigger all the expired ones, we can do this in the
same loop that selects the new best timer.
 2013-09-17 23:29:56 +02:00
Wim Taymans
fa1ef3328b jitterbuffer: rearrange timer update code 
Also update the timers when retransmission is disabled. We need to
do this because when we added LOST timers when we detected missing packets and
we need to remove those timers when the packet finally arrives.
 2013-09-17 23:29:56 +02:00
Tim-Philipp Müller
7a76595b22 videomixer: link to libm for maths stuff 
Fixes undefined references to rint and pow on ubuntu
build bot.
 2013-09-17 22:02:04 +01:00
Wim Taymans
232fdd8b56 jitterbuffer: release lock on shutdown 2013-09-17 15:19:42 +02:00
Matej Knopp
b2982bb749 qtmux: remove MAX_TOLERATED_LATENESS 
https://bugzilla.gnome.org/show_bug.cgi?id=707411
 2013-09-16 11:11:12 -03:00
Wim Taymans
4de919a17a jitterbuffer: use separate thread for timeouts 
Use a separate thread for scheduling the timeouts instead of using the
downstream streaming thread that might block at any time.
 2013-09-16 15:55:55 +02:00
Matej Knopp
b363832c2c qtmux: set first_ts to DTS for streams that have DTS 
https://bugzilla.gnome.org/show_bug.cgi?id=707340
 2013-09-16 12:14:00 +02:00
Matej Knopp
39f7e52266 qtmux: make sure duration is a valid number for last buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=707340
 2013-09-16 12:14:00 +02:00
Matej Knopp
4e3c13c87c qtmux: use segment.start or last buffer end time in case of missing DTS 
https://bugzilla.gnome.org/show_bug.cgi?id=707340
 2013-09-16 12:14:00 +02:00
Matej Knopp
85728c04c4 Revert qtmux: Use buffer PTS if DTS is not set" 
This reverts commit f72c3cf71fde622067f41f31a53978ba4c94469d.

https://bugzilla.gnome.org/show_bug.cgi?id=707340
 2013-09-16 12:13:54 +02:00
Sebastian Dröge
d646a34681 videomixer: Update orc generated files 
https://bugzilla.gnome.org/show_bug.cgi?id=708131
 2013-09-16 11:03:06 +02:00
Olivier Crête
b9ceafe5af rtpsession: Demux RTCP buffers from the RTP stream 
If there are RTCP buffers in the RTP stream, process them as
RTCP. This way, we want receive streams following RFC 5761

https://bugzilla.gnome.org/show_bug.cgi?id=687657
 2013-09-13 16:25:49 +02:00
Jan Schmidt
299d3f5c42 rtp: Remove bogus extra caps from L24 template. 
The extra caps entry in the template was making it sometimes
get plugged for any dynamically allocated payload type.
 2013-09-13 23:27:49 +10:00
Wim Taymans
28e5f90988 rtpbin: use PacketInfo for the sender 
Avoid mapping the packet multiple times when sending RTP.
 2013-09-13 14:34:28 +02:00
Wim Taymans
a02c9473d8 rtpbin: store more in the PacketInfo 
Store all info in the PacketInfo so that we can avoid mapping the packet
multiple times.
 2013-09-13 14:34:28 +02:00
Wim Taymans
e5c789abd6 session: store more in the PacketInfo structure 2013-09-13 14:34:28 +02:00
Wim Taymans
47662f9ca4 rtpbin: RTPArrivalStats -> RTPPacketInfo 
Rename a structure because we are also going to use this for the sender
bits.
 2013-09-13 14:34:28 +02:00
Wim Taymans
c795b72988 source: small cleanups 2013-09-13 14:34:27 +02:00
Thiago Santos
566b0dce40 qtdemux: only update stop position if seek requests it 
Check for GST_SEEK_TYPE_NONE for stop poistion and only update
the stop time if it is requested. Otherwise just maintain whatever
was stored at the segment

https://bugzilla.gnome.org/show_bug.cgi?id=707530
 2013-09-13 09:21:12 -03:00
Rico Tzschichholz
8ed1ff6821 rtp: Add missing headers tp fix make dist 
In addition to a956a6ceb2
 2013-09-13 14:06:13 +02:00
Sebastian Dröge
b95ddd55cd flacparse: Make sure we have enough data to read image tags 
Thanks to iputinei for reporting this on IRC.
 2013-09-12 15:39:51 +02:00
Wim Taymans
9f9ba21404 jitterbuffer: handle segments with non-0 start 
We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to
transform it back to a buffer timestamp before pushing out the buffer.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931
 2013-09-12 15:04:30 +02:00
Seán de Búrca
9d3dbd6581 matroskademux: Fix off-by-one in validation of UTF-8 
https://bugzilla.gnome.org/show_bug.cgi?id=707933
 2013-09-12 09:19:15 +02:00
Thibault Saunier
9f4a8ccdf4 videomixer: Do not check if caps are empty when they are NULL 
In the case the caps are actually NULL, we should just concider it the
same way as empty caps in that case.
 2013-09-11 14:33:31 -03:00
Seán de Búrca
268058eb37 videomixer: fix build if orc is not installed 
https://bugzilla.gnome.org/show_bug.cgi?id=707886
 2013-09-11 00:17:44 +01:00
Thiago Santos
193ce9110e matroskademux: Preserve seqnum when pushing seek upstream 
After converting a seek from time to bytes, use the same seqnum
on the event that goes upstream
 2013-09-10 17:57:49 -03:00
Thiago Santos
be0eeae491 qtdemux: track streams that are EOS on push mode to finish earlier 
When the segment has a defined stop position, qtdemux should check
when streams reach this position and mark those as EOS. When all
streams are EOS it will return GST_FLOW_EOS to upstream to allow
the pipeline to finish instead of continuously consume buffers
from upstream that are not useful for the segment.

https://bugzilla.gnome.org/show_bug.cgi?id=707530
 2013-09-10 16:43:17 -03:00
Thiago Santos
33cf8b679d qtdemux: preserve stop of segment when doing seeks in push mode 
When handling seeks in push mode, qtdemux converts the seek to bytes
and pushes upstream. It needs to keep track of the seek and the
subsequent segment to be able to map them back to the requested
seek time and properly preserve the segment stop of the seek.

This is done by using the start offset in bytes of the seek,
that should be the same of the segment from upstream. And this
is also backwards compatible with what qtdemux already was using.

https://bugzilla.gnome.org/show_bug.cgi?id=707530
 2013-09-10 16:42:36 -03:00
Mathieu Duponchelle
8db40a8c7f videomixer: Add colorspace conversion 
https://bugzilla.gnome.org/show_bug.cgi?id=704950
 2013-09-10 10:37:23 +02:00
Mathieu Duponchelle
707e39fe7a videomixer: Don't send reconfigure event when formats or PAR are different 
It is racy with multiple pads.

https://bugzilla.gnome.org/show_bug.cgi?id=704950
 2013-09-10 10:36:48 +02:00
Mathieu Duponchelle
8db3648544 videomixer: Bundle private copies of videoconvert code 
Ideally, this would be part of libgstvideo.
Prefixes videoconvert symbols with videomixer_.

https://bugzilla.gnome.org/show_bug.cgi?id=704950
 2013-09-10 10:36:30 +02:00
Wim Taymans
9f9bcbc405 rtspsrc: only wait if we flushed 
Only wait for the STREAM_LOCK when we flushed something when sending
a command for PAUSED or PLAYING.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611
 2013-09-09 15:13:46 +02:00
Wim Taymans
7b2e002879 rtspsrc: return when a flush was issued 
Make gst_rtspsrc_loop_send_cmd() return TRUE when the current
action has been flushed
 2013-09-09 15:13:46 +02:00
David Holroyd
a956a6ceb2 rtp: add L24 pay and depayloader 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734
 2013-09-09 15:13:46 +02:00
Matej Knopp
a5ceab82dd matroskademux: fix leaking buffer and caps 
https://bugzilla.gnome.org/show_bug.cgi?id=707688
 2013-09-07 15:50:36 +01:00
Tim-Philipp Müller
60e72b0254 udpsrc: fix build on win32 
gstudpsrc.c:855:15: error: #if with no expression
 2013-09-05 19:46:37 +01:00
Wim Taymans
5d2ff288b3 avidemux: handle unseekable streams 
Handle streams that we can't seek in and ignore them in the
seek logic.
 2013-09-04 15:53:05 +02:00
Wim Taymans
6f0e8a8b87 avidemux: only check video compression for video streams 
Or else we might deref a stream with a NULL strf.vids and segfault
 2013-09-04 15:53:05 +02:00
Alex Ashley
a965185dee qtdemux: Add support for the avc3 sample entry format of the AVC file format 
Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
structure for fragmented MP4 called "avc3". The principal difference
between AVC1 and AVC3 is the location of the codec initialisation
data (e.g. SPS, PPS). In AVC1 this data is placed in the initial
MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data
goes in the first sample of every fragment (i.e. the first sample in
each mdat box).  The principal reason for avc3 is to make it easier
for client implementations, because it removes the requirement to
insert the SPS+PPS in to the decoder pipeline every time there is a
representation change.

This commit adds support for the "avc3" atom, which is almost identical
to the "avc1" atom, except it does not contain any SPS or PPS data.

https://bugzilla.gnome.org/show_bug.cgi?id=702004
 2013-09-04 13:33:22 +02:00
Mathieu Duponchelle
b68f419b6f videomixer: Don't set EOS to FALSE when the collectpad *is* EOS 
https://bugzilla.gnome.org/show_bug.cgi?id=707238
 2013-09-04 11:09:04 +02:00
Matej Knopp
349afc633a flacparse: cleanup on error after state change 
https://bugzilla.gnome.org/show_bug.cgi?id=707229
 2013-09-03 18:06:18 +02:00
Sebastian Dröge
7f59436979 udpsrc: Bind to multicast addresses on non-Windows systems 
On Windows it's not possible to bind to a multicast address
but the OS will make sure to filter out all packets that
arrive not for the multicast address the socket joined.

On Linux and others it is necessary to bind to a multicast
address to let the OS filter out all packets that are received
on the same port but for different addresses than the multicast
address

And deprecate the multicast-group property and replace it with the
address property.

https://bugzilla.gnome.org/show_bug.cgi?id=707042
 2013-09-03 11:23:24 +02:00
Matej Knopp
73751dbbe7 flacparse: Free GstBaseParseFrame if pushing a header failed 2013-09-03 10:10:49 +02:00
Sebastian Dröge
edf6d28765 udpsrc: Refactor address resolval into its own function 2013-09-03 10:10:49 +02:00
Tim-Philipp Müller
966f848edb replaygain: fix taglist leak in rganalysis 
And add some FIXMEs.
 2013-09-02 23:00:29 +01:00
Sebastian Dröge
1971c43279 flacparse: Properly propagate downstream flow returns upstream 
https://bugzilla.gnome.org/show_bug.cgi?id=707229
 2013-09-02 11:56:33 +02:00
Tim-Philipp Müller
1dfc1f2686 Don't use setlocale in plugins() 
Only apps should call setlocale(), not libraries.
 2013-09-01 21:18:38 +01:00
Wim Taymans
d851b8a8b4 rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay 
RTP buffer allocation should not be done with padding for the specific MPEG2
header as the padding is done at the end of the buffer and the last byte is
the size of the padding.

https://bugzilla.gnome.org/show_bug.cgi?id=706970
 2013-08-29 13:15:15 +02:00
Bernhard Miller
f7528d274b autovideosink: add sync property 
https://bugzilla.gnome.org/show_bug.cgi?id=706955
 2013-08-29 12:23:24 +02:00
Bernhard Miller
2fa68fce07 autoaudiosink: introduce sync property 
https://bugzilla.gnome.org/show_bug.cgi?id=706955
 2013-08-29 12:23:23 +02:00
Thiago Santos
9549289a18 qtdemux: push buffers after segment stop until reaching a keyframe 
This should make decoders able to precisely push buffers until the stop
time in case they need the next keyframe to do it.

Also, according to gst_segment_clip, it should only push a buffer that
the starting ts is strictly smaller than the segment stop, so we change
the min < comparison for <=
 2013-08-28 12:58:56 -03:00
Sebastian Dröge
76293efd72 Release 1.1.4 2013-08-28 12:52:25 +02:00
Wim Taymans
2a8566ddec matroska-mux: remove framerate restriction 
Remove the framerate restriction on the caps.
 2013-08-27 15:25:16 +02:00
Wim Taymans
f1106cde66 session: only update next check time when reconsidering 
Don't update the next RTCP check time in all cases but only when we
reconsidered. This avoids delaying sending a full RTCP packet when we
are doing early feedback.
 2013-08-27 09:55:52 +02:00
Wim Taymans
47065db0b6 session: add more debug 2013-08-27 09:55:52 +02:00
Wim Taymans
454d75951e jitterbuffer: fix types of the retransmission event 2013-08-27 09:55:52 +02:00
Wim Taymans
dd4af0d11c jitterbuffer: only timeout EXPECTED timers on gap 
Only timeout the EXPECTED timers when we detect a large seqnum gap.
 2013-08-27 09:44:18 +02:00
Wim Taymans
4b7bcc2ec1 rtsession: fix locking 
We need to take the session lock when getting and manipulating the
source.
 2013-08-26 11:50:27 +02:00
Wim Taymans
3f46527f75 rtpsession: add some more debug 2013-08-26 11:50:13 +02:00
Mathieu Duponchelle
5d21f8f2e3 videomixer: don't send flush_stop twice. 
If we get flush start and a seek we need to only send flush_stop once.

More info at #706441
 2013-08-23 20:17:11 -04:00
Tim-Philipp Müller
9b0bcc01a0 multipartdemux: propagate discont 2013-08-23 15:57:46 +01:00
Tim-Philipp Müller
c3af414cbf multipartdemux: remove dynamic sourcpads when going from PAUSED to READY 2013-08-23 15:57:46 +01:00
Tim-Philipp Müller
7d78a68c8d multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last 
https://bugzilla.gnome.org/show_bug.cgi?id=637754
 2013-08-23 15:57:46 +01:00
Wim Taymans
54e7e7547a rtxqueue: add property to configure queue size 2013-08-23 15:47:25 +02:00
Wim Taymans
84833bed11 rtpbin: proxy jitterbuffer do-retransmission property 2013-08-23 12:10:19 +02:00
Michael Olbrich
23d4044e2c avimux: unmap the correct buffer 
The audio buffer was mapped so unmap it and not the video buffer

https://bugzilla.gnome.org/show_bug.cgi?id=706642
 2013-08-23 11:32:52 +02:00
Wim Taymans
89b9019e3e rtx: various improvements 
Use locking
Don't push from the event handler, collected packets in a queue and push from
the chain function.
Clear queues on shutdown.
 2013-08-21 17:02:27 +02:00
Wim Taymans
ee15bc9284 session: generate events correctly 
Do correct shifting of the bitmask for lost packets.
 2013-08-21 17:02:27 +02:00
Wim Taymans
67523d3ecb rtp: register rtx element better 2013-08-21 17:02:26 +02:00
Wim Taymans
f626e29897 jpegdepay: add some more debug 2013-08-21 12:56:35 +02:00
Wim Taymans
77ed44a88a rtpgstdepay: only push events when they changed 
Keep track of the STREAM_START and TAG events and only push them
when they changed.
 2013-08-21 12:10:00 +02:00
Wim Taymans
b144809b7c rtpgstpay: taglists should not be merged in 1.0 2013-08-21 10:52:59 +02:00
Wim Taymans
69b0dcd7df rtpgstdepay: flush on FLUSH_STOP event 2013-08-21 10:28:50 +02:00
Wim Taymans
5ff9093843 rtpgstpay: reset on state change 
Do full reset on state change to READY
 2013-08-21 10:03:52 +02:00
Wim Taymans
ae9239aac7 rtpgstpay: reset on FLUSH_STOP 
Clear the adapter and pending buffer list on FLUSH_STOP.
 2013-08-21 09:55:20 +02:00
Wim Taymans
2e8955df39 rtpgstpay: don't use clock for config interval 
We can't use the clock to time our config-interval because we are not
live (or there might not be a clock or the clock might not be running).
Instead just simply take the timestamp diff.
 2013-08-21 09:39:30 +02:00
Wim Taymans
182f96ff79 rtpgstay: don't use // comments 2013-08-21 09:33:04 +02:00
Youness Alaoui
e22f7e91c4 rtspsrc: Fix response argument in handle-request signal 2013-08-21 09:06:02 +02:00
Youness Alaoui
6636efd31a rtspsrc: Add sdes property and proxy it to rtpbin 2013-08-21 09:06:02 +02:00
Youness Alaoui
62a6f58697 Send a stream-start whenever we send tags 
This is to make sure tags are cleared on the client if the
stream-start was previously lost, otherwise, the client may end
up with a merged taglist of multiple songs
 2013-08-21 09:06:01 +02:00
Youness Alaoui
05bcfee5a3 rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval 
This is useful in case the packet containing the inlined caps was lost
or if new client joins an already running RTP stream and they missed
the previous tag events.
This also makes the payloader keep a list of merged tags so the retransmitted
tag event contains all previously received. A STREAM_START event will
flush the list of tags.
 2013-08-21 09:06:01 +02:00
Youness Alaoui
1f4ca28868 rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time 2013-08-21 09:06:01 +02:00
Youness Alaoui
9257409613 rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps 2013-08-21 09:06:01 +02:00
Youness Alaoui
2d53289b6b rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList 
This is necessary to fix event/caps sending. If we send a STREAM_START
packet, it will cause an error because the stream didn't receive its
caps and new-segment events, so we must wait for the first buffer before
sending the stream-start event buffer. However, the caps will be sent
at the same time and so the 'inline caps' will be set for the event.
We need to be able to payload individual packets (data, caps or events)
and only send them when we call flush.
 2013-08-21 09:06:01 +02:00
Youness Alaoui
0070ba76f2 rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START 2013-08-21 09:06:01 +02:00
Youness Alaoui
6155b27971 rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3 2013-08-21 09:06:01 +02:00
Wim Taymans
587dc055e9 jitterbuffer: handle EOS 
When the queue is empty, and we received EOS, pause and push an EOS
event downstream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387
 2013-08-20 14:36:59 +02:00
Wim Taymans
533f26fc99 jitterbuffer: update docs 2013-08-20 10:26:15 +02:00
Wim Taymans
c7f9ef8012 jitterbuffer: update all timers 
Keep looping over all registered timers so that we can mark them lost instead of
stopping as soon as we find the timer for the current seqnum.
 2013-08-20 10:25:17 +02:00
Wim Taymans
5debda9ca1 jitterbuffer: remove unused variables 2013-08-20 08:55:50 +02:00
Wim Taymans
a88db5fa2c jitterbuffer: reorganize timer handling 
Restructure handling of incomming packet and the gap with the expected seqnum
and register all timers from the _chain function.
Convert a timer to a LOST packet timer when the max amount of retransmission
requests has been reached.
 2013-08-19 22:04:51 +02:00
Wim Taymans
d9d6eac4bb jitterbuffer: refactor packet spacing calculation 2013-08-19 22:04:50 +02:00
Wim Taymans
c4dc159656 jitterbuffer: keep track of last seqnum and dts 2013-08-19 22:04:50 +02:00
Wim Taymans
652ce95ca6 jitterbuffer: small cleanups 2013-08-19 22:04:50 +02:00
Wim Taymans
b4a35bbe82 jitterbuffer: reset retransmission timers in add/reschedule 
Reset the retransmission timers when adding and rescheduling a timer.
 2013-08-19 22:04:50 +02:00
Wim Taymans
cf8a0652f3 jitterbuffer: rename variables for packet spacing 2013-08-19 22:04:50 +02:00
Wim Taymans
ec82e4ab7c jitterbuffer: remove lost timer when we get the packet 
When we receive a packet, also remove the LOST timer for it.
 2013-08-19 22:04:50 +02:00
Wim Taymans
2f03b43b21 jitterbuffer: expected seqnum must increase 
Only update the expected seqnum when it is bigger than the previous expected
seqnum.
 2013-08-19 22:04:50 +02:00
Wim Taymans
c5bf376bb5 jitterbuffer: add more debug 2013-08-19 22:04:50 +02:00
Wim Taymans
ff825a2919 rtxqueue: add retransmission queue element 2013-08-19 22:04:50 +02:00
Wim Taymans
5fe18ee432 session: add some docs 2013-08-19 22:04:49 +02:00
Wim Taymans
482dacfb54 session: handle NACK feedback and generate events 
Handle and parse the feedback NACK packets and generate a Retransmission
event for each NACKed packet
 2013-08-19 22:04:49 +02:00
Thibault Saunier
e47ffb203b videomixer: Do not send flush_stop ourself after a flush_start 
When we receive a flush_start, we should wait for the next flush_stop
and foward it, not create a flush_stop ourself.
 2013-08-17 11:40:27 +02:00
Wim Taymans
db90f6e68d h264depay: init debug category early 
Init the debug variable when we register the element because it is also used by
the payloader element when it calls the add_sps_pps method.
 2013-08-16 17:12:19 +02:00
Chris Bass
3e9dea3f8c qtdemux: check denominator isn't zero before scaling duration. 
When gst_qtdemux_configure_stream sets fps_d, check that n_samples is
non-zero before using it as a denominator to scale the stream duration.

https://bugzilla.gnome.org/show_bug.cgi?id=706076
 2013-08-16 10:14:30 +02:00
Wim Taymans
f11c2c9b3b jitterbuffer: forward flush before stopping dataflow 
First forward the flush event and then stop our loop function.
 2013-08-14 16:19:32 +02:00
Olivier Crête
4c6e636720 rtph264pay: Use the SPS/PPS handling function from the depayloader 
Remove duplicated copies

https://bugzilla.gnome.org/show_bug.cgi?id=705553
 2013-08-13 10:38:23 -04:00
Olivier Crête
742b90747d rtph264depay: Make the SPS/PPS deduplication function generic 
Make it not touch any internals of the depayloader

https://bugzilla.gnome.org/show_bug.cgi?id=705553
 2013-08-13 10:38:23 -04:00
Chris Bass
b40bf67526 aacparse: allow conversion from raw AAC to ADTS 
This patch will prepend ADTS headers to raw AAC audio frames, allowing
upstream elements to link to decoders that only support AAC in ADTS format.

Note that no error correction bits are added to ADTS frames in this code.

https://bugzilla.gnome.org/show_bug.cgi?id=615740
 2013-08-13 15:58:23 +02:00
Sebastian Dröge
282afae244 rtspsrc: Only free GCheckSum after its last usage 
https://bugzilla.gnome.org/show_bug.cgi?id=705760
 2013-08-13 12:44:11 +02:00
Matej Knopp
2269ac8f28 qtdemux: elst should offset samples instead of buffers 
The current approach where buffers are offset is not ideal, as during seek
and loop current time is compared to sample times.

https://bugzilla.gnome.org/show_bug.cgi?id=700264
 2013-08-12 13:48:04 +02:00
Thibault Saunier
6c349d6ec3 videomixer: Send EOS if buf_end >= segment.stop 
That means the whole segment is already played, and we are sure we
are EOS at that point.

Also handle segment seeks, and do not send EOS in that case.
 2013-08-11 19:05:18 +02:00
Matej Knopp
96afba915a avidemux: send proper stream_start event 
https://bugzilla.gnome.org//show_bug.cgi?id=705449
 2013-08-08 11:57:32 +02:00
Sebastian Dröge
9863e08839 matroskademux: Don't print warnings during flushing and stop as soon as possible 
https://bugzilla.gnome.org//show_bug.cgi?id=705442
 2013-08-08 11:53:15 +02:00
Tim-Philipp Müller
957c8e3e61 rtpvp8depay: mark key frames and delta frames properly 
https://bugzilla.gnome.org/show_bug.cgi?id=705550
 2013-08-07 11:14:38 +01:00
Wim Taymans
48174164eb session: add NACK feedback in RTCP 2013-08-06 15:50:19 +02:00
Wim Taymans
4379ed1dee source: add methods to register NACK 
Add a method to register a missing packet for an ssrc along with
methods to get the missing packets and clear them.
 2013-08-06 15:50:19 +02:00
Wim Taymans
50638b8106 session: handle Retransmission event and schedule NACK 
Handle the retransmission event from downstream and use it to schedule a NACK
request.
 2013-08-06 15:50:19 +02:00
Wim Taymans
0bddbd682d session: pass data to remove func 
Pass the data to the remove function because we are going to deref it when there
is pli or fir.
 2013-08-06 15:50:19 +02:00
Thibault Saunier
38946bd9f4 qtdemux: Fix compilation 2013-08-06 15:31:38 +02:00
Thibault Saunier
593a31f2b4 qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE 2013-08-06 15:17:44 +02:00
Thibault Saunier
c5fa4666b7 videomixer: Make sure to send EOS if the buffer end time equals the segment end time 
Otherwize EOS never gets sent in that particular case.
 2013-08-06 12:21:33 +02:00
Sjoerd Simons
d14d4c436c goom: Ensure src caps are writable 
In some cases the src caps determined by goom weren't writable, causing
a bunch of assertion failures and failed caps. Fixed by always
explicitely making the caps writable

https://bugzilla.gnome.org/show_bug.cgi?id=705475
 2013-08-05 15:33:39 +02:00
Wim Taymans
3c82de59f9 session: use common send_rtcp method 
Reuse the send_rtcp method that already asks for the current time when
requesting a keyframe.
 2013-08-05 15:02:59 +02:00
Wim Taymans
3c14c6021c session: Don't use ClockTimeDiff for unsigned delays 2013-08-05 15:02:59 +02:00
Edward Hervey
4f4f6432cc qtmux: Use buffer PTS if DTS is not set 
Avoids ending up with completely bogus scaled duration/pts when new
buffers have invalid DTS.
 2013-08-04 17:15:38 +02:00
Tim-Philipp Müller
7272dec5fe rtpdec: use generic marshaller 2013-08-04 11:20:41 +01:00
Tim-Philipp Müller
fe098e3aff udp: remove unused marshal and enumtypes files 2013-08-04 11:03:07 +01:00
Tim-Philipp Müller
7469cd3a4c rtpmanager: use generic marshaller 2013-08-04 11:03:07 +01:00
Wim Taymans
7584f91f31 jitterbuffer: send event in right direction 2013-08-04 00:24:36 +02:00
Wim Taymans
9613e481ad session: add FIR and PLI like other RTCP packets 
Add the FIR and PLI packets like the other RTCP packet instead of from the
on-sending-rtcp default signal handler.
 2013-08-03 00:33:24 +02:00
Wim Taymans
743e1b1191 jitterbuffer: fix property ranges 2013-08-02 17:22:55 +02:00
Wim Taymans
cd0164f4cc jitterbuffer: push retransmission events 2013-08-02 16:43:59 +02:00
Wim Taymans
9a13267e85 jitterbuffer: add support for retransmission retry 
When we didn't receive a packet after requesting retransmission, retry
asking for retransmission for a certain period.
 2013-08-02 14:54:56 +02:00
Wim Taymans
e9ad5126db jitterbuffer: add properties 
Add properties to control retransmission parameters
 2013-08-02 14:47:56 +02:00
Wim Taymans
a8c7ff7489 jitterbuffer: use corrected timeout when rescheduling 
When we recalculate the timeout, use the corrected timeout value depending on
the timer type.
 2013-08-02 12:44:58 +02:00
Wim Taymans
9c7e3e3455 jitterbuffer: update timers after queueing 
Else we might update the timer needlessly for duplicates.
 2013-08-02 12:43:00 +02:00
Wim Taymans
ebd6b8f8ab jitterbuffer: move method up 2013-08-02 12:42:08 +02:00
Wim Taymans
f6b6797874 jitterbuffer: small cleanup 2013-08-02 06:28:32 +02:00
Wim Taymans
0e41414926 jitterbuffer: unschedule old expected packets 
When we receive a new packet, unschedule old outstanding packets when their
seqnum is too far away.
 2013-08-01 23:36:07 +02:00
Wim Taymans
70695466ed jitterbuffer: refactor timer update 2013-08-01 23:32:00 +02:00
Wim Taymans
4ab3f5d3da jitterbuffer: update timers when removing 
Update the timers when we remove a timer.
Handle canceled timers, make them unschedule the current timer and
trigger the timeout code.
 2013-08-01 23:24:29 +02:00
Wim Taymans
b983cf675b jitterbuffer: fix typo 2013-08-01 23:22:02 +02:00
Wim Taymans
f3c658cbe6 jitterbuffer: improve timeout management 
If we change the seqnum of an existing timer and we were waiting for
that timer, unschedule it. If we change the timeout of an existing timer and we
were waiting on it, only unschedule when the new time is smaller.
 2013-08-01 15:40:52 +02:00
Wim Taymans
77e5d320ab jitterbuffer: install timer for expected arrival 
Install a timer that is triggered when the expected arrival time of a packet
expired.
 2013-08-01 15:11:13 +02:00
Wim Taymans
f08d98404e jitterbuffer: improve unschedule of timers 
Conflicts:
	gst/rtpmanager/gstrtpjitterbuffer.c
 2013-08-01 14:57:11 +02:00
Wim Taymans
9d3b824e2a jitterbuffer: move code around 2013-08-01 12:21:53 +02:00
Wim Taymans
fe32e80c92 jitterbuffer: estimate inter packet spacing 
When we see two packets with consecutive seqnums and a different RTP time, use
the DTS difference as the inter packet spacing estimate.
 2013-08-01 12:07:11 +02:00
Wim Taymans
255b7106f5 jitterbuffer: keep track of current timeout 2013-08-01 12:01:15 +02:00
Wim Taymans
7e43dba19b jitterbuffer: cleanup timer handling 2013-08-01 11:49:10 +02:00
Wim Taymans
9d88ac9cbb jitterbuffer: reset is only possible with a GAP 2013-08-01 11:40:41 +02:00
Wim Taymans
f864131227 jitterbuffer: operate on DTS 
Make the jitterbuffer schedule the timeouts based on the DTS instead
of the PTS. This makes it all smoother with reordered frames and gives
the decoder time to reorder the frames in time.
 2013-08-01 11:36:56 +02:00
Wim Taymans
80c5934290 jitterbuffer: rename timout variable 2013-08-01 11:14:12 +02:00
Wim Taymans
aa951433ee jitterbuffer: small cleanup 2013-07-31 17:08:58 +02:00
Wim Taymans
69c78f72d5 jitterbuffer: block output in paused or buffering 2013-07-31 16:59:58 +02:00
Wim Taymans
4fbbc53a49 jitterbuffer: store pts in timer 
Only store the pts in the timer so that we can both do timeouts with timings on
the input and output of the jitterbuffer.
 2013-07-31 16:59:09 +02:00
Wim Taymans
77846d35c6 rtpjitterbuffer: refactor jitterbuffer 
Refactor the jitterbuffer code. Make separate function for peeking a buffer,
pushing the next buffer, waiting for timeouts and handling the timeouts.

The main loop now tries to push as many buffers as it can until it runs out of
buffers or when it detects a seqnum discont. Then it will wait for some event to
happen before attempting to push more buffers.

Make methods to register timeouts in an array. These timeouts are registered
when we detect a missing packet, sync for the first packet or when we find an
estimation for the end-of-stream.

This greatly simplifies and clarifies the code and also makes it possible to
register more complicated timeout schemes later.
 2013-07-30 23:24:23 +02:00
Wim Taymans
ea931d4f57 rtpjitterbuffer: use NULL to ignore percent 
If we pass NULL to pop and push we ignore the percent result.
 2013-07-30 23:24:23 +02:00
Wim Taymans
b3e8a85a54 jitterbuffer: refactor 
Move eos estimation into separate function
 2013-07-30 23:24:22 +02:00
Tim-Philipp Müller
a5532b4510 flvdemux: don't leak stream_id string 
https://bugzilla.gnome.org/show_bug.cgi?id=705142
 2013-07-30 14:28:19 +01:00
Sebastian Dröge
2e35b36aab gst: Don't swap start/stop for negative rates in the SEGMENT query 2013-07-29 12:12:41 +02:00
Matej Knopp
47ed79fb1c qtdemux: Check for data size when parsing h264 codec data from strf atom 2013-07-29 11:53:07 +02:00
Sebastian Dröge
722ef42196 matroskademux: Implement SEGMENT query 2013-07-29 10:53:54 +02:00
Sebastian Dröge
d135373beb flvdemux: Implement SEGMENT query 2013-07-29 10:53:47 +02:00
Sebastian Dröge
4e78974c87 avidemux: Implement SEGMENT query 2013-07-29 10:50:59 +02:00
Matej Knopp
2dcdfe07f7 qtdemux: Support H264 fourcc 
https://bugzilla.gnome.org/show_bug.cgi?id=704996
 2013-07-29 09:11:39 +02:00
Sebastian Dröge
1fbb6d30a6 avidemux: Fix duration reporting in push mode 
https://bugzilla.gnome.org/show_bug.cgi?id=700933
 2013-07-28 17:38:56 +02:00
Sebastian Dröge
89a3dc2ecd avidemux: Don't forget unmapping and unreffing buffer 2013-07-28 17:32:59 +02:00
Matej Knopp
1947587784 avidemux: unmap buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=704951
 2013-07-28 17:32:59 +02:00
Wim Taymans
02359f9219 session: don't make buffer writable prematurely 
There is no reason to make the SR buffer writable at this point. This is better
delayed until needed.
 2013-07-26 22:31:41 +02:00
Wim Taymans
0261199fc4 session: ignore RTCP for inactive sources 2013-07-26 22:31:23 +02:00
Wim Taymans
a4b4ca53c0 session: small cleanup 2013-07-26 22:25:17 +02:00
Wim Taymans
e0abd2e9b5 session: handle partial RTCP report blocks 
When we have more SSRCs to report than what fit in an RTCP packet, use a
generation counter to make sure all of them end up in a packet eventually.
 2013-07-26 17:29:10 +02:00
Wim Taymans
6cce6fb04c session: create SSRC before doing session cleanup 
Make the internal source before we do session cleanup
 2013-07-26 17:29:10 +02:00
Wim Taymans
5b0298c63e session: reorganize the report block code 2013-07-26 17:29:10 +02:00
Matej Knopp
7335b81c47 matroskademux: fix memory leak in check_subtitle_buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=704921
 2013-07-26 17:11:31 +02:00
Wim Taymans
3c44cd7c83 session: refactor active and sender checks 2013-07-26 14:21:40 +02:00
Wim Taymans
e952f7ba43 session: remove internal sources on timeout 
When an internal source times out and becomes a receiver, remove it.
 2013-07-26 12:18:01 +02:00
Wim Taymans
e9e2fe3950 session: create an internal source for RTCP 
When we need to do RTCP and we don't have an internal source yet,
make one.
 2013-07-26 12:18:01 +02:00
Wim Taymans
bd0709c15c session: remove old code to change SSRC 
Remove code used to change the SSRC after a collision. We now send
a RECONFIGURE event upstream to make the upstream element change the SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
88f5a5f355 source: don't update packet SSRC 
Remove the code to update the SSRC in packets, it can never be called now that
we always use a source with matching packet SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
abc90da1dc session: delay allocation of internal source 
Allocate the internal source when we receive a caps with the SSRC or when we see
a buffer with the SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
e0a1ce1291 session: generate reconfigure on collision 
When we detect a collision, change the SSRC that we suggest upstream
and trigger RECONFIGURE. This should make upstream select a new SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
495d43c089 session: produce RTCP for all internal sources 
Loop over all the internal sources and produce RTCP. We also need
to queue the RTCP packets and send them when we are finished.
 2013-07-26 12:18:00 +02:00
Wim Taymans
9505fd4150 session: deprecate internal source and ssrc properties 
Deprecate the internal source and internal ssrc properties. There might
be more than one internal source.
 2013-07-26 12:17:59 +02:00
Wim Taymans
3d6ee1fb5e session: internal sources don't use probation 2013-07-26 12:17:59 +02:00
Wim Taymans
0e53e9109e session: give caps to session 
Let the session parse the caps and update its SSRC when needed.
 2013-07-26 12:17:59 +02:00
Wim Taymans
c06482a2cb session: make method to suggest available SSRC 
Make a method to suggest the best available SSRC. This is the SSRC of the last
created internal source and is used to instruct upstream to produce this
SSRC.
 2013-07-26 12:17:59 +02:00
Wim Taymans
33ce50e8b1 session: keep SDES and set on new internal sources 
Keep track of the SDES ourselves and set it on all newly created
internal sources.
 2013-07-26 12:17:59 +02:00
Wim Taymans
5652f02b76 session: make method to make internal sources 
Add a method to obtain an internal source and use it to create
our internal source
 2013-07-26 12:17:59 +02:00
Wim Taymans
7f83927c95 session: count internal sources and how many are senders 2013-07-26 12:17:58 +02:00
Wim Taymans
719343c206 rtpsession: separate BYE marking and scheduling 
First mark sources with BYE and then schedule the BYE RTCP message.
 2013-07-26 12:17:58 +02:00
Wim Taymans
391943ba82 session: get SSRC from RTCP packet itself 
Get the SSRC from the RTCP packet instead.
 2013-07-26 12:17:57 +02:00
Wim Taymans
a3f75a17ef session: fix bandwidth calculation 
We iterate over all sources and the internal one is also in the
hashtable so avoid adding it twice.
 2013-07-26 12:17:57 +02:00
Wim Taymans
9eaef9d332 session: add some docs 2013-07-26 12:17:56 +02:00
Wim Taymans
2163355a47 session: Rearrange RTCP reporting a little 
Make a function to generate an RTCP packet for a source, pass the source as a
parameter.
Move timeout of collisions to session cleanup phase.
 2013-07-26 12:17:56 +02:00
Wim Taymans
a3bf374351 session: move check for is_early around 
Move the check for the early RTCP to where it is needed and used.
 2013-07-26 12:17:56 +02:00
Wim Taymans
b069db6a2e session: parse packet outside of the session lock 2013-07-26 12:17:56 +02:00
Wim Taymans
57c27ec319 session: do nicer checks for internal sources 2013-07-26 12:17:56 +02:00
Wim Taymans
93d07298ff session: let source keep track if it sent BYE 2013-07-26 12:17:56 +02:00
Wim Taymans
0c9c1434a8 source: reset more 2013-07-26 12:17:56 +02:00
Wim Taymans
1d02496d15 source: also use the source for bye_reason 
Store the BYE reason in our internal source object. Rename the methods on the
source object a little because now the BYE can be received in RTCP or
set when the session wants to send BYE.
 2013-07-26 12:17:56 +02:00
Wim Taymans
ddd071e54c session: configure sdes with structure only 
Remove code to configure the SDES with methods and types, only
allow configuration with GstStructure
 2013-07-26 12:17:55 +02:00
Wim Taymans
0060e1d45d session: refactor add and find source 
Make functions to find and add a source to the hashtable.
 2013-07-26 12:17:55 +02:00
Wim Taymans
adb0d68c07 session: remove source from sync_rtcp 
We don't need to know the sender source of the session in the
callback, the SR packet is for all participants in the session.
 2013-07-26 12:17:55 +02:00
Wim Taymans
bf7d8173b3 jitterbuffer: add some more debug 2013-07-26 12:17:55 +02:00
Vincent Penquerc'h
91d4abceaa aacparse: allow conversion from ADTS to raw AAC 
Some muxers (eg, qtmux) only support raw AAC, so this allows linking
an encoder that outputs ADTS only to those muxers.

The conversion is simple (omit the first 7 or 9 bytes of the frame),
but has to be done in pre_push instead of handle_frame as 1.0 does
not seem to allow skipping bytes there as 0.10 used to.

Other conversions are not supported (yet).
 2013-07-26 09:44:11 +01:00
Vincent Penquerc'h
55e9338846 aacparse: fix object_type parsing off-by-one in ADTS frame 
According to http://wiki.multimedia.cx/index.php?title=ADTS,
the value stored in ADTS headers is one less than the object
type of the AAC stream.

A look at ffmpeg shows it also adds 1 to the value read off
the ADTS header.

Note that this might break other things that happen to have
an inverse off by one to match the existing code.
 2013-07-26 09:44:10 +01:00
Thiago Santos
7eac4c7c03 avidemux: fix seqnum handling for seeks 
Use the same seqnum as the seek for flushes/segments that are
caused by the seek. Also do the same for segment events

Fixes #676242
 2013-07-25 15:24:31 -03:00
Thiago Santos
8bd12e12b3 matroskademux: fix seqnum handling for seeks 
Use the same seqnum as the seek for flushes/segments that are
caused by the seek. Also do the same for segment events

Fixes #676242
 2013-07-25 15:24:31 -03:00
Thiago Santos
e49b6e7c35 qtdemux: correctly handle seqnum for seeks and segments 
Use the same seqnum on messages and events for derived events.
Fixed for flushes / stream-start / segment after a seek, and segment
after a segment.

Fixes #676242
 2013-07-25 15:24:31 -03:00
Wim Taymans
c44a29bd53 bin: fix compilation 2013-07-24 14:17:45 +02:00
Wim Taymans
cc92ef1db2 vrawdepay: fix UYVP format 2013-07-24 12:42:31 +02:00
Wim Taymans
8191b6fcd2 vrawpay: fix UYVP format 2013-07-24 12:41:58 +02:00
Wim Taymans
37af93c361 vrawpay: fix caps 2013-07-24 12:41:44 +02:00
Wim Taymans
f87875e35b rtpjitterbuffer: fix locking 
Take the lock earlier so that we do things that follow with the right
locking.
 2013-07-24 10:49:03 +02:00
Wim Taymans
dece8413ef rtpsession: don't use invalid times in RTCP timeouts 
An invalid timeout can be calculated when we disabled RTCP by setting the
bandwidth to 0. Make sure all code can handle this case.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626
 2013-07-23 17:41:48 +02:00
Wim Taymans
25e0f0d6b6 rtpsession: lock session when changing bandwidth 
Take the session lock when changing the bandwidth properties so that we don't
end up with inconsistent behaviour.
 2013-07-23 17:41:48 +02:00
Wim Taymans
c337265ee4 session: reset some RTCP variables 
The early_send time was set to 0 and always triggering an early RTCP packet.
 2013-07-23 17:41:48 +02:00
Edward Hervey
3d48d25756 qtdemux: Add all the mpeg XDCAM variants 
This should cover all known XDCAM variants (which are all mpeg2 video)

Fixes #672227
 2013-07-23 15:03:31 +02:00
Carlos Rafael Giani
95429f1d4b rtpbin: added custom downstream sync event 
rtpbin can now send a custom in-band downstream event which informs
downstream that the bin has received an RTCP SR packet. This is useful
for applications which want to drop the initial unsynchronized received
RTP packets.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
 2013-07-23 06:25:20 +02:00
Tim-Philipp Müller
f18b1f7e80 deinterlace: fix on-the-fly changing of "mode" and "fields" properties 
We call setcaps() to reconfigure ourselves, but we need to pass
the current *sink* caps, not the source caps then. Also fix a
caps leak.

https://bugzilla.gnome.org/show_bug.cgi?id=641599
 2013-07-22 18:00:16 +01:00
Sebastian Dröge
0c2ff91a5c wavparse: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
169b490664 rtspsrc: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
5a9f4a3cbc rtpsession: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
57dd1189d5 matroskademux: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
1a0278ed64 qtdemux: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +02:00
Sebastian Dröge
1122698491 flvdemux: Add support for group-id in the stream-start event 2013-07-22 15:30:12 +02:00
Sebastian Dröge
6cc16da531 avidemux: Add support for group-id in the stream-start event 2013-07-22 15:30:12 +02:00
Mathieu Duponchelle
d67a671bfb videomixer: use gst_util_uint64_scale*_round. 
There could be a case where:
      1) you do a new set_caps after buffers have been processed.
      2) ts_offset gets set to a different value, eg 0.033333333
      3) your pads get EOS, but the check dor that doesn't work
         because you use ts_offset + a truncated value < segment.stop
      4) so in the next collected, you end up comparing for example:
      0.9999999999 > 1., which is false and means you don't send EOS.

Also adds scale_round in two other places where it potentially could
have caused problems.
 2013-07-21 19:21:57 -04:00
Olivier Crête
96a8fb92e2 qtdemux: Add WRLE support 2013-07-19 14:58:30 -04:00
Tim-Philipp Müller
aa7d597120 qtdemux: make files from Vivotek camera play 
Skip tracks of 'vivo' subtype with empty stsd instead of
erroring out saying that the file is broken.

https://bugzilla.gnome.org/show_bug.cgi?id=699791
 2013-07-19 19:38:30 +01:00
Tim-Philipp Müller
ce52b319ff qtmux: when streaming don't try to seek when stopping 
It might cause errors in sinks that are not seekable and
have reported this (like e.g. fdsink)

https://bugzilla.gnome.org/show_bug.cgi?id=696228
 2013-07-19 17:31:38 +01:00
Wim Taymans
bdd3c31902 qtdemux: simplify some helpers 
Some helper functions are not needed anymore or can be simplified.
 2013-07-19 17:26:54 +02:00
Wim Taymans
61a8937ced qtdemux: for non-raw video, move palette in caps 
We only need to append the palette to raw video buffers, non-raw video has the
palette in the caps still.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292
 2013-07-19 17:14:46 +02:00
Arnaud Vrac
40ab78825c qtdemux: nitpicking in esds parsing 2013-07-19 14:26:18 +02:00
Arnaud Vrac
d0d25a5e1f qtdemux: set proper caps for mpeg-1 audio 
Remove AAC specific fields from mpeg-1 audio caps, remove assumption
that the mpeg1 audio layer is 3, and set `parsed' field.

https://bugzilla.gnome.org/show_bug.cgi?id=704548
 2013-07-19 14:26:08 +02:00
Arnaud Vrac
5def061d20 qtdemux: remove chapter stream 
Remove all streams that are actually table of contents, since we will
never need the data after parsing them.
 2013-07-18 11:48:12 +02:00
Arnaud Vrac
ae67c13416 qtdemux: send gap event for sparse streams in push mode 
This allows to pre-roll at least if the next subtitle buffer
is far away.
 2013-07-18 11:48:11 +02:00
Arnaud Vrac
1237898351 qtdemux: do not use indexes from sparse stream when seeking in push mode 
This makes seeking more accurate in push mode, since the previous
keyframe on a sparse stream might be far away.
 2013-07-18 11:48:11 +02:00
Arnaud Vrac
e561d12655 qtdemux: advertise subtitle streams as sparse 2013-07-18 11:48:11 +02:00
Arnaud Vrac
6e26f1d067 mastrokademux: do not push discont buffers if they aren't discont 
Unset the discont flag instead of posssibly pushing a buffer with
a flag that's still set.

https://bugzilla.gnome.org/show_bug.cgi?id=682110
 2013-07-17 18:10:11 +01:00
Wim Taymans
4c97701650 qtdemux: extract the palette from stsd 
Sometimes a palette is inside the stsd, extract it instead of always using
the default one
 2013-07-17 15:17:19 +02:00
Sebastian Dröge
9f73447229 goom2k1: Fix event handling and negotiate as soon as possible 2013-07-17 14:30:16 +02:00
Sebastian Dröge
78c7c16e9e goom: Fix event handling and negotiate as soon as possible 2013-07-17 14:28:43 +02:00
Wim Taymans
6b82c89562 qtdemux: add support for WRAW 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292
 2013-07-17 09:57:17 +02:00
Wim Taymans
f698483bb3 qtdemux: palette is appended to buffers, not in caps 
Fix the palette handling, in 1.0 we append the palette to the buffer instead of
placing it on the caps.

See also https://bugzilla.gnome.org/show_bug.cgi?id=704292
 2013-07-17 09:57:16 +02:00
Olivier Crête
54c5a7f690 rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders 2013-07-16 15:37:49 -04:00
Arnaud Vrac
54bba4f60c qtdemux: reset segment on flush stop 
cca2f555d1 introduces a regression, where the demux segment is not
reset on flush stop, so the next upstream segment event will calculate
an invalid base time on the new segment to be sent downstream.

https://bugzilla.gnome.org/show_bug.cgi?id=704255
 2013-07-16 10:47:20 +02:00
Matej Knopp
ca32442f86 qtdemux: offset samples according to edit list 
https://bugzilla.gnome.org/show_bug.cgi?id=700264
 2013-07-15 09:59:23 +02:00
Matej Knopp
ae92ea21a1 aacparse: be less verbose when parsing LOAS streams 
https://bugzilla.gnome.org/show_bug.cgi?id=704162
 2013-07-15 07:55:08 +02:00
Matej Knopp
3111161e8a qtdemux: unselect instead of ignoring disabled track, detect chapter track 
https://bugzilla.gnome.org/show_bug.cgi?id=704007
 2013-07-12 11:45:33 +02:00
Kyosuke Nekomura
4d517e94ef audioecho: Fix handling of delay property in PLAYING/PAUSED state 
https://bugzilla.gnome.org/show_bug.cgi?id=703901
 2013-07-12 09:36:16 +02:00
Olivier Crête
3aa20e7c8d rtpmux: Enable proxy caps on the src pads 2013-07-11 17:21:22 -04:00
Matej Knopp
7b69f427f1 qtdemux: correct argument order in gst_util_uint64_scale_int_round 
https://bugzilla.gnome.org/show_bug.cgi?id=703350
 2013-07-10 09:20:17 +02:00
Olivier Crête
1997acc8b2 rtpmux: Keep caps order from the peer or the filter 2013-07-09 17:43:31 -04:00
Sebastian Dröge
3d0988f46f videomixer: Fix handling of buffers without a duration 
We'll have to pop buffer from collectpads and store it
internally only to get the timestamp of the next buffer.
If we continue to keep it in collectpads, no new buffer
to calculate the end time will ever arrive.

https://bugzilla.gnome.org/show_bug.cgi?id=703743
 2013-07-09 12:42:17 +02:00
Sebastian Dröge
9e9d2ce098 videomixer: Fix negotiation with 0/1 framerates 
https://bugzilla.gnome.org/show_bug.cgi?id=703743
 2013-07-09 11:53:28 +02:00
Jonas Holmberg
beebe2b7af matroskademux: Unlock stream lock after use 
Stream lock of sink pad was not unlocked after non-updating seek.
 2013-07-09 11:25:14 +02:00
Ognyan Tonchev
aa2d96c46b multipartmux: Re-set need_segment flag after FLUSH_STOP 
https://bugzilla.gnome.org/show_bug.cgi?id=703182
 2013-07-09 09:16:20 +02:00
Sebastian Dröge
0cc77d8e30 rtph263ppay: Don't pass upstream filter caps to downstream 
Downstream usually can't accept video/x-h263 but only application/x-rtp,
so we would always get an empty intersection here.

https://bugzilla.gnome.org/show_bug.cgi?id=702632
 2013-07-08 14:10:44 +02:00
Wim Taymans
ab24598443 rtspsrc: avoid some strdup 2013-07-02 11:13:25 +02:00
Wim Taymans
7c950ef3f2 rtspsrc: add select-stream signal 
Add a signal to let the app select what streams will be selected.

See https://bugzilla.gnome.org/show_bug.cgi?id=634419
 2013-07-02 10:40:35 +02:00
Wim Taymans
2d276e1bcb rtspsrc: avoid strdup 2013-07-02 10:40:35 +02:00
J. Rick Ramstetter
f01b751e52 rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin 
https://bugzilla.gnome.org/show_bug.cgi?id=703426
 2013-07-02 10:12:17 +02:00
Wim Taymans
1db7e62060 rtspsrc: add signal to notify of the SDP 
This way, the app can look and modify the SDP.
 2013-07-01 17:31:30 +02:00
Matej Knopp
4053e1d6ac qtdemux: compute framerate from average sample duration 
https://bugzilla.gnome.org/show_bug.cgi?id=703350
 2013-07-01 12:53:17 +02:00
Alban Browaeys
97015d3c93 flvdemux: Add flvversion 1 to the flash-video caps 
This allows using avdec_flv which requires this field to be
present in the caps. FLV only supports flash-video version 1
right now.

https://bugzilla.gnome.org/show_bug.cgi?id=703076
 2013-07-01 11:43:46 +02:00
Sebastian Dröge
5f6469fe2a deinterleave: Don't hold object lock while sending events downstream 
Based on a patch by Kishore Arepalli <kishore.arepalli@gmail.com>

https://bugzilla.gnome.org/show_bug.cgi?id=703114
 2013-07-01 11:37:00 +02:00
Sebastian Dröge
75b5a54f17 matroskademux: Add MPEG4 video profile/level to the caps 2013-07-01 11:01:13 +02:00
Sebastian Dröge
423bddac6a matroskademux: Add AAC profile/level to the caps 
https://bugzilla.gnome.org/show_bug.cgi?id=703312
 2013-07-01 11:01:13 +02:00
Wim Taymans
c469434ea8 vorbispay: add support for config-interval 
Align code with the theora payloader and add support for the config-interval to
periodically send out the config headers.
 2013-06-28 15:21:56 +02:00
Wim Taymans
006562c9f4 theorapay: small cleanups 2013-06-28 15:21:12 +02:00
Wim Taymans
cdc66462ce theorapay: handle streamheaders as well 2013-06-28 12:08:19 +02:00
Wim Taymans
3169432ed4 vorbispay: always collect headers on data 
When we see a data packet, always check if we need to collect any previous
headers.
 2013-06-28 12:07:58 +02:00
Wim Taymans
6c716dfc25 vorbispay: handle streamheader as well 
Take config strings from the streamheader when we can

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=664312
 2013-06-28 11:43:17 +02:00
David Svensson Fors
692206d3a7 rtph264pay: avoid double buffer unmap on error 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703171
 2013-06-27 17:14:11 +02:00
Wim Taymans
3289a2963b rtspsrc: reset-sync before play 
Call reset-sync on the rtpbin before we go to playing. This makes us require SR
packets for all streams again before we attempt to sync them. If we don't reset,
it might be that we combine SR packets from before and after the PAUSE/PLAYING
state change and end up with huge bogus offsets.
 2013-06-27 17:02:14 +02:00
Wim Taymans
519305d14d jitterbuffer: improve sync on first packets 
Don't throw away the first RTCP packet if it arrives before the first
RTP packet but remember and use it to signal sync once we get the
RTP packet.

See https://bugzilla.gnome.org/show_bug.cgi?id=691400
 2013-06-27 16:23:20 +02:00
Wim Taymans
8969f00661 jitterbuffer: only signal loop when active 
Only signal the loop function when it is active.
 2013-06-27 16:15:45 +02:00
Wim Taymans
4bd2ffb26e jitterbuffer: signal timestamp discont 
We can now use the RESYNC buffer flag to mark a timestamp discont when we update
the ts-offset property.
 2013-06-27 16:13:37 +02:00
Wim Taymans
4258ddcc36 jpegpay: turn some errors into warnings 
Turn some errors into warnings, we can continue processing so this should
not be fatal.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=657079
 2013-06-26 20:49:41 +02:00
Wim Taymans
bb9d42b976 rtspsrc: avoid some flushes 2013-06-26 14:58:53 +02:00
Wim Taymans
f39ef2ab68 rtspsrc: handle data message when waiting for reply 
When we are waiting for a server reply, handle data messages instead of
ignoring them.
 2013-06-26 14:41:36 +02:00
Wim Taymans
61219dc6ed rtspsrc: handle data messages in separate method 
Refactor and make a method to handle a data message.
 2013-06-26 14:41:36 +02:00
Wim Taymans
a4be0c6de3 rtspsrc: add some more docs to handle-request signal 
See https://bugzilla.gnome.org/show_bug.cgi?id=702705
 2013-06-25 20:36:18 +02:00
Youness Alaoui
52e440c91b Send a clock_provide message on the bus when we get a netclock 2013-06-25 14:50:47 +02:00
Youness Alaoui
547df8e14f rtspsrc: Expose use-pipeline-clock property 2013-06-25 14:50:33 +02:00
Wim Taymans
35f6e79b94 udpsink: bind to the given interface 
Actually call BINDTODEVICE to bind to the interface as given by the
property.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702819
 2013-06-24 17:13:05 +02:00
Sebastian Dröge
3c9aba91dc matroska: Add initial VP9 support 2013-06-21 18:22:13 +02:00
Youness Alaoui
95906b8f1c rtsp: go back into the loop after doing pause 
After we do a pause request, go back to loop mode so that we can listen
for server messages again.

See https://bugzilla.gnome.org/show_bug.cgi?id=702705
 2013-06-21 10:42:20 +02:00
Olivier Crête
2cd6f53e24 rtpptdemux: Wait after the caps to forward the other events 
First forward the stream-start, then the caps, then the rest
 2013-06-20 23:16:59 -04:00
Wim Taymans
b96d931bf4 rtspsrc: fix race in state change to paused 
When we go to paused, we first flush the connection and then send the pause
command. As a result of the flushing, the scheduled paused command can get
lost. Wait until the connection is completely flushed and the rtsp task is
waiting before issuing the paused or playing request.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702705
 2013-06-20 14:43:47 +02:00
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