Sebastian Dröge [Mon, 9 Sep 2013 12:44:58 +0000 (14:44 +0200)]
v4l2bufferpool: Also fix strides for other semi-planar video formats
Andreea Fulger [Mon, 9 Sep 2013 12:41:42 +0000 (14:41 +0200)]
v4l2bufferpool: Fix stride for NV12/NV21
https://bugzilla.gnome.org/show_bug.cgi?id=707758
Matej Knopp [Sat, 7 Sep 2013 14:37:03 +0000 (16:37 +0200)]
matroskademux: fix leaking buffer and caps
https://bugzilla.gnome.org/show_bug.cgi?id=707688
Tim-Philipp Müller [Thu, 5 Sep 2013 18:46:37 +0000 (19:46 +0100)]
udpsrc: fix build on win32
gstudpsrc.c:855:15: error: #if with no expression
Wim Taymans [Wed, 4 Sep 2013 13:50:42 +0000 (15:50 +0200)]
avidemux: handle unseekable streams
Handle streams that we can't seek in and ignore them in the
seek logic.
Wim Taymans [Wed, 4 Sep 2013 13:25:39 +0000 (15:25 +0200)]
avidemux: only check video compression for video streams
Or else we might deref a stream with a NULL strf.vids and segfault
Alex Ashley [Tue, 18 Jun 2013 12:27:20 +0000 (13:27 +0100)]
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
Mathieu Duponchelle [Tue, 3 Sep 2013 22:27:50 +0000 (00:27 +0200)]
videomixer: Don't set EOS to FALSE when the collectpad *is* EOS
https://bugzilla.gnome.org/show_bug.cgi?id=707238
Matej Knopp [Tue, 3 Sep 2013 15:32:41 +0000 (17:32 +0200)]
flacparse: cleanup on error after state change
https://bugzilla.gnome.org/show_bug.cgi?id=707229
Sebastian Dröge [Tue, 3 Sep 2013 09:23:24 +0000 (11:23 +0200)]
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
Matej Knopp [Tue, 3 Sep 2013 08:10:01 +0000 (10:10 +0200)]
flacparse: Free GstBaseParseFrame if pushing a header failed
Sebastian Dröge [Mon, 2 Sep 2013 14:02:37 +0000 (16:02 +0200)]
udpsrc: Refactor address resolval into its own function
Tim-Philipp Müller [Mon, 2 Sep 2013 22:00:29 +0000 (23:00 +0100)]
replaygain: fix taglist leak in rganalysis
And add some FIXMEs.
Tim-Philipp Müller [Mon, 2 Sep 2013 21:50:58 +0000 (22:50 +0100)]
tests: rganalysis: rename function for clarity
Christoph Reiter [Mon, 18 Mar 2013 13:32:07 +0000 (14:32 +0100)]
tests: fix skipped rganalysis tests
In 0.10 elements would post tag messages on the bus
directly, and rganalysis would only post a tag message
when it changed tags. In 1.0, only sinks post tag
messages when they receive the serialised tag event.
This means that we get an additional tag message on
the bus now where we didn't expect one before.
https://bugzilla.gnome.org/show_bug.cgi?id=695090
Sebastian Dröge [Mon, 2 Sep 2013 09:46:52 +0000 (11:46 +0200)]
flacparse: Properly propagate downstream flow returns upstream
https://bugzilla.gnome.org/show_bug.cgi?id=707229
Tim-Philipp Müller [Sun, 1 Sep 2013 20:18:38 +0000 (21:18 +0100)]
Don't use setlocale in plugins()
Only apps should call setlocale(), not libraries.
Wim Taymans [Thu, 29 Aug 2013 11:15:15 +0000 (13:15 +0200)]
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
Bernhard Miller [Wed, 28 Aug 2013 08:51:32 +0000 (10:51 +0200)]
autovideosink: add sync property
https://bugzilla.gnome.org/show_bug.cgi?id=706955
Bernhard Miller [Wed, 28 Aug 2013 05:15:00 +0000 (07:15 +0200)]
autoaudiosink: introduce sync property
https://bugzilla.gnome.org/show_bug.cgi?id=706955
Thiago Santos [Tue, 27 Aug 2013 20:33:40 +0000 (17:33 -0300)]
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 <=
Sebastian Dröge [Wed, 28 Aug 2013 11:26:47 +0000 (13:26 +0200)]
Back to development
Sebastian Dröge [Wed, 28 Aug 2013 10:52:25 +0000 (12:52 +0200)]
Release 1.1.4
Sebastian Dröge [Wed, 28 Aug 2013 10:52:16 +0000 (12:52 +0200)]
Update .po files
Sebastian Dröge [Wed, 28 Aug 2013 10:32:10 +0000 (12:32 +0200)]
po: update translations
Wim Taymans [Tue, 27 Aug 2013 13:25:16 +0000 (15:25 +0200)]
matroska-mux: remove framerate restriction
Remove the framerate restriction on the caps.
Wim Taymans [Tue, 27 Aug 2013 07:38:16 +0000 (09:38 +0200)]
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.
Wim Taymans [Tue, 27 Aug 2013 07:37:33 +0000 (09:37 +0200)]
session: add more debug
Wim Taymans [Tue, 27 Aug 2013 07:34:46 +0000 (09:34 +0200)]
jitterbuffer: fix types of the retransmission event
Wim Taymans [Tue, 27 Aug 2013 07:33:03 +0000 (09:33 +0200)]
jitterbuffer: only timeout EXPECTED timers on gap
Only timeout the EXPECTED timers when we detect a large seqnum gap.
Sebastian Dröge [Mon, 26 Aug 2013 11:47:53 +0000 (13:47 +0200)]
configure.ac: Don't set BZ2_LIBS if bz2 is not found
Wim Taymans [Mon, 26 Aug 2013 09:50:27 +0000 (11:50 +0200)]
rtsession: fix locking
We need to take the session lock when getting and manipulating the
source.
Wim Taymans [Mon, 26 Aug 2013 09:50:13 +0000 (11:50 +0200)]
rtpsession: add some more debug
Mathieu Duponchelle [Tue, 20 Aug 2013 20:12:03 +0000 (22:12 +0200)]
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
Tim-Philipp Müller [Fri, 23 Aug 2013 14:56:43 +0000 (15:56 +0100)]
multipartdemux: propagate discont
Tim-Philipp Müller [Fri, 23 Aug 2013 14:49:47 +0000 (15:49 +0100)]
multipartdemux: remove dynamic sourcpads when going from PAUSED to READY
Tim-Philipp Müller [Fri, 23 Aug 2013 14:29:28 +0000 (15:29 +0100)]
multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last
https://bugzilla.gnome.org/show_bug.cgi?id=637754
Wim Taymans [Fri, 23 Aug 2013 13:47:25 +0000 (15:47 +0200)]
rtxqueue: add property to configure queue size
Wim Taymans [Fri, 23 Aug 2013 10:07:55 +0000 (12:07 +0200)]
tests: add retransmission example
Wim Taymans [Fri, 23 Aug 2013 09:55:02 +0000 (11:55 +0200)]
rtpbin: proxy jitterbuffer do-retransmission property
Michael Olbrich [Fri, 23 Aug 2013 09:17:45 +0000 (11:17 +0200)]
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
Olivier Crête [Mon, 19 Aug 2013 03:32:22 +0000 (23:32 -0400)]
pulsesink: Add property to find out the device currently in use
https://bugzilla.gnome.org/show_bug.cgi?id=590768
Olivier Crête [Mon, 19 Aug 2013 03:31:15 +0000 (23:31 -0400)]
pulsesink: De-duplicate code to get the current sink input info
https://bugzilla.gnome.org/show_bug.cgi?id=590768
Olivier Crête [Mon, 19 Aug 2013 02:27:37 +0000 (22:27 -0400)]
pulsesink: Implement changing the device while playing
https://bugzilla.gnome.org/show_bug.cgi?id=590768
Olivier Crête [Mon, 19 Aug 2013 03:32:22 +0000 (23:32 -0400)]
pulsesrc: Add property to find out the device currently in use
https://bugzilla.gnome.org/show_bug.cgi?id=590768
Olivier Crête [Mon, 19 Aug 2013 03:31:15 +0000 (23:31 -0400)]
pulsesrc: De-duplicate code to get the current source output info
https://bugzilla.gnome.org/show_bug.cgi?id=590768
Olivier Crête [Mon, 19 Aug 2013 02:27:37 +0000 (22:27 -0400)]
pulsesrc: Implement changing the device while playing
https://bugzilla.gnome.org/show_bug.cgi?id=590768
Sebastian Dröge [Thu, 22 Aug 2013 12:55:14 +0000 (14:55 +0200)]
configure: Fix bz2 configure check for Windows
Due to function decorations on Windows AC_CHECK_LIB can't be used to check for bz2.
https://bugzilla.gnome.org/show_bug.cgi?id=465924
Akihiro Tsukada [Fri, 22 Feb 2013 11:57:00 +0000 (20:57 +0900)]
pulsesink: Add support for AAC pass-through
https://bugzilla.gnome.org/show_bug.cgi?id=694445
Kishore Arepalli [Mon, 24 Jun 2013 15:29:37 +0000 (17:29 +0200)]
gdkpixbufoverlay: crashes if any property changes during playback when location property is not set
https://bugzilla.gnome.org/show_bug.cgi?id=702988
Olivier Crête [Wed, 21 Aug 2013 18:54:26 +0000 (14:54 -0400)]
pulse: Share static caps definition between src and sink
The src was also missing 24-bit sample formats
Wim Taymans [Wed, 21 Aug 2013 14:53:59 +0000 (16:53 +0200)]
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.
Wim Taymans [Wed, 21 Aug 2013 14:50:59 +0000 (16:50 +0200)]
session: generate events correctly
Do correct shifting of the bitmask for lost packets.
Wim Taymans [Wed, 21 Aug 2013 14:47:40 +0000 (16:47 +0200)]
rtp: register rtx element better
Sebastian Dröge [Wed, 21 Aug 2013 14:32:50 +0000 (16:32 +0200)]
directsoundsink: WAVEFORMATEX is unsigned for 8 bit integers, and signed for others
Probably fixes
https://bugzilla.gnome.org/show_bug.cgi?id=705477
Tim-Philipp Müller [Wed, 21 Aug 2013 12:03:34 +0000 (13:03 +0100)]
jpegenc: don't ignore return value from _finish_frame()
gst_video_encoder_finish_frame() will return FLOW_OK here if
there's no output buffer.
Wim Taymans [Wed, 21 Aug 2013 10:56:35 +0000 (12:56 +0200)]
jpegdepay: add some more debug
Wim Taymans [Wed, 21 Aug 2013 10:10:00 +0000 (12:10 +0200)]
rtpgstdepay: only push events when they changed
Keep track of the STREAM_START and TAG events and only push them
when they changed.
Wim Taymans [Wed, 21 Aug 2013 08:52:59 +0000 (10:52 +0200)]
rtpgstpay: taglists should not be merged in 1.0
Wim Taymans [Wed, 21 Aug 2013 08:28:50 +0000 (10:28 +0200)]
rtpgstdepay: flush on FLUSH_STOP event
Wim Taymans [Wed, 21 Aug 2013 08:03:52 +0000 (10:03 +0200)]
rtpgstpay: reset on state change
Do full reset on state change to READY
Wim Taymans [Wed, 21 Aug 2013 07:55:20 +0000 (09:55 +0200)]
rtpgstpay: reset on FLUSH_STOP
Clear the adapter and pending buffer list on FLUSH_STOP.
Wim Taymans [Wed, 21 Aug 2013 07:39:30 +0000 (09:39 +0200)]
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.
Wim Taymans [Wed, 21 Aug 2013 07:33:04 +0000 (09:33 +0200)]
rtpgstay: don't use // comments
Youness Alaoui [Thu, 8 Aug 2013 15:55:22 +0000 (11:55 -0400)]
rtspsrc: Fix response argument in handle-request signal
Youness Alaoui [Thu, 8 Aug 2013 15:54:41 +0000 (11:54 -0400)]
rtspsrc: Add sdes property and proxy it to rtpbin
Youness Alaoui [Wed, 7 Aug 2013 13:47:35 +0000 (09:47 -0400)]
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
Youness Alaoui [Fri, 26 Jul 2013 01:12:05 +0000 (21:12 -0400)]
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.
Youness Alaoui [Fri, 26 Jul 2013 01:10:10 +0000 (21:10 -0400)]
rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time
Youness Alaoui [Fri, 26 Jul 2013 01:03:34 +0000 (21:03 -0400)]
rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps
Youness Alaoui [Fri, 26 Jul 2013 00:54:50 +0000 (20:54 -0400)]
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.
Youness Alaoui [Thu, 25 Jul 2013 21:56:38 +0000 (17:56 -0400)]
rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START
Youness Alaoui [Thu, 25 Jul 2013 21:52:16 +0000 (17:52 -0400)]
rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3
Wim Taymans [Tue, 20 Aug 2013 12:36:59 +0000 (14:36 +0200)]
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
Wim Taymans [Tue, 20 Aug 2013 08:26:15 +0000 (10:26 +0200)]
jitterbuffer: update docs
Wim Taymans [Tue, 20 Aug 2013 08:25:17 +0000 (10:25 +0200)]
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.
Wim Taymans [Tue, 20 Aug 2013 06:55:50 +0000 (08:55 +0200)]
jitterbuffer: remove unused variables
Wim Taymans [Mon, 19 Aug 2013 19:10:00 +0000 (21:10 +0200)]
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.
Wim Taymans [Mon, 19 Aug 2013 19:37:44 +0000 (21:37 +0200)]
jitterbuffer: refactor packet spacing calculation
Wim Taymans [Mon, 19 Aug 2013 19:34:38 +0000 (21:34 +0200)]
jitterbuffer: keep track of last seqnum and dts
Wim Taymans [Mon, 19 Aug 2013 19:29:49 +0000 (21:29 +0200)]
jitterbuffer: small cleanups
Wim Taymans [Mon, 19 Aug 2013 19:21:08 +0000 (21:21 +0200)]
jitterbuffer: reset retransmission timers in add/reschedule
Reset the retransmission timers when adding and rescheduling a timer.
Wim Taymans [Mon, 19 Aug 2013 19:12:13 +0000 (21:12 +0200)]
jitterbuffer: rename variables for packet spacing
Wim Taymans [Mon, 19 Aug 2013 12:58:01 +0000 (14:58 +0200)]
jitterbuffer: remove lost timer when we get the packet
When we receive a packet, also remove the LOST timer for it.
Wim Taymans [Mon, 19 Aug 2013 12:56:49 +0000 (14:56 +0200)]
jitterbuffer: expected seqnum must increase
Only update the expected seqnum when it is bigger than the previous expected
seqnum.
Wim Taymans [Mon, 19 Aug 2013 12:55:49 +0000 (14:55 +0200)]
jitterbuffer: add more debug
Wim Taymans [Mon, 12 Aug 2013 14:15:54 +0000 (16:15 +0200)]
rtxqueue: add retransmission queue element
Wim Taymans [Mon, 12 Aug 2013 12:53:33 +0000 (14:53 +0200)]
session: add some docs
Wim Taymans [Tue, 6 Aug 2013 14:29:54 +0000 (16:29 +0200)]
session: handle NACK feedback and generate events
Handle and parse the feedback NACK packets and generate a Retransmission
event for each NACKed packet
Olivier Crête [Mon, 19 Aug 2013 17:19:42 +0000 (13:19 -0400)]
v4l2: Add forward declaration for gst_v4l2_object_get_format_list
Olivier Crête [Mon, 22 Oct 2012 21:58:07 +0000 (17:58 -0400)]
v4l2: De-duplicate caps probing between src and sink
Olivier Crête [Tue, 13 Aug 2013 21:32:17 +0000 (17:32 -0400)]
pulse: Remove unused GstPulseProbe
Olivier Crête [Mon, 19 Aug 2013 16:46:45 +0000 (12:46 -0400)]
v4l2: Use G_DEFINE_ macros for added thread safety
Thibault Saunier [Sat, 17 Aug 2013 09:28:13 +0000 (11:28 +0200)]
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.
Wim Taymans [Fri, 16 Aug 2013 15:10:31 +0000 (17:10 +0200)]
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.
Sebastian Dröge [Fri, 16 Aug 2013 11:26:28 +0000 (13:26 +0200)]
flacenc: Properly set headers via the base class instead of just pushing them downstream
Prevents buffers from being send before the caps and segment events.
Chris Bass [Thu, 15 Aug 2013 09:59:10 +0000 (10:59 +0100)]
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
Sebastian Dröge [Thu, 15 Aug 2013 13:08:05 +0000 (15:08 +0200)]
ext: Use new flush vfunc of video codec base classes and remove reset implementations
Wim Taymans [Wed, 14 Aug 2013 14:19:32 +0000 (16:19 +0200)]
jitterbuffer: forward flush before stopping dataflow
First forward the flush event and then stop our loop function.
Tim-Philipp Müller [Wed, 14 Aug 2013 12:10:32 +0000 (13:10 +0100)]
configure: require libsoup >= 2.38
Bump libsoup requirement for newer API used, like headers_get_one().
2.38 is from early 2012 and is in linen with our GLib requirement.