Commit graph

13825 commits

Author SHA1 Message Date
Dan Kegel
99bf36b55d v4l2: Remove obsolete definition GST_V4L2_MISSING_BUFDECL
The only use was removed by 9edc0c0365

https://bugzilla.gnome.org/show_bug.cgi?id=723446
2014-02-04 14:06:13 +01:00
Rafał Mużyło
ac4df5e2c5 gst: Don't use endianness-specific S8 audio format
It does not exist.

https://bugzilla.gnome.org/show_bug.cgi?id=723331
2014-02-04 13:44:29 +01:00
Julien Isorce
90b01fce61 cairooverlay: add support for RGB16
https://bugzilla.gnome.org/show_bug.cgi?id=723289
2014-01-31 14:17:54 +00:00
Per x Johansson
46bc1677a4 matroskamux: Fix constantly growing used uid list
Moves the used uid list to the class to avoid having it grow forever.

https://bugzilla.gnome.org/show_bug.cgi?id=723269
2014-01-30 11:59:28 -03:00
Edward Hervey
480ae10884 Automatic update of common submodule
From d48bed3 to 1a07da9
2014-01-30 10:44:05 +01:00
Mike Sheldon
659939f0f0 wavparse: Ignore Broadcast Wave Format (BWF) tags when searching for 'fmt' chunk
https://bugzilla.gnome.org/show_bug.cgi?id=723125
2014-01-29 20:16:48 +01:00
Edward Hervey
ceb602073a check: Use fakesink sync=True instead of an audio sink
Ensures the test can run on systems without alsa (or any audio output for
that matter), and will avoid people running build slaves wondering what
the hell was beeping during the night :)
2014-01-29 10:37:53 +01:00
Mark Nauwelaerts
d25a183ccc ac3parse: custom get_sink_caps handling for private stream caps
... now that those are transformed rather than parsed, some transforming
of caps is required as well to make auto-plugging succeed.
2014-01-27 20:07:41 +01:00
Ryan Lortie
a46f667853 v4l2: guard use of ENODATA with #ifdef
Not all systems with v4l have ENODATA defined, so check that we have it
before attempting to use it.

https://bugzilla.gnome.org/show_bug.cgi?id=722953
2014-01-25 18:51:41 +01:00
Sebastian Dröge
8054cd5df3 Revert "rtspsrc: Proxy rtpjitterbuffer do-retransmission property"
This reverts commit 9f7b1128b1.

This should be handled automatically be rtspsrc if the AVPF profile
is used, and manual enabling of it can be done with the new-manager
signal.
2014-01-24 12:37:39 +01:00
Wim Taymans
43feb82feb rtspsrc: add signal to notify of new manager
So that you can configure and connect to signals on the rtpbin.

See https://bugzilla.gnome.org/show_bug.cgi?id=722866
2014-01-24 10:22:59 +01:00
Aleix Conchillo Flaqué
9f7b1128b1 rtspsrc: Proxy rtpjitterbuffer do-retransmission property
https://bugzilla.gnome.org/show_bug.cgi?id=722866
2014-01-24 09:14:59 +01:00
Wim Taymans
204bd715d2 rtpjitterbuffer: handle expected packet being an RTX packet
If the expected packet (do_next_seqnum is TRUE) is the one we requested
for retranmission earlier, do the logic to update the retransmission
statistics as well before setting up the timers for the next expected
packet.
Also reset the retransmission counter if the timer is reused for another
seqnum.
2014-01-21 17:52:44 +01:00
Wim Taymans
ddb0b9c422 rtpbin: add a caps accumulator for the request-pt-map signal
Add an accumulator that stops the signal emission as soon as a caps has
been retrieved. Otherwise the default handler would continue emitting
the signal and possibly overwrite the result with NULL again.
2014-01-21 15:48:20 +01:00
Wim Taymans
ef20dfe031 rtxreceive: copy flags and timestamps from original buffer 2014-01-21 15:29:27 +01:00
Wim Taymans
9a3d4d7cbe rtpjitterbuffer: ignore invalid timestamps in rtt calculation
When the input buffer does not have a valid timestamp, don't try to
calculate the round-trip-time.
2014-01-21 15:29:26 +01:00
Reynaldo H. Verdejo Pinochet
cf0c780138 matroskaparse: better default caps when none set
Uses information gathered during EBML parsing to
forge a more suitable set of caps instead of blindly
assuming everything is video/x-matroska.

For consistency, stream type reset was added to
matroska-demux too.

https://bugzilla.gnome.org/show_bug.cgi?id=722311
2014-01-21 11:11:46 -03:00
George Kiagiadakis
016e1562a6 tests: rtprtx::test_rtxreceive_data_reconstruction: remove useless code for triggering retransmission
There is no need anymore to push yet another buffer in rtxsend
in order to trigger the previously requested retransmissions
to actually happen.
2014-01-21 15:00:54 +01:00
George Kiagiadakis
184553151d tests: rtprtx::test_rtxreceive_data_reconstruction: fix race condition
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.

This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
2014-01-21 15:00:53 +01:00
George Kiagiadakis
7677aec2fa tests: rtprtx::test_rtxsender_packet_retention: fix race condition
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.

This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
2014-01-21 15:00:53 +01:00
George Kiagiadakis
7011f98d7e tests: rtprtx::test_push_forward_seq: fix race condition
Now with rtprtxsend pushing rtx buffers from a different thread,
this is necessary to ensure that the result of the test is deterministic.

This code makes use of GstCheck's global GMutex and GCond that are
being used inside GstCheck's sink pad chain() function in order
to synchronize with it.
2014-01-21 15:00:53 +01:00
George Kiagiadakis
c702e37091 tests: rtprtx::test_push_forward_seq: fix buffer refcounting 2014-01-21 15:00:53 +01:00
George Kiagiadakis
1a300eb509 rtprtxsend: ensure that no rtx buffers are sent after EOS
To do that, enqueue the EOS event to be sent from the srcpad task
thread and flush the queue right afterwards, so that no more rtx
buffers can be sent, even if there are more requests coming in.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722370
2014-01-21 15:00:37 +01:00
George Kiagiadakis
133913a11a rtprtxsend: run a new GstTask on the src pad
The reason behind this is to minimize the retransmission delay.
Previously, when a NACK was received, rtprtxsend would put a
retransmission packet in a queue and it would send it from chain(),
i.e. only after a new buffer would arrive.

This unfortunately was causing big delays, in the order of 60-100 ms,
which can be critical for the receiver side.

By having a separate GstTask for pushing buffers out of rtxsend,
we can push buffers out right after receiving the event, without
waiting for chain() to get called.
2014-01-21 14:54:01 +01:00
Tim-Philipp Müller
4480a25844 shout2send: error out if no caps were received
Instead of assuming that input is ogg.
2014-01-21 10:56:19 +00:00
Tim-Philipp Müller
d7210befdc shout2send: accept audio/webm, audio/ogg and video/ogg as well
Those are advertised in the template caps, but the
setcaps handler didn't handle them. But then oggmux
and oggparse at least for now still always output
application/ogg anyway, so that wasn't a real problem.
2014-01-21 00:49:44 +00:00
Sebastian Dröge
e178cf60ae rtpvp8pay: Don't leak input buffers
https://bugzilla.gnome.org/show_bug.cgi?id=722414
2014-01-20 10:13:19 +01:00
Mark Nauwelaerts
829cec51c7 avimux: reset some more audio pad data when needed 2014-01-19 17:53:45 +01:00
Mark Nauwelaerts
3ea338ce27 avimux: write correct blockalign for vbr audio
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720659
2014-01-19 17:53:45 +01:00
Aleix Conchillo Flaqué
cdbb2ba6b8 rtpjitterbuffer: do not drop serialized events when latency is set
Serialized events are now queued in the jitter buffer, so we don't
want to drop them even latency is set.

https://bugzilla.gnome.org/show_bug.cgi?id=722372
2014-01-18 10:38:46 +01:00
Michael Olbrich
447556fe6b avimux: don't make the buffer writable unless absolutely necessary
https://bugzilla.gnome.org/show_bug.cgi?id=722396
2014-01-17 19:25:15 -03:00
Michael Olbrich
3978f6ae6c v4l2: set GST_BUFFER_FLAG_DELTA_UNIT when appropriate
https://bugzilla.gnome.org/show_bug.cgi?id=722394
2014-01-17 10:04:34 -05:00
Michael Olbrich
3d2bd4ff49 v4l2bufferpool: don't ref the newly created allocator
Otherwise the allocator will never be deleted.

https://bugzilla.gnome.org/show_bug.cgi?id=712612
2014-01-17 08:32:28 -05:00
Sebastian Dröge
809d105982 matroskademux: Don't skip all video frames until the first keyframe
Instead do it like all other demuxers and let parsers and decoders
handle that. The keyframe information inside the container might
be completely wrong like in the sample file of the bug report,
and if it is correct and we push no keyframes, then the parsers
and decoders will handle that properly anyway.

https://bugzilla.gnome.org/show_bug.cgi?id=682276
2014-01-15 22:49:58 +01:00
Thiago Santos
52fc078310 qtdemux: remove elst_offset variables
They are not used anymore
2014-01-15 15:33:45 -03:00
Thiago Santos
5fe1b3eb28 qtdemux: remember reverse playback when verifying the segment end
Check if the rate is positive or negative to correctly compare the current
position with the segment to make reverse playback work
2014-01-15 15:33:45 -03:00
Thiago Santos
90a5565229 qtdemux: do not ignore empty segments
Make sure empty segments are used and pushed with a gap event
to represent its data (or lack of it)

Each QtSegment is mapped into a GstSegment with the corresponding
media range. For empty QtSegments a gap event is pushed instead
of GstBuffers and it advances to the next QtSegment.

To make this work with seeks, need to keep track of the starting
'base' to make sure it remains consistently increasing when
pushing new segment events.
For example: if a seek makes qtdemux start from 5s, the first
segment will have a base=0. When the next segment is activated,
its base time will be QtSegment.time - qtdemux.segment_base so
that it doesn't include the first 5s that weren't played and
shouldn't be accounted on the running time

This purposedly will remove the fix made for
https://bugzilla.gnome.org/show_bug.cgi?id=700264, at this
point it was decided to respect the gaps, even if they cause
a delay on playback, because that's the way the file was crafted.

https://bugzilla.gnome.org/show_bug.cgi?id=345830
2014-01-15 15:33:45 -03:00
Olivier Crête
8a143dfcbc tests: Remove usage of the system clock from the rtprtx test 2014-01-15 10:13:12 +01:00
Olivier Crête
f0a4f26fa7 tests: Initial segment in rtpcollision test 2014-01-15 10:13:12 +01:00
George Kiagiadakis
a7823bc522 examples/*-rtpaux: specify payload type association for the audio stream, so that rtx works also for audio 2014-01-15 10:13:12 +01:00
George Kiagiadakis
397c4ed7a0 rtprtxsend: remove wrong check for payload type not having been set
1) pt can be lower than 96
2) there is no point in checking that because rtprtxsend will not
   even store buffers for payload types that it doesn't know about,
   so this case will never be reached
2014-01-15 10:13:12 +01:00
George Kiagiadakis
55746eaa4c rtprtxsend: fix data locking when creating rtx packets
This patch moves the creation of rtx packets to be done early,
in the src_event() function, when they are requested. The purpose
is to run gst_rtp_rtx_buffer_new() with the object locked to
protect internal data, because if it is done at the pushing stage,
we would have to lock and unlock multiple times in a row while we
are pushing the rtx buffers.

Previously there was no locking at all, which was terribly wrong.
2014-01-15 10:13:11 +01:00
George Kiagiadakis
3d9ca102c9 rtprtxsend: lock access to internal data in sink_event() function 2014-01-15 10:13:11 +01:00
George Kiagiadakis
ee8ae3000e rtprtxsend: remove unnecessary call to reset() from finalize()
...and use _free_full() on the pending buffers queue now that
reset() is not being called
2014-01-15 10:13:11 +01:00
George Kiagiadakis
f9f7e6e721 rtprtxsend: remove unused parameter from the internal reset() method 2014-01-15 10:13:11 +01:00
George Kiagiadakis
6d588ad6bb rtprtxsend: Use g_slice_* for allocating internal structures 2014-01-15 10:13:11 +01:00
George Kiagiadakis
75859ae924 rtprtxreceive: remove stupid mutex unlock in the middle of chain() 2014-01-15 10:13:11 +01:00
George Kiagiadakis
bf347dc50c rtprtxreceive: use GST_DEBUG_OBJECT / GST_WARNING_OBJECT instead of GST_DEBUG / g_warning 2014-01-15 10:13:11 +01:00
George Kiagiadakis
47788929d3 rtprtxreceive: fix integer format specifiers in GST_DEBUG
seqnum in this function is 32-bit, so G_GUINT16_FORMAT would
produce undefined output on big endian systems
2014-01-15 10:13:11 +01:00
George Kiagiadakis
252dfc34c8 rtprtxsend: change the rtx_pt_map directly in set_property() instead of delaying it for chain()
The same lock is held, so there is no point in complicating it...
2014-01-15 10:13:11 +01:00