Commit graph

14593 commits

Author SHA1 Message Date
Sebastian Dröge
543c8772cb aacparse: Fix unit test now that we always have profile/level in the caps 2014-10-27 11:08:40 +01:00
Tim-Philipp Müller
52e4bbab9e Parallelise 'make check-valgrind'
Some of the RTP unit tests are very flaky and will
fail more often with the CPU maxed out fully. Those
tests need to be fixed in any case though, they also
fail on slower machines and also occasionally with
normal 'make check'.
2014-10-26 15:01:01 +00:00
Sebastian Dröge
4aac09e708 aacparse: Always set profile/level on the caps
We have the information already, so why not use it?
2014-10-26 11:47:25 +01:00
Tim-Philipp Müller
b02d73a0ed rtpjitterbuffer: fix crash on some 32-bit systems
Make sure to pass right number of bits to gst_structure_new()
which is a vararg function.

Fixes elements/rtpaux unit test on ppc32.
2014-10-25 12:45:31 +01:00
Tim-Philipp Müller
146702e226 tests: fix rgvolume test on big-endian systems 2014-10-25 11:09:57 +01:00
Tim-Philipp Müller
e6f6d9045c tests: fix mulawdec/mulawenc test for big endian systems 2014-10-25 11:09:57 +01:00
Tim-Philipp Müller
401782c19d interleave: intersect result with filter caps in caps query
Fixes crash in audiotestsrc because of an unsupported format
getting negotiated on big-endian systems with
audiotestsrc ! interleave ! audioconvert ! wavenc
2014-10-25 11:08:48 +01:00
Tim-Philipp Müller
666b596aa2 pulse: remove some unused typedefs 2014-10-23 15:46:29 +01:00
Ananda
ec3af50cc2 speex: Fix segfault when resetting the codecs multiple times
https://bugzilla.gnome.org/show_bug.cgi?id=738793
2014-10-23 10:30:26 +02:00
Arun Raghavan
163155715f pulsesink: Temporarily disable stream status posting
We need a mechanism in PulseAudio to allow running code outside the
mainloop lock. Then we'd be able to post to the bus (taking the
GST_OBJECT_LOCK), without worrying about locking order with the mainloop
lock, which is the current cause of deadlocks while trying to post the
stream status messages.

https://bugzilla.gnome.org/show_bug.cgi?id=736071
2014-10-22 23:12:38 +05:30
Wim Taymans
bd09dc96e9 rtpjitterbuffer: limit the retry frequency
When the RTT and jitter are very low (such as on a local network), the
calculated retransmission timeout is very small. Set some sensible lower
boundary to the timeout by adding a new property. We use the packet
spacing as a lower boundary by default.
2014-10-22 15:04:24 +02:00
Miguel París Díaz
4b5243c43d gstrtpjitterbuffer: add "rtx-min-delay" property
This property is useful to set a min time to wait before sending a
retransmission event.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=735378
2014-10-22 15:00:27 +02:00
Wim Taymans
0b81b316b5 jitterbuffer: Refactor code
Refactor some code dealing with calculating various timeouts.

See https://bugzilla.gnome.org/show_bug.cgi?id=735378
2014-10-22 14:59:57 +02:00
Miguel París Díaz
e6504e3a65 rtpsession: fix Early Feedback Transmission
In early retransmission we are allowed to schedule 1 regular RTCP packet
at an earlier time. When we do that, we need to set allow_early to FALSE
and ignore/drop (or merge) all future requests for early transmission.
We now first check if we can schedule an early RTCP and if we can,
actually prepare the data for the next RTCP interval.

After we send the next regular RTCP after the early RTCP, we set
allow_early to TRUE again to allow more early requests.

Remove the condition for the immediate feedback for now.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
2014-10-22 13:13:47 +02:00
Tim-Philipp Müller
59b2f6e895 Automatic update of common submodule
From a8c8939 to 84d06cd
2014-10-21 13:01:32 +01:00
Wim Taymans
09f179139d rtpjitterbuffer: make debug line less confusing 2014-10-21 13:10:53 +02:00
Stefan Sauer
6a3a85f4b2 Automatic update of common submodule
From 36388a1 to a8c8939
2014-10-21 12:58:13 +02:00
Wim Taymans
2e7f5c08cf jitterbuffer: rework resync handling
Add a need-resync state, this is when we need to try to lock on to a
time/RTPtime pair.
Always check the RTP timestamps and if they go backwards, mark ourselves
as need-resync.
Only resync when need-resync is TRUE and we have a valid time. Otherwise
we keep the old values. This avoids locking on to an invalid time and
causing us to timestamp everything with -1.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
2014-10-21 11:57:34 +02:00
Aleix Conchillo Flaqué
bd392d72ee rtspsrc: set full stream caps on internal src TCP pads
Set the complete stream caps on the TCP internal src pads. Otherwise,
ptdemux will not properly detect the caps change.

https://bugzilla.gnome.org/show_bug.cgi?id=737868
2014-10-21 11:33:01 +02:00
Sjoerd Simons
0ee384b251 rtpmux: Don't set PROXY_CAPS flag on the src pad
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.

https://bugzilla.gnome.org/show_bug.cgi?id=738722
2014-10-21 10:52:00 +02:00
Vineeth T M
1131db8c1f videobox: critical error when element properties set as max/min
left, right, top, bottom can be set from range of -2147483648 to 2147483647
when i launch the videobox element with that values, it gives a critical error

(gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed
This happens because min cannot be equal to max.

https://bugzilla.gnome.org/show_bug.cgi?id=738838
2014-10-20 12:53:51 +02:00
Tim-Philipp Müller
f3fec86bc9 Revert "rtp: add h265 RTP payloader + depayloader"
This reverts commit d06ba9051f.

This breaks the build, as it depends on parser API in -bad.
2014-10-15 17:48:46 +01:00
Jurgen Slowack
d06ba9051f rtp: add h265 RTP payloader + depayloader 2014-10-15 17:34:50 +02:00
Peter G. Baum
b5e46c05d7 wavenc: Support RF64 format
https://bugzilla.gnome.org/show_bug.cgi?id=725145
2014-10-14 10:24:50 +02:00
David Sansome
8154c90c9b equalizer: Don't call iirequalizer's transform_ip in passthrough mode
It tries to map the read-only buffer with GST_MAP_READWRITE and crashes.

https://bugzilla.gnome.org/show_bug.cgi?id=737886
2014-10-13 08:30:03 +02:00
Olivier Crête
51a8bedced rtpsource: Rename seqnum-base to seqnum-offset in caps
This was modified back in 1.0 in GstRtpBasePayload
2014-10-10 18:33:34 -04:00
Olivier Crête
155ed569c3 rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-10-10 18:12:32 -04:00
Olivier Crête
b3069634bd rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
These were renamed in GstRTPBasePayload in 1.0
2014-10-10 18:12:23 -04:00
Luis de Bethencourt
cff880401d goom2k1: removing block of code that does nothing
The loop in zoomFilterSetResolution is meant to change the values in the
zf->firedec[] array. Each iteration writes the value of decc onto the arrya,
but no conditions that change the value of decc are ever met and the array is
filled with zero for each element. Which is the initial state of the
array before the loop begins.

The loop does nothing.

https://bugzilla.gnome.org/show_bug.cgi?id=728353
2014-10-08 14:07:56 +01:00
Stefan Sauer
98222a67ff rtpjitterbuffer: don't log all clock_rate changes as warnings.
We never initialize clock_rate explicitly, therefore it is 0 by default. The
parameter is a uint32 and the only caller ensure that it is >0, therefore it
won't become -1 ever.
2014-10-04 17:17:13 +02:00
Nirbheek Chauhan
f35f3ccf7c souphttpclientsink: Fix lifetime of stream headers and queued buffers
Stream headers are updated whenever ::set_caps is called, so we can't assume
they'll be valid before the message body is written out. We *can* assume that
for queued buffers, but SOUP_MEMORY_STATIC is still wrong for those.

Also, add some debug logging for stream header interactions.

https://bugzilla.gnome.org/show_bug.cgi?id=737771
2014-10-02 12:47:36 +03:00
Matej Knopp
e1d275cfec aacparse: fix memory leak when prepending ADTS headers
https://bugzilla.gnome.org/show_bug.cgi?id=737761
2014-10-02 10:41:28 +03:00
Antonio Ospite
7ae7f657fa interleave: interleave samples following the Default Channel Ordering
In order to have a full mapping between channel positions in the audio
stream and loudspeaker positions, the channel-mask alone is not enough:
the channels must be interleaved following some Default Channel Ordering
as mentioned in the WAVEFORMATEXTENSIBLE[1] specification.

As a Default Channel Ordering use the one implied by
GstAudioChannelPosition which follows the ordering defined in SMPTE
2036-2-2008[2].

NOTE that the relative order in the Top Layer is not exactly the same as
the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users
using so may channels are already aware of such discrepancies.

[1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx
[2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf

Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127
2014-10-02 10:21:26 +03:00
Sebastian Dröge
7729f4ce81 wavenc: Send CAPS event after the pad was activated
Otherwise the CAPS event will be dropped and we never configure any caps at
all, leading to weird behaviour in many situations. Especially header
rewriting is not going to work if a capsfilter is after wavenc.

https://bugzilla.gnome.org/show_bug.cgi?id=737735
2014-10-02 10:10:11 +03:00
Nirbheek Chauhan
374552a720 souphttpclientsink: Add some more useful debug logging 2014-10-02 09:48:49 +03:00
Nirbheek Chauhan
745d497318 souphttpclientsink: Free queued buffers in ::reset
::render sets a new callback for writing out new buffers only if there aren't
already buffers queued for writing with a previously-scheduled callback.
However, if the previously-scheduled callback is interrupted by a state change
(either manually or due to an error) and there are still buffers in the queue,
restarting the pipeline will result in buffers being queued forever, and no
callbacks will ever be scheduled, and no buffers will be written out.

https://bugzilla.gnome.org/show_bug.cgi?id=737739
2014-10-02 09:48:27 +03:00
Sebastian Dröge
1a2adf5123 videomixer: Actually use the correct GstVideoInfo for conversion 2014-10-01 17:29:29 +03:00
Sebastian Dröge
c1a96113db videomixer: Revert the last commit and handle resolutions differences properly
This is about converting the format, not about converting any widths and
heights. Subclasses are expected to handler different resolutions themselves,
like the videomixers already do properly.
2014-10-01 17:24:59 +03:00
Sebastian Dröge
af7916ca4a videomixer: GstVideoConverter currently can't rescale and will assert
Leads to ugly assertions instead of properly erroring out:
CRITICAL **: gst_video_converter_new: assertion 'in_info->width == out_info->width' failed
2014-10-01 17:12:59 +03:00
Sebastian Dröge
23a3377b1e vp8enc/vp9enc: Protect the encoder with a mutex in all situations 2014-09-30 11:35:42 +03:00
Sebastian Dröge
df053c997c vp9enc: Allow caps renegotiation
https://bugzilla.gnome.org/show_bug.cgi?id=726329
2014-09-30 11:35:42 +03:00
Sebastian Dröge
ced5d657e3 vp8enc: finish() and drain() should return a GstFlowReturn 2014-09-30 11:35:42 +03:00
Jose Antonio Santos Cadenas
a2e2012ae3 vp8enc: Allow caps renegotiation
https://bugzilla.gnome.org/show_bug.cgi?id=726329
2014-09-30 11:35:35 +03:00
Aurélien Zanelli
cfb4c02187 v4l2object: set colorspace for output devices
When the v4l2 device is an output device, the application shall set the
colorspace. So map GStreamer colorimetry info to V4L2 colorspace and set
on set_format. In case we have no colorimetry information, we try to
guess it according to pixel format and video size.

https://bugzilla.gnome.org/show_bug.cgi?id=737579
2014-09-29 21:23:01 -04:00
Arun Raghavan
2a3adec2f7 pulse: Add some documentation about threading and synchronisation
This gives a quick introduction to how the pulsesink/pulsesrc code
interacts with the pa_threaded_mainloop that we start up to communicate
with the server.
2014-09-30 06:28:50 +05:30
Arun Raghavan
0ed08ac3fd pulsesink: Make emitting stream status messages synchronous
The stream status messages are emitted in the PA mainloop thread, which
means the mainloop lock is taken, followed by the Gst object lock (by
gst_element_post_message()). In all other locations, the order of
locking is reversed (this is unavoidable in a bunch of cases where the
object lock is taken by GstBaseSink or GstAudioBaseSink, and then we get
control to take the mainloop lock).

The only way to guarantee that the defer callback for stream status
messages doesn't deadlock is to either stop posting those messages, or
make sure that the message emission is completed before we proceed to
any point that might take the object lock before the mainloop lock
(which is what we do after this patch).

https://bugzilla.gnome.org/show_bug.cgi?id=736071
2014-09-30 06:28:50 +05:30
Antonio Ospite
eca3e2474d wavenc: print channel masks in hexadecimal 2014-09-29 17:45:59 +03:00
Tim-Philipp Müller
46df0cedb7 v4l2: remove redundant struct declaration 2014-09-27 16:01:21 +01:00
Sebastian Dröge
d1c7f2e4d1 rtspsrc: Fix compiler warnings
gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
      'GstRTSPResult' [-Werror,-Wenum-conversion]
    res = gst_sdp_message_new (&sdp);
        ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
      'GstRTSPResult' [-Werror,-Wenum-conversion]
    res = gst_sdp_message_parse_uri (uri, sdp);
        ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
2014-09-26 13:46:16 +03:00
Jonas Holmberg
1371fa0c61 matroskademux: make demuxer reusable
Remove pads from flow combiner and reset last
flow return to FLOW_OK by resetting the flow combiner.
This prevents FLOW_FLUSHING when trying to re-use the
demuxer after setting it back to NULL/READY state.

https://bugzilla.gnome.org/show_bug.cgi?id=737359
2014-09-25 16:14:18 +01:00