platform/upstream/gstreamer.git
4 years agortpjitterbuffer: don't use the timer-object after JBUF_UNLOCK
Havard Graff [Mon, 17 Feb 2020 14:03:28 +0000 (15:03 +0100)]
rtpjitterbuffer: don't use the timer-object after JBUF_UNLOCK

It could have been freed (rtp_timer_free) in the meantime.

4 years agortpmanager: Google Transport-Wide Congestion Control RTP Extension
Havard Graff [Sat, 29 Jun 2019 16:06:11 +0000 (18:06 +0200)]
rtpmanager: Google Transport-Wide Congestion Control RTP Extension

Generating and parsing the RTCP-messages described in:
https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01

4 years agortpfunnel: various cleanups
Håvard Graff [Fri, 14 Feb 2020 10:08:05 +0000 (10:08 +0000)]
rtpfunnel: various cleanups

* Organize GstRtpFunnelPad and GstRtpFunnel separately
* Use G_GNUC_UNUSED instead of (void) casts
* Don't call an event "caps"
* Use semicolons after GST_END_TEST (helps gst-indent)

4 years agoqtdemux: Merge sample tables for raw audio streams with one container sample per...
Sebastian Dröge [Wed, 29 Jan 2020 21:51:45 +0000 (23:51 +0200)]
qtdemux: Merge sample tables for raw audio streams with one container sample per audio sample

Instead of having chunks with one sample per raw audio sample, have
chunks with a single sample that contains lots of raw audio samples. If
necessary these are still split again later when reading the stream.

With this we are allocating a lot less memory for the parsed sample
tables and can play files that previously triggered our limit of 200MB
for the sample table. For example, one file here would previously
allocate 3.5GB for the sample table and now only allocates 70KB.

4 years agoqtdemux: Add a minimum buffer size for raw audio to not output one buffer per frame
Sebastian Dröge [Mon, 13 Jan 2020 09:55:42 +0000 (11:55 +0200)]
qtdemux: Add a minimum buffer size for raw audio to not output one buffer per frame

Outputting 48000 buffers per second is not a good idea performance-wise.
If a container sample is less than 1024 raw audio frames, combine
multiple samples to get at least 1024 raw audio samples as long as
they're stored contiguous in the file.

For the other direction, if a container sample contains more than 4096
samples there is already code for splitting them up.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692750

4 years agortspsrc: fix requested range
Mathieu Duponchelle [Tue, 11 Feb 2020 20:52:41 +0000 (21:52 +0100)]
rtspsrc: fix requested range

When the server replies with a range "now-", it is presumed to
be a "live" stream and we should request a similar range.

This was the case prior to my refactoring to make use of
gst_rtsp_range_to_string in 5f1a732bc7b76a6f1b8aa5f26b6e76fbca0261c7,
this commit restores the behaviour for that case.

4 years agortpptdemux: set payload to caps inside gst_rtp_pt_demux_get_caps
Mikhail Fludkov [Thu, 13 Jul 2017 11:49:07 +0000 (13:49 +0200)]
rtpptdemux: set payload to caps inside gst_rtp_pt_demux_get_caps

Refactoring to remove duplicate code and add test

4 years agortpptdemux: Fix debug to use GST_DEBUG_OBJECT
Stian Selnes [Thu, 16 Mar 2017 19:57:54 +0000 (20:57 +0100)]
rtpptdemux: Fix debug to use GST_DEBUG_OBJECT

4 years agortpbin: use max-streams on rtpssrcdemux
Mikhail Fludkov [Wed, 14 Sep 2016 14:49:26 +0000 (16:49 +0200)]
rtpbin: use max-streams on rtpssrcdemux

The proper way of capping on max-streams is to do it in rtpssrcdemux.
This patch uses the newly introduced property on rtpssrcdemux. Previous
behavior would not prevent rtpssrcdemux spawning new pads for every new
ssrc and potentialy causing performance trouble during teardown.

4 years agortpssrcdemux: Handle RTCP APP packets
John Bassett [Wed, 18 Jan 2017 14:32:03 +0000 (14:32 +0000)]
rtpssrcdemux: Handle RTCP APP packets

Fix crash when processing RTCP APP packets.

4 years agortpssrcdemux: Bad RTP/RTCP packet is not fatal
John Bassett [Thu, 12 Jan 2017 16:05:59 +0000 (16:05 +0000)]
rtpssrcdemux: Bad RTP/RTCP packet is not fatal

When used for processing bundled media streams within rtpbin the rtpssrcdemux element may
receive bad RTP and RTCP packets, these should not be treated as a fatal error.

4 years agortpssrcdemux: introduce max-streams property
Mikhail Fludkov [Wed, 14 Sep 2016 14:41:02 +0000 (16:41 +0200)]
rtpssrcdemux: introduce max-streams property

The property is useful against atacks when the sender changes SSRC for
every RTP packet. The property with the same name introduced in rtpbin
was not enough, because we still can end up with thousands of pads
allocated in rtpssrcdemux.

4 years agortpssrcdemux: fix test warnings
Havard Graff [Mon, 10 Feb 2020 13:22:47 +0000 (14:22 +0100)]
rtpssrcdemux: fix test warnings

4 years agortspsrc: Fix for segmentation fault when handling set/get_parameter requests
Alexander Lapajne [Fri, 7 Feb 2020 09:03:49 +0000 (10:03 +0100)]
rtspsrc: Fix for segmentation fault when handling set/get_parameter requests

gstrtspsrc uses a queue, set_get_param_q, to store set param and get
param requests. The requests are put on the queue by calling
get_parameters() and set_parameter(). A thread which executs in
gst_rtspsrc_thread() then pops requests from the queue and processes
them. The crash occured because the queue became empty and a NULL
request object was then used. The reason that the queue became empty
is that it was popped even when the thread was NOT processing a get
parameter or set parameter command. The fix is to make sure that the
queue is ONLY popped when the command being processed is a set
parameter or get parameter command.

4 years agortpsession: Add test for packet rate maths
Olivier Crête [Fri, 27 Sep 2019 20:52:06 +0000 (16:52 -0400)]
rtpsession: Add test for packet rate maths

4 years agortpstats: Base the packet rate average on the packet rate itself
olivier.crete@collabora.com [Tue, 10 Sep 2019 18:03:02 +0000 (19:03 +0100)]
rtpstats: Base the packet rate average on the packet rate itself

Do this so that the average update speed is in time instead of varying
based on the actual packet arrival rate.

4 years agortpstats: Don't save the ts & seqnum if the avg is not updated
olivier.crete@collabora.com [Tue, 10 Sep 2019 17:59:02 +0000 (18:59 +0100)]
rtpstats: Don't save the ts & seqnum if the avg is not updated

This makes it update correctly when you have more than one packet per
frame.

4 years agov4l2: map GST_VIDEO_FORMAT_BGR15
Guillaume Desmottes [Wed, 5 Feb 2020 07:18:45 +0000 (12:48 +0530)]
v4l2: map GST_VIDEO_FORMAT_BGR15

The GstVideoFormat to v4l2 conversion was missing for BGR15.

4 years agov4l2: fix crash on invalid caps
Guillaume Desmottes [Wed, 5 Feb 2020 06:30:00 +0000 (12:00 +0530)]
v4l2: fix crash on invalid caps

gst_v4l2_object_set_format_full() was returning FALSE without setting
an error. Caller code (gst_v4l2src_fixate()) was then derefing a
NULL pointer when trying to handle the error.

4 years agosplitmuxsink: Include actual sink element in the fragment-opened/closed messages
Sebastian Dröge [Mon, 27 Jan 2020 14:00:30 +0000 (16:00 +0200)]
splitmuxsink: Include actual sink element in the fragment-opened/closed messages

If not configuring the sinks via the "location" property this can be
useful to know for which sink the fragment was actually opened/closed,
especially if finalization of the fragments is happening asynchronously.

4 years agortpjitterbuffer: fix scaling from RTP-time to NTP-time
Juergen Werner [Wed, 29 Jan 2020 11:05:07 +0000 (12:05 +0100)]
rtpjitterbuffer: fix scaling from RTP-time to NTP-time

The scaling was inverse.

4 years agortprtxsend: allow generic input caps
Mathieu Duponchelle [Mon, 27 Jan 2020 22:59:05 +0000 (23:59 +0100)]
rtprtxsend: allow generic input caps

When connected to an upstream rtpfunnel element, payload-type,
ssrc and clock-rate will not be present in the received caps.

rtprtxsend can already deal with only the clock rate being
present there, a new property is exposed to allow users to
provide a payload-type -> clock-rate map, this enables the
use of the max-size-time property for bundled streams.

4 years agovp8enc/vp8enc: set 1 for the default value of VP8E_SET_STATIC_THRESHOLD
Julien Isorce [Mon, 27 Jan 2020 23:17:27 +0000 (15:17 -0800)]
vp8enc/vp8enc: set 1 for the default value of VP8E_SET_STATIC_THRESHOLD

In Google webrtc, the setting VP8E_SET_STATIC_THRESHOLD is set to 1
(except when the content is known to be static very often in which
case it is set to 100, i.e. when sharing screen with Google Hangouts).

The cpu usage drops a lot when using 1 for above setting because it
allows the encoder to skip static/low content blocks. The current
0 default value uses too much cpu and confuses the user regarding
the cpu usage expectations. User expects vp8enc to use low cpu by
default.

Documentation of VP8E_SET_STATIC_THRESHOLD:
  https://github.com/webmproject/libvpx/blob/master/vpx/vp8cx.h#L188

chromium/webrtc:
  https://chromium.googlesource.com/external/webrtc/+/b484ec0082948ae086c2ba4142b4d2bf8bc4dd4b/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc#822

Closes #58

4 years agojpegdec: Check return value of gst_buffer_map()
Nicolas Dufresne [Mon, 27 Jan 2020 22:16:02 +0000 (17:16 -0500)]
jpegdec: Check return value of gst_buffer_map()

Without this check, the element will crash instead of returning an
error.

4 years agosplitmuxsink: Check the correct sink class for the existence of the "location" property
Sebastian Dröge [Mon, 27 Jan 2020 13:52:42 +0000 (15:52 +0200)]
splitmuxsink: Check the correct sink class for the existence of the "location" property

4 years agoqtdemux: Always prefer information from v1/v2 sound sample description over sample...
Sebastian Dröge [Mon, 13 Jan 2020 09:58:12 +0000 (11:58 +0200)]
qtdemux: Always prefer information from v1/v2 sound sample description over sample description entry

ffmpeg is doing the same and various files in the wild have bogus
information in the sample description if the same information is also
duplicated afterwards in the v1/v2 sound sample desription.

Previously we only did this for non-raw audio due to
  https://bugzilla.gnome.org/show_bug.cgi?id=374914
but this specific file is already worked around differently. It still
works after this change.

Also remove ad-hoc GST_READ_DOUBLE_BE re-implementation and move the
switch for legacy audio formats after reading all the sample
descriptions as we want to override the values from there.

4 years agoavimux: Add support for >2 raw audio channels
Sebastian Dröge [Mon, 13 Jan 2020 18:02:58 +0000 (20:02 +0200)]
avimux: Add support for >2 raw audio channels

For this case write a WAVEFORMATEXTENSIBLE header and also reorder the
raw audio channels to the AVI channel order if needed.

4 years agowavenc: Fix writing of the channel mask with >2 channels
Sebastian Dröge [Mon, 13 Jan 2020 18:07:01 +0000 (20:07 +0200)]
wavenc: Fix writing of the channel mask with >2 channels

The channel position is an enum but the conversion code assumed it's a
mask. Convert accordingly.

4 years agortph265pay: TID for NALU type 48 was always set to 7
Kristofer Björkström [Fri, 10 Jan 2020 15:30:33 +0000 (16:30 +0100)]
rtph265pay: TID for NALU type 48 was always set to 7

A typo bug: | instead of & resulted in TID alwasy being set to 7
for the aggregated NALU of type 48

4 years agoimagefreeze: Add support for replacing the output buffer
Sebastian Dröge [Fri, 10 Jan 2020 12:54:26 +0000 (14:54 +0200)]
imagefreeze: Add support for replacing the output buffer

By default imagefreeze will still reject new buffers after the first one
and immediately return GST_FLOW_EOS but the new allow-replace property
allows to change this.

Whenever updating the buffer we now also keep track of the configured
caps of the buffer and from the source pad task negotiate correctly
based on the potentially updated caps.

Only the very first time negotiation of a framerate with downstream is
performed, afterwards only the caps themselves apart from the framerate
are updated.

4 years agoqtdemux: Fix race on pad reconnection
Alicia Boya García [Thu, 9 Jan 2020 18:43:02 +0000 (18:43 +0000)]
qtdemux: Fix race on pad reconnection

Elements emitting frames through several srcpads should use a
flow combiner to aggregate the chain returns and therefore only return
GST_FLOW_NOT_LINKED to upstream when all the downstream pads have
received GST_FLOW_NOT_LINKED.

In addition to that, in order to handle pads being relinked downstream,
the flow combiner should be reset in response to RECONFIGURE events.
This ensures that a both srcpads process a chain operation before a
GST_FLOW_NOT_LINKED can be propagated upstream (which would usually stop
the pipeline).

Otherwise, in a configuration with two srcpads, only one linked at a
time, after the relink the element could chain data through the now
unlinked pad and the flow combiner would resolve as GST_FLOW_NOT_LINKED
(stopping the pipeline) just because the now linked pad has not been
chained yet to update the flow combiner.

This patch adds handling of RECONFIGURE events to qtdemux. Also, since
this event handling causes the flow combiner to be used from a thread
other than the qtdemux streaming thread, usages of the flow combiner
has been guarded by the object lock.

4 years agosplitmuxsink: Fix assertion failure on set_property()
Seungha Yang [Mon, 6 Jan 2020 16:20:24 +0000 (01:20 +0900)]
splitmuxsink: Fix assertion failure on set_property()

GValue might have null object.

(gst-inspect-1.0:10304): GStreamer-CRITICAL ...
    gst_object_ref_sink: assertion 'object != NULL' failed

4 years agovideocrop: allow properties to be animated by GstController
Daniel Molkentin [Fri, 3 Jan 2020 14:16:02 +0000 (15:16 +0100)]
videocrop: allow properties to be animated by GstController

4 years agortspsrc: improved handling of control concatenation with base
Aaron Boxer [Tue, 24 Dec 2019 13:24:51 +0000 (08:24 -0500)]
rtspsrc: improved handling of control concatenation with base

Also, `control_url` variable has been renamed to `control_path`,
as it is actually a path.

4 years agortspsrc: append aggregate control string to base URL before query string
Aaron Boxer [Fri, 6 Dec 2019 17:34:15 +0000 (12:34 -0500)]
rtspsrc: append aggregate control string to base URL before query string

Appending control string to end of query changes meaning of query string
Fixes #650

4 years agoaasink & cacasink: add filter aatv & cacatv
Eric Marks [Sat, 28 Dec 2019 23:01:19 +0000 (23:01 +0000)]
aasink & cacasink: add filter aatv & cacatv

Add transform filter capabilities to aasink and cacasink in the form of new elements aatv and cacatv.

4 years agoalpha: Cleanup using G_DECLARE_FINAL_TYPE
Niels De Graef [Thu, 6 Jun 2019 09:03:34 +0000 (11:03 +0200)]
alpha: Cleanup using G_DECLARE_FINAL_TYPE

We started depending on GLib 2.44, so we can clean up all the GObject
boilerplate macros.

5 years agogood: use of g_value_dup_string
Stéphane Cerveau [Wed, 18 Dec 2019 15:07:18 +0000 (16:07 +0100)]
good: use of g_value_dup_string

Use helper method to get string from GValue.

5 years agortpbin: fix shutdown crash in rtpbin
Havard Graff [Thu, 19 Dec 2019 22:48:09 +0000 (23:48 +0100)]
rtpbin: fix shutdown crash in rtpbin

The key is to make sure the jitterbuffer is set to NULL *before* the
ptdemux.

The race that existed would basically happen when ptdemux had reached
READY, and the jitterbuffer would then push a buffer, triggering a new
pad with a new payloadtype being added and ghosted to the rtpbin itself.

However, the srcpad of the ptdemux would now be inactive, and all the
sticky-event pushed on it would be swallowed, not allowing any to reach
the ghost-pad. Then the buffer in-flight would come to the ghostpad,
and we would assert that a buffer arrived before the necessary
events.

By simply re-ordering the state-changes, we ensure that there will be
no buffer racing into the ptdemux while its state is being changed,
and the problem disappears completely.

Notice also that there is not point in disconnecting the signals on the
ptdemux before this point, since we need the push-thread to settle
down before we can do this in a non-racy way.

5 years agortspsrc: avoid seek DISCONT when only rate changes in same direction
Aaron Boxer [Thu, 12 Sep 2019 20:22:10 +0000 (14:22 -0600)]
rtspsrc: avoid seek DISCONT when only rate changes in same direction

Not setting DISCONT avoids a noticable delay when seeking
with only rate changing, in the same direction as current
rate.

5 years agortspsrc: Remove deprecated GTimeVal
Olivier Crête [Tue, 10 Dec 2019 23:13:11 +0000 (18:13 -0500)]
rtspsrc: Remove deprecated GTimeVal

GTimeVal won't work past 2038

5 years agoosxaudio: Remove deprecated GTimeVal
Olivier Crête [Tue, 10 Dec 2019 22:13:45 +0000 (17:13 -0500)]
osxaudio: Remove deprecated GTimeVal

5 years agoavimux: Add support for S24LE and S32LE raw audio
Sebastian Dröge [Wed, 18 Dec 2019 10:19:27 +0000 (12:19 +0200)]
avimux: Add support for S24LE and S32LE raw audio

avidemux already handles this correctly.

5 years agoavimux: Allow muxing v210 video into AVI
Sebastian Dröge [Mon, 16 Dec 2019 19:07:08 +0000 (21:07 +0200)]
avimux: Allow muxing v210 video into AVI

avidemux already handles this.

5 years agoflvdemux: Don't replace video codec data when we receive a PAR
Vivia Nikolaidou [Mon, 16 Dec 2019 16:43:44 +0000 (18:43 +0200)]
flvdemux: Don't replace video codec data when we receive a PAR

Receiving a pixel-aspect-ratio should trigger a caps change, but not
replace the existing video codec tag

5 years agoqtmux: protect access to GstElement.sinkpads
Mathieu Duponchelle [Thu, 12 Dec 2019 19:20:35 +0000 (20:20 +0100)]
qtmux: protect access to GstElement.sinkpads

5 years agoqtmux: port to GstAggregator
Mathieu Duponchelle [Tue, 3 Dec 2019 14:30:06 +0000 (15:30 +0100)]
qtmux: port to GstAggregator

5 years agogstrtspsrc: Add missing lock on free set_get_param_q
Joakim Johansson [Mon, 16 Dec 2019 12:03:51 +0000 (13:03 +0100)]
gstrtspsrc: Add missing lock on free set_get_param_q

Otherwise is it possible to get a crash in gst_rtspsrc_set_parameter.

5 years agosplitmuxsink: Increment fragment_id even if no fragment location was provided
Sebastian Dröge [Thu, 12 Dec 2019 16:53:00 +0000 (18:53 +0200)]
splitmuxsink: Increment fragment_id even if no fragment location was provided

Applications might handle locations and generally configuration of the
sink by themselves instead of having splitmuxsink set the location on
the sink. Nonetheless it makes sense to increment the fragment_id that
is passed to the signal so that applications know which fragment is
requested.

5 years agoflvmux: Use the last DTS for the metadata timestamp
Jan Alexander Steffens (heftig) [Thu, 12 Dec 2019 09:59:35 +0000 (10:59 +0100)]
flvmux: Use the last DTS for the metadata timestamp

This avoids creating a timestamp regression during a stream.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/429

5 years agoqtdemux: send GAP events for lagging audio and video streams too
Mathieu Duponchelle [Wed, 11 Dec 2019 16:30:50 +0000 (17:30 +0100)]
qtdemux: send GAP events for lagging audio and video streams too

The logic is taken straight from matroskademux, see
77403d0afee635f2de6c2e53a23e1f50ad0d00fa

5 years agoflvmux: Use thread-safe gmtime_r if available
Seungha Yang [Tue, 10 Dec 2019 14:48:35 +0000 (23:48 +0900)]
flvmux: Use thread-safe gmtime_r if available

gmtime on *nix is not thread-safe.

5 years agosplitmuxsink: provides a start-index property
Stéphane Cerveau [Thu, 5 Dec 2019 14:58:40 +0000 (14:58 +0000)]
splitmuxsink: provides a start-index property

Allow to change the fragment-id start index.

5 years agoqmlglsink: fix build on EGL platform without X11 headers
Philipp Zabel [Tue, 3 Dec 2019 10:36:07 +0000 (11:36 +0100)]
qmlglsink: fix build on EGL platform without X11 headers

If Mesa is built without X11 headers, building against Mesa EGL headers
requires a dependency on egl.pc, to define MESA_EGL_NO_X11_HEADERS.
This fixes a build error when compiling ext/qt/gstqtglutility.cc:

  In file included from /usr/include/EGL/egl.h:39,
                   from /usr/include/gstreamer-1.0/gst/gl/egl/gstegl.h:44,
                   from ../gst-plugins-good-1.16.1/ext/qt/gstqtglutility.cc:43:
  /usr/include/EGL/eglplatform.h:124:10: fatal error: X11/Xlib.h: No such file or directory

5 years agortpjpegdepay: outputs framed jpeg
Tim-Philipp Müller [Wed, 4 Dec 2019 01:03:49 +0000 (01:03 +0000)]
rtpjpegdepay: outputs framed jpeg

Add parsed=true to output caps, as we always output
whole frames, timestamped and all. Means also that
the output can be decoded by avdec_mjpeg wihout
plugging an extra parser (which has no rank).

5 years agoflvmux: Correct metadata handling in file and stream mode
Jan Alexander Steffens (heftig) [Tue, 3 Dec 2019 12:47:22 +0000 (13:47 +0100)]
flvmux: Correct metadata handling in file and stream mode

In file mode, only push one onMetaData at the start of the stream.

In stream mode, always push complete onMetaData. They get replaced, not
merged.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/418

5 years agoflvmux: Don't calculate duration in streamable mode
Jan Alexander Steffens (heftig) [Tue, 3 Dec 2019 12:46:09 +0000 (13:46 +0100)]
flvmux: Don't calculate duration in streamable mode

There's no header to rewrite, so the duration is left unused.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/418

5 years agortpL16depay: don't crash if data is not modulo channels*width
Havard Graff [Wed, 30 Nov 2016 14:55:01 +0000 (15:55 +0100)]
rtpL16depay: don't crash if data is not modulo channels*width

5 years agopkgconfig: remove gst-plugins-good-1.0-uninstalled.pc
Tim-Philipp Müller [Mon, 2 Dec 2019 19:00:45 +0000 (19:00 +0000)]
pkgconfig: remove gst-plugins-good-1.0-uninstalled.pc

This was never installed and it was only used by the uninstalled
autotools dev environment to locate the -good plugins for use
in unit tests in gstreamer modules higher up the stack.

It is no longer needed now that we no longer have an autotools build.

5 years agomeson.build: use join_paths() on prefix
Håvard Graff [Tue, 10 Oct 2017 13:45:28 +0000 (15:45 +0200)]
meson.build: use join_paths() on prefix

So that "/" are correct on Windows.

5 years agortpopuspay: use baseclass allocator for buffers
Havard Graff [Fri, 30 Jun 2017 07:48:58 +0000 (09:48 +0200)]
rtpopuspay: use baseclass allocator for buffers

That way we get some of the meta -> rtp-extension goodies.

5 years agovp9dec: Fix broken 4:4:4 8bits decoding
Seungha Yang [Fri, 29 Nov 2019 11:46:26 +0000 (20:46 +0900)]
vp9dec: Fix broken 4:4:4 8bits decoding

VPX_IMG_FMT_I444 pixel format with sRGB colorspace means
GBR data.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/651
5 years agortpsession: add test for requesting FIR after having requested PLI
Havard Graff [Fri, 18 Oct 2019 15:45:43 +0000 (17:45 +0200)]
rtpsession: add test for requesting FIR after having requested PLI

5 years agortpjitterbuffer: make test more stable
Havard Graff [Tue, 26 Nov 2019 14:00:18 +0000 (15:00 +0100)]
rtpjitterbuffer: make test more stable

5 years agortpsession: add locking for clear-pt-map
Havard Graff [Fri, 29 Nov 2019 13:23:49 +0000 (14:23 +0100)]
rtpsession: add locking for clear-pt-map

...or it will segfault from time to time...

5 years agomatroskamux: Add property to set DateUTC
Linus Svensson [Thu, 31 May 2018 08:29:43 +0000 (10:29 +0200)]
matroskamux: Add property to set DateUTC

Add a property that makes it possible for an application to set the
DateUTC header field in matroska files. This is useful for live feeds,
where the DateUTC header can be set to a UTC timestamp, matching the
beginning of the file.

Needs gstreamer!323

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/481

5 years agomatroskamux: Use nanosecond precision for DateUTC
Linus Svensson [Thu, 31 May 2018 09:20:36 +0000 (11:20 +0200)]
matroskamux: Use nanosecond precision for DateUTC

DateUTC is specified with nanosecond precision in matroska, make use of
that.

5 years agov4l2bufferpool: Queue number of allocated buffers to capture
Nicolas Dufresne [Wed, 17 Oct 2018 02:28:13 +0000 (02:28 +0000)]
v4l2bufferpool: Queue number of allocated buffers to capture

Before we do streamon, we queue all capture buffers by calling
resurrect. When the driver supports CREATE_BUFS, this would lead
to buffers being allocated till the maximum of 32 is reached.

Instead, we now save the number of allocated buffers and queue this
amount.

5 years agomatroskamux: Pass the right size to gst_collect_pads_add_pad
Jan Alexander Steffens (heftig) [Tue, 19 Nov 2019 13:23:48 +0000 (14:23 +0100)]
matroskamux: Pass the right size to gst_collect_pads_add_pad

We were lucky that GstMatroskamuxPad is larger than GstMatroskaPad.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/393

5 years agov4l2object: Workaround bad TRY_FMT colorimetry implementation
Nicolas Dufresne [Mon, 18 Nov 2019 18:27:42 +0000 (13:27 -0500)]
v4l2object: Workaround bad TRY_FMT colorimetry implementation

libv4l2 reset the colorpace to 0 and does not do any request to the
driver. This yields an invalid colorspace which currently cause a
negotiation failure. This workaround by ignoring bad values during the
TRY_FMT step.

5 years agoaacparse: fix wrong offset of adts channel
aogun [Mon, 4 Nov 2019 09:18:30 +0000 (17:18 +0800)]
aacparse: fix wrong offset of adts channel

5 years agosplitmuxsink: Don't take lock during posting message
Seungha Yang [Mon, 7 Oct 2019 03:45:00 +0000 (12:45 +0900)]
splitmuxsink: Don't take lock during posting message

An application might try to access splitmuxsink from sync message handler
by g_object_{get,set} which takes lock also. In general, we don't
take lock around message handler.

5 years agojpegdec: Fix incorrect logic in EOI tag detection
Scott Kanowitz [Thu, 12 Sep 2019 19:21:24 +0000 (15:21 -0400)]
jpegdec: Fix incorrect logic in EOI tag detection

This change fixes the reversed logic in the EOI tag detection
code.

5 years agoDon't pass default GLib marshallers for signals
Niels De Graef [Mon, 26 Aug 2019 06:03:24 +0000 (08:03 +0200)]
Don't pass default GLib marshallers for signals

By passing `NULL` to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.

Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.

5 years agortpjitterbuffer: Check the exit condition after executing timers
Nicolas Dufresne [Thu, 14 Nov 2019 22:33:08 +0000 (17:33 -0500)]
rtpjitterbuffer: Check the exit condition after executing timers

The do_expected_timeout() function may release the JBUF_LOCK, so we need
to check if nothing wanted the timer thread to exit after this call.
The side effect was that we may endup going back into waiting for a timer
which will cause arbitrary delay on tear down (or deadlock when test
clock is used).

Fixes #653

5 years agortpjitterbuffer: Check exit condition immediately after JBUF_WAIT
Nicolas Dufresne [Thu, 14 Nov 2019 22:20:51 +0000 (17:20 -0500)]
rtpjitterbuffer: Check exit condition immediately after JBUF_WAIT

JBUF_WAIT_QUEUE drops the JBUF_LOCK, which means the stop condition
for the chain function may have changed (change_state to NULL). Check
this immediately after the wait so that we don't delay shutting down.

5 years agovideocrop: Also update the coordinate when in-place
Nicolas Dufresne [Tue, 12 Nov 2019 22:28:22 +0000 (17:28 -0500)]
videocrop: Also update the coordinate when in-place

This update is needed when the output caps is not changed (e.g. we are
moving a viewport around).

Fixes #669

5 years agovideocrop: Don't always re-run the allocation query
Nicolas Dufresne [Mon, 11 Nov 2019 18:19:08 +0000 (13:19 -0500)]
videocrop: Don't always re-run the allocation query

When in-place, running an allocation is not useful since videocrop
is not implicated in the allocation. So only force the allocation
query for the case it was in passthrough. This is needed since the
change in the crop region will likely pull us out of this mode. For the
case we where neither in passthrough or in-place, the allocation query
is already ran by the baseclass, so nothing special is needed.

This fixes performance issues when changing the crop region per frame.
This was reproduced using videocrop2-test.

5 years agovideocrop: Cleanup spurious assignment
Nicolas Dufresne [Mon, 11 Nov 2019 18:18:52 +0000 (13:18 -0500)]
videocrop: Cleanup spurious assignment

These are just writing the same thing a second time.

5 years agojpegdec: don't overwrite the last valid line
Michael Olbrich [Wed, 7 Nov 2018 08:00:02 +0000 (09:00 +0100)]
jpegdec: don't overwrite the last valid line

If the the height is not a multiple of the macro block size then the memory
of the last line is reused for all extra lines. This is no problem if the
last line is duplicated properly. However, if the extra lines are not
initialized properly during encoding, then the last visible line is
overwritten with undefined data.
Use a extra buffer to avoid this problem.

5 years agosplitmuxsink: add fakesink support
Stéphane Cerveau [Thu, 7 Nov 2019 11:28:58 +0000 (12:28 +0100)]
splitmuxsink: add fakesink support

fakesink does not support "location" property and was generating
a warning.

5 years agomultiudpsink: don't lose scope_id
Sergey Nazaryev [Wed, 12 Dec 2018 16:07:39 +0000 (19:07 +0300)]
multiudpsink: don't lose scope_id

5 years agovpx: Error out if enabled and no features found
Nirbheek Chauhan [Tue, 5 Nov 2019 16:11:55 +0000 (21:41 +0530)]
vpx: Error out if enabled and no features found

Seee: https://gitlab.freedesktop.org/gstreamer/cerbero/issues/200

5 years agov4l2object: update match_buffer_layout() debug messages
Guillaume Desmottes [Sat, 25 May 2019 19:19:21 +0000 (21:19 +0200)]
v4l2object: update match_buffer_layout() debug messages

It's no longer used only to try importing buffers.

5 years agov4l2object: try matching buffer layout from downstream
Guillaume Desmottes [Thu, 23 May 2019 08:49:39 +0000 (10:49 +0200)]
v4l2object: try matching buffer layout from downstream

Ask v4l2 to produce buffers matching the buffer layout requested
downstream.

5 years agov4l2object: factor out gst_v4l2_object_match_buffer_layout()
Guillaume Desmottes [Tue, 21 May 2019 08:31:46 +0000 (10:31 +0200)]
v4l2object: factor out gst_v4l2_object_match_buffer_layout()

No semantic change.

5 years agortpjitterbuffer: make sure not to drop packets based on skew
Havard Graff [Sun, 20 Oct 2019 10:17:25 +0000 (12:17 +0200)]
rtpjitterbuffer: make sure not to drop packets based on skew

One of the jitterbuffers functions is to try and make sense of weird
network behavior.

It is quite unhelpful for the jitterbuffer to start dropping packets
itself when what you are trying to achieve is better network resilience.

In the case of a skew, this could often mean the sender has restarted
in some fashion, and then dropping the very first buffer of this "new"
stream could often mean missing valuable information, like in the case
of video and I-frames.

This patch simply reverts back to the old behavior, prior to https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/commit/8d955fc32b552b2db933c67f3cfa31d987f36b81
and includes the simplest test I could write to demonstrate the behavior,
where a single packet arrives "perfectly", then a 50ms gap happens,
and then two more packets arrive in perfect order after that.

# Conflicts:
# tests/check/elements/rtpjitterbuffer.c

5 years agov4l2transform: use alignments from upstream when importing on sink
Guillaume Desmottes [Wed, 17 Apr 2019 07:10:22 +0000 (12:40 +0530)]
v4l2transform: use alignments from upstream when importing on sink

Try configuring the v4l2 output with the alignments from upstream when
importing its buffers. This allows us to support importing with
non-standard strides and/or heights if supported by the driver.

5 years agov4l2object: add support for vertical padding when importing buffers
Guillaume Desmottes [Wed, 17 Apr 2019 06:55:14 +0000 (12:25 +0530)]
v4l2object: add support for vertical padding when importing buffers

We were already supporting horizontal padding by setting bytesperline to
the buffer stride but not vertical one.

We are now updating the format height with the padded height and crop to
the actual video resolution if needed.

5 years agov4l2object: fix debug message if driver rejects stride
Guillaume Desmottes [Wed, 17 Apr 2019 06:16:10 +0000 (11:46 +0530)]
v4l2object: fix debug message if driver rejects stride

The 'want' and 'got' strides were inversed.

5 years agov4l2: improve logs when importing buffers
Guillaume Desmottes [Mon, 15 Apr 2019 06:13:41 +0000 (11:43 +0530)]
v4l2: improve logs when importing buffers

Log strides and offsets from upstream.

Also fix a typo.

5 years agov4l2videodec: ensure pool exists before orphaning it
James Cowgill [Tue, 29 Oct 2019 14:05:48 +0000 (14:05 +0000)]
v4l2videodec: ensure pool exists before orphaning it

In commit e2ff87732d0b ("v4l2videodec: support orphaning") support for
orphaning the capture buffer pool was added when the format is
renegotiated. However, the commit forgot to check that a pool existed
before doing this. This is needed because it's possible for the format
to be renegotiated before a capture pool is allocated, which would
result in trying to orphan a NULL pool and lead to a NULL pointer
dereference.

Fix this by checking a pool exists first. If the pool doesn't exist,
there are no buffers to be reclaimed, so skip the allocation query in
that case.

5 years agoqmlglsrc: read from the back buffer when use-default-fbo = TRUE
Matthew Waters [Fri, 25 Oct 2019 11:03:18 +0000 (22:03 +1100)]
qmlglsrc: read from the back buffer when use-default-fbo = TRUE

glReadBuffer(GL_COLOR_ATTACHMENT0) on the default framebuffer (0) is
invalid GL API usage and would result in a GL error being thrown.

5 years agoqmlglsrc: fix vertical flip matrix
Matthew Waters [Fri, 25 Oct 2019 10:47:01 +0000 (21:47 +1100)]
qmlglsrc: fix vertical flip matrix

Some time ago libgstgl defined the majorness of matrices it uses.
The majorness used by qmlglsrc was incompatible with the libgstgl.

5 years agoqtmux: Fix memory leak while pushing fragmented data
Patricia Muscalu [Tue, 30 Jul 2019 10:07:18 +0000 (12:07 +0200)]
qtmux: Fix memory leak while pushing fragmented data

The memory leak occurs in the case when the buffer has been
added to the fragment_buffers array of the current pad and
never been sent because of the push failure of the previous
buffers: moof or mdat header or fragmented buffer(s).

5 years agogood: Avoid usage of deprecated API
Edward Hervey [Fri, 11 Oct 2019 12:20:15 +0000 (14:20 +0200)]
good: Avoid usage of deprecated API

GTimeval and related functions are now deprecated in glib.
Replacement APIs have been present since 2.26

5 years agoosxaudio: misspelled dependency
Javier Celaya [Mon, 15 Jul 2019 05:46:56 +0000 (07:46 +0200)]
osxaudio: misspelled dependency

When building osxaudio, the required 'AudioToolbox' dependency is
misspelled as 'AudioToolBox', which crashes the build with error:
ld: framework not found AudioToolBox

5 years agoRemove autotools build system
Tim-Philipp Müller [Sat, 8 Jun 2019 23:43:00 +0000 (00:43 +0100)]
Remove autotools build system

5 years agov4l2videoenc: fix wrong type cast
Tim-Philipp Müller [Sun, 13 Oct 2019 11:46:58 +0000 (12:46 +0100)]
v4l2videoenc: fix wrong type cast

Follow-up to commit 1b752c0f !361

5 years agois a typo here? gstv4l2object.c
HuQian [Wed, 25 Sep 2019 12:36:32 +0000 (12:36 +0000)]
is a typo here? gstv4l2object.c