Markus Ebner [Mon, 9 Mar 2020 22:23:50 +0000 (23:23 +0100)]
videocrop: Use G_VALUE_INIT to initialize GValues
Sebastian Dröge [Fri, 28 Feb 2020 17:35:34 +0000 (19:35 +0200)]
jpegdec: Configure JPEG chroma-siting for YUV formats
Ognyan Tonchev [Thu, 6 Feb 2020 08:23:24 +0000 (09:23 +0100)]
rtph26x: Use gst_memory_map() instead of gst_buffer_map() in avc mode
gst_buffer_map () results in memcopying when a GstBuffer contains
more than one GstMemory and when AVC (length-prefixed) alignment is used.
This has quite an impact on performance on systems with limited amount of
resources. With this patch the whole GstBuffer will not be mapped at once,
instead each individual GstMemory will be iterated and mapped separately.
Milian Wolff [Tue, 26 Nov 2019 14:08:20 +0000 (15:08 +0100)]
qmlgl: ensure Qt defines GLsync to fix compile on some platforms
By explictly including QtGui/qopengl.h we force the code path that
defines GLsync in the Qt-specific way. Without that, some platforms
failed to compile the qmlgl plugin, since neither Qt nor gstreamer
defined GLsync then, leading to e.g.:
```
make[4]: Entering directory '/.../gst-plugins-good-1.16.1/ext/qt'
CXX libgstqmlgl_la-qtitem.lo
In file included from gstqtgl.h:32,
from qtitem.h:27,
from qtitem.cc:28:
/.../usr/include/gstreamer-1.0/
gst/gl/gstglfuncs.h:93:17: error: expected identifier before ‘*’ token
ret (GSTGLAPI *name) args;
^
/.../usr/include/gstreamer-1.0/
gst/gl/glprototypes/sync.h:27:1: note: in expansion of macro
‘GST_GL_EXT_FUNCTION’
GST_GL_EXT_FUNCTION (GLsync, FenceSync,
^~~~~~~~~~~~~~~~~~~
```
Havard Graff [Mon, 2 Mar 2020 12:50:55 +0000 (13:50 +0100)]
rtptwcc: make RTPTWCCManager a GObject
Havard Graff [Wed, 4 Mar 2020 10:17:16 +0000 (11:17 +0100)]
rtpjitterbuffer: fix stalling when resetting timers
When calling gst_rtp_jitter_buffer_reset you pass in a seqnum.
This is considered the starting-point for a new stream.
However, the old behavior would unref this buffer, basically lying to
the thread that is pushing out buffers saying that it can expect
this buffer, when it would never arrive. The resulting effect being no
more buffer pushed out of the jitterbuffer, and it would buffer
incoming data indefinitely.
By instead inserting the buffer in the gap_packets queue, the _reset()
function will take responsibility for using that as the first buffer
of the new stream.
Fixes #703
Jan Schmidt [Thu, 20 Feb 2020 15:14:11 +0000 (02:14 +1100)]
splitmux: Avoid negative DTS
In order to concatenate fragments, splitmuxsrc offsets
the start of each fragment PTS to 0 to align it with the
previous file. This means that DTS can go negative for
the first fragment, with really bad results.
Add a fixed offset to outgoing timestamp ranges to
avoid that.
Jan Schmidt [Tue, 3 Mar 2020 16:43:51 +0000 (03:43 +1100)]
qtmux: Remove warning in the log for mono video
Vanilla mono video was generating a spurious warning into the debug log
that's just misleading. Handle mono caps explicitly to avoid the warning.
Guillaume Desmottes [Mon, 27 Jan 2020 06:59:18 +0000 (12:29 +0530)]
deinterlace: add alternate support
In this mode each field is carried using its own buffer.
Allow deinterlace to negotiate caps with the Interlaced feature and
adjust the algorithm fetching lines.
Fix #620
Guillaume Desmottes [Mon, 3 Feb 2020 07:38:39 +0000 (13:08 +0530)]
deinterlace: add wrapper to get field lines from history
No semantic change so far, will be used to implement alternate support.
Guillaume Desmottes [Tue, 4 Feb 2020 11:18:21 +0000 (16:48 +0530)]
deinterlace: stop checking line index boundaries
The LINE2() macro already prevents out of bound indexes using CLAMP_HI()
and CLAMP_LOW().
Guillaume Desmottes [Mon, 20 Jan 2020 07:00:12 +0000 (12:30 +0530)]
deinterlace: fix video info on output frames
Output frames used to have their interlace mode set to the same one as
the input. This breaks their field and comp heights when deinterlacing
an alternate stream.
Guillaume Desmottes [Tue, 14 Jan 2020 09:21:07 +0000 (14:51 +0530)]
deinterlace: use output caps to compute buffer size
In interlace-mode=alternate the input buffers have half the size of the
output ones as each field has its own buffer.
Jennifer Berringer [Sat, 29 Feb 2020 13:10:56 +0000 (08:10 -0500)]
flacparse: fix broken reordering of flac metadata
Each FLAC metadata block starts with a flag denoting whether it is the
last metadata block. The existing flacparse code moves any existing
VORBISCOMMENT block to immediately follow the STREAMINFO block without
changing any block's last-metadata-block flag. If no VORBISCOMMENT block
exists, it created one with the last-metadata-block flag set to true.
This results in gstflacdec sometimes giving bad headers to libflac when
trying to play perfectly valid FLAC files depending on the file's
metadata ordering. Depending on the contents of the other metadata
blocks, current versions of libflac may or may not return
FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER when given this broken
metadata. This is most noticeable with files that have a large cover art
image attached where VORBISCOMMENT is the very last metadata block with
no PADDING afterwards.
This patch changes that behavior so that:
1. For FLAC files that already have a VORBISCOMMENT block, the metadata
order is preserved.
2. For FLAC files that do not have a VORBISCOMMENT block, the generated
dummy VORBISCOMMENT is placed immediately after STREAMINFO and
inherits the last-metadata-block flag from STREAMINFO.
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/484
Yeongjin Jeong [Thu, 27 Feb 2020 05:50:51 +0000 (14:50 +0900)]
tests: flvmux: Instead of using the testclock, just send eos event for drain
When using the testclock for determining clock in test, it is sometimes observed
that the clock entry is not registered in time by the aggregator. So deadlock occurs
between the aggregator and the test thread.
Sebastian Dröge [Fri, 28 Feb 2020 12:23:51 +0000 (14:23 +0200)]
qtdemux: Try to infer useful header values for raw audio if the sound sample descriptions contain zero values
Sebastian Dröge [Fri, 28 Feb 2020 12:00:51 +0000 (14:00 +0200)]
qtdemux: Also use the enda atom for determining endianess of in32, fl32 and fl64 formats
Previously it was only used for in24.
Sebastian Dröge [Fri, 28 Feb 2020 11:59:42 +0000 (13:59 +0200)]
qtdemux: Fix up header information for various fixed-format raw audio formats
Sometimes the headers contain useless, wrong or zero values for e.g. the
sample size with these formats. There's only a single valid value for
them so let's set these instead.
Sebastian Dröge [Fri, 28 Feb 2020 11:59:06 +0000 (13:59 +0200)]
qtdemux: Don't print "unhandled type" warnings for various other raw audio fourccs
Sebastian Dröge [Fri, 28 Feb 2020 11:57:37 +0000 (13:57 +0200)]
qtdemux: Add some more raw audio fourccs to the header instead of duplicating them
Nirbheek Chauhan [Tue, 25 Feb 2020 15:44:54 +0000 (21:14 +0530)]
rtpjitterbuffer: Don't use glib format modifiers with sscanf
We do not have a way to know the format modifiers to use with string
functions provided by the system. G_GUINT64_FORMAT and other string
modifiers only work for glib string formatting functions. We cannot
use them for string functions provided by the stdlib. See:
https://developer.gnome.org/glib/stable/glib-Basic-Types.html#glib-Basic-Types.description
```
../gst/rtpmanager/gstrtpjitterbuffer.c: In function 'gst_jitter_buffer_sink_parse_caps':
../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: unknown conversion type character 'l' in format [-Werror=format=]
|| sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
^~~~~~~~~~
In file included from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/gtypes.h:32,
from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/galloca.h:32,
from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib.h:30,
from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/gst.h:27,
from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/rtp/gstrtpbuffer.h:27,
from ../gst/rtpmanager/gstrtpjitterbuffer.c:108:
/home/nirbheek/cerbero/build/dist/windows_x86/lib/glib-2.0/include/glibconfig.h:69:28: note: format string is defined here
#define G_GUINT64_FORMAT "llu"
^
../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: too many arguments for format [-Werror=format-extra-args]
|| sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
^~~~~~~~~~
```
See also: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/379
Sebastian Dröge [Mon, 24 Feb 2020 13:25:07 +0000 (15:25 +0200)]
qtmux: Add support for 8k resolutions in prefill mode with ProRes
Sebastian Dröge [Tue, 25 Feb 2020 09:06:43 +0000 (11:06 +0200)]
rtpjitterbuffer: Include string.h for memcpy() / memset()
Usually something else is pulling it in somehow already, but not on
Windows.
Håvard Graff [Mon, 24 Feb 2020 13:06:27 +0000 (13:06 +0000)]
rtpsession: fix crash when no extension-header present for twcc
Johan Bjäreholt [Fri, 21 Feb 2020 08:34:30 +0000 (09:34 +0100)]
matroska-mux: Fix incorrect rounding of timestamps
Previously we saved the buffer_timestamp straight into
mux->cluster_time. Since the cluster time saved into the file does not
have as high precision as GstClockTime depending on the timecodescale
the rounding of relative_timestamp was invalid as mux->cluster_time
which it was calculated relative to was not equal to the cluster time
written to the matroska file.
Example of "mkvinfo -v" of how it looks before and after this change in
an scenario where previously timestamps got out of order because of this
issue.
Notice the timestamp of the SimpleBlock right before and right after the
Cluster now being in order. The consequence of this however is that the
cluster timestamp is not necessarily the same as the timestamp of the
first buffer in the cluster however (in case it's rounded up).
Before
| + SimpleBlock (track number 1, 1 frame(s), timecode 126.922s = 00:02:06.922)
| + Frame with size 432
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.933s = 00:02:06.933)
| + Frame with size 329
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.955s = 00:02:06.955)
| + Frame with size 333
|+ Cluster
| + Cluster timecode: 126.954s
| + Cluster previous size: 97344
| + SimpleBlock (key, track number 1, 1 frame(s), timecode 126.954s = 00:02:06.954)
| + Frame with size 61239
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.975s = 00:02:06.975)
| + Frame with size 338
After
| + SimpleBlock (track number 1, 1 frame(s), timecode 135.456s = 00:02:15.456)
| + Frame with size 2260
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.468s = 00:02:15.468)
| + Frame with size 332
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.490s = 00:02:15.490)
| + Frame with size 335
|+ Cluster
| + Cluster timecode: 135.489s
| + Cluster previous size: 158758
| + SimpleBlock (key, track number 1, 1 frame(s), timecode 135.490s = 00:02:15.490)
| + Frame with size 88070
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.511s = 00:02:15.511)
| + Frame with size 336
Jake Barnes [Wed, 19 Feb 2020 04:59:19 +0000 (15:59 +1100)]
souphttpsrc: Fix cookies property
Disable session sharing and cookie jar when cookies property is set.
The cookie jar actually replaces or removes any existing Cookie header
set on the message, so the cookies property was effectively being
ignored. There doesn't appear to be a way to inject the cookies into the
jar without having to specify matching domains etc., so it's not
possible to simulate the old behaviour of unconditionally sending the
cookies with all messages, besides simply disabling the cookie jar.
Stefano Buora [Thu, 20 Feb 2020 08:06:10 +0000 (09:06 +0100)]
rtspsrc: remove useless function calls
Comparing gst_rtspsrc_loop_interleaved and gst_rtspsrc_loop_udp, and investigating on timeout issues, it sounds like a piece of code has been originally copied from udp to the interleaved one. The timeout variable is never used inside the interleaved one. No side effect has been seen in the removed function calls.
The debug message removed is pointless as the timeout used is "src->tcp_timeout" that is fixed.
The presence of the two timeout drove my team in investigating if the reference to the tcp_timeout was correct (it is). Hence we removed the misleading reference to the local timeout variable.
Matthew Waters [Thu, 20 Feb 2020 02:43:13 +0000 (13:43 +1100)]
rtpbin: fix typo setting max-dropout/misorder-time
we were setting the max-dropout-time to the value of the
max-misorder-time which by default has a factor of 30 difference in
value.
Seungha Yang [Wed, 19 Feb 2020 11:27:54 +0000 (20:27 +0900)]
qtdemux: Parse VP Codec Configuration Box
The VP Codec Configuration Box (vpcC) contains vp9 profile and
colorimetry information. Especially the profile information might
be useful for downstream to select capable decoder element.
Yeongjin Jeong [Tue, 18 Feb 2020 09:36:36 +0000 (18:36 +0900)]
tests: flvmux: Add test for rollover timestamp
The timestamps that exceed uint32 maximum value should be handled to rollover.
Yeongjin Jeong [Tue, 18 Feb 2020 05:58:00 +0000 (14:58 +0900)]
flvmux: Support rollover in timestamp
For live streams, if we keep the stream for a long time, the timestamp
will be larger than max_uint32. In that case, timestamp should be handled
as a rollover timestamp rather than a backward timestamp.
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.
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
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)
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.
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
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.
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
Stian Selnes [Thu, 16 Mar 2017 19:57:54 +0000 (20:57 +0100)]
rtpptdemux: Fix debug to use GST_DEBUG_OBJECT
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.
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.
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.
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.
Havard Graff [Mon, 10 Feb 2020 13:22:47 +0000 (14:22 +0100)]
rtpssrcdemux: fix test warnings
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.
Olivier Crête [Fri, 27 Sep 2019 20:52:06 +0000 (16:52 -0400)]
rtpsession: Add test for packet rate maths
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.
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.
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.
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.
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.
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.
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.
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
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.
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
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.
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.
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.
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
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.
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.
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
Daniel Molkentin [Fri, 3 Jan 2020 14:16:02 +0000 (15:16 +0100)]
videocrop: allow properties to be animated by GstController
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.
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
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.
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.
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.
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.
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.
Olivier Crête [Tue, 10 Dec 2019 23:13:11 +0000 (18:13 -0500)]
rtspsrc: Remove deprecated GTimeVal
GTimeVal won't work past 2038
Olivier Crête [Tue, 10 Dec 2019 22:13:45 +0000 (17:13 -0500)]
osxaudio: Remove deprecated GTimeVal
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.
Sebastian Dröge [Mon, 16 Dec 2019 19:07:08 +0000 (21:07 +0200)]
avimux: Allow muxing v210 video into AVI
avidemux already handles this.
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
Mathieu Duponchelle [Thu, 12 Dec 2019 19:20:35 +0000 (20:20 +0100)]
qtmux: protect access to GstElement.sinkpads
Mathieu Duponchelle [Tue, 3 Dec 2019 14:30:06 +0000 (15:30 +0100)]
qtmux: port to GstAggregator
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.
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.
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
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
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.
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.
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
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).
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
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
Havard Graff [Wed, 30 Nov 2016 14:55:01 +0000 (15:55 +0100)]
rtpL16depay: don't crash if data is not modulo channels*width
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.
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.
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.
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
Havard Graff [Fri, 18 Oct 2019 15:45:43 +0000 (17:45 +0200)]
rtpsession: add test for requesting FIR after having requested PLI
Havard Graff [Tue, 26 Nov 2019 14:00:18 +0000 (15:00 +0100)]
rtpjitterbuffer: make test more stable
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...
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
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.
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.
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