Xavier Claessens [Tue, 24 Mar 2020 01:32:04 +0000 (21:32 -0400)]
Fix usage of C99
It's 2020, way too early for that, let's stick to C89 for now.
Seungha Yang [Mon, 23 Mar 2020 07:34:46 +0000 (16:34 +0900)]
v4l2bufferpool: Use unique name for v4l2bufferpool object
Assign unique sequence number to an object name for better debugging
Matthew Waters [Mon, 23 Mar 2020 03:02:22 +0000 (14:02 +1100)]
qmlgloverlay: don't leak resources freed on a different GL thread
deleting a QOpenGLFrameBufferObject needs to occur on the same thread it
was created on in order to actually free the relevant resources
immediately. Otherwise, they will be queued for deletion and not freed
until the associated QOpenGLContext is destroyed.
Matthew Waters [Thu, 19 Mar 2020 22:14:01 +0000 (09:14 +1100)]
qt: reorganize include defines
Havard Graff [Thu, 19 Mar 2020 22:17:21 +0000 (23:17 +0100)]
rtptimerqueue: remove ->num from the timer
This concept was only used by the "multi"-lost timer, and since that
one is not around any longer, the "num" concept is superfluous.
Havard Graff [Thu, 19 Mar 2020 22:37:26 +0000 (23:37 +0100)]
rtpjitterbuffer: remove the concept of "already-lost"
This is a concept that only applies when a buffer arrives in the chain
function, and it has already been scheduled as part of a "multi"-lost
timer.
However, "multi"-lost timers are now a thing of the past, making this
whole concept superflous, and this buffer is now simply counted as "late",
having already been pushed out (albeit as a lost-event).
Havard Graff [Thu, 19 Mar 2020 22:12:04 +0000 (23:12 +0100)]
rtpjitterbuffer: immediately insert a lost-event on multiple lost packets
There is a problem with the code today, where a single timer will
be scheduled for a series of lost packets, and then if the first packet
in that series arrives, it will cause a rescheduling of that timer, going
from a "multi"-timer to a single-timer, causing a lot of the packets
in that timer to be unaccounted for, and creating a situation in where
the jitterbuffer will never again push out another packet.
This patch solves the problem by instead of scheduling those lost packets
as another timer, it instead asks to have that lost-event pushed straight
out.
This very much goes with the intent of the code here: These packets are
so desperately late that no cure exists, and we might as well get the
lost-event out of the way and get on with it.
This change has some interesting knock-on effect being presented in
later commits. It completely removes the concept of "already-lost", so
that is why that test has been disabled in this commit, to be
removed later.
Havard Graff [Thu, 19 Mar 2020 22:03:50 +0000 (23:03 +0100)]
rtpjitterbuffer: rework large-gap tests
Make sure to set the time the buffer is supposed to arrive at, so
as not to trigger an artificial situation.
Havard Graff [Thu, 19 Mar 2020 11:17:22 +0000 (12:17 +0100)]
rtpjitterbuffer: refactor lost_timeout code
Split it up in code related to the timer, (do_lost_timeout) and code
to insert a lost-item/event and update private jitterbuffer-variables.
Havard Graff [Fri, 18 Oct 2019 15:43:36 +0000 (17:43 +0200)]
test/check: split out rtptimerqueue-tests in a separate file
Dmitriy Purgin [Wed, 5 Feb 2020 08:56:23 +0000 (09:56 +0100)]
gstqmlgl: Link to opengl32.lib on MinGW
Seungha Yang [Thu, 19 Mar 2020 14:51:47 +0000 (23:51 +0900)]
qtmux: Fix build warning
gstqtmux.c(644): warning C4133: '=':
incompatible types - from 'gboolean (__cdecl *)(GstAggregator *,GstAggregatorPad *,GstEvent *)'
to 'GstFlowReturn (__cdecl *)(GstAggregator *,GstAggregatorPad *,GstEvent *)'
Jan Schmidt [Thu, 19 Mar 2020 12:05:49 +0000 (23:05 +1100)]
splitmuxsink: Reset cleanly for reuse
Reset the splitmuxsink completely when changing states so that
it can be reused.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1241
Zebediah Figura [Tue, 18 Feb 2020 04:37:10 +0000 (22:37 -0600)]
mpegaudioparse: Use a constant bit rate to convert between time and bytes if possible.
This should result in no worse accuracy than the base parse element, and may
result in better accuracy. In particular, the number of bytes processed at any
given point, as accumulated by baseparse, can be only accurate to
(1 / # of frames) bytes per second, and if we try to seek immediately after
pausing the pipeline to a large offset, this small inaccuracy can propagate to
something noticeable.
The use case that prompted this patch is a 45-minute MPEG-1 layer 3 file, which
has a constant bit rate but no seek tables. Trying to seek the pipeline
immediately after pauisng it, without the ACCURATE flag, to a location 41
minutes in, yields a location that is, even with <https://gitlab.freedesktop.org/gstreamer/gstreamer/merge_requests/374>,
still audibly incorrect. This patch yields a much closer position, no longer
audibly incorrect, and likely within a frame of the most correct position.
Mathieu Duponchelle [Wed, 4 Mar 2020 21:10:40 +0000 (22:10 +0100)]
qtmux: fix renegotiation check
By the time sink_event is called, the pad's current caps have
already been updated. To address this, implement sink_event_pre_queue,
and check if the pad can be renegotiated there.
Fixes #707
Seungha Yang [Thu, 12 Mar 2020 11:34:47 +0000 (20:34 +0900)]
splitmuxsink: Decouple keyframe request and the decision for fragmentation
Split the decision for keyframe request and fragmentation in order to
ensure periodic keyframe request.
Matthew Waters [Wed, 26 Feb 2020 07:29:06 +0000 (18:29 +1100)]
qt: add a qml overlay filter element [part 2]
It takes a qml scene description and renders it using a possible input
stream.
Currently supported on GLX and WGL.
Follow up to (as that MR had an old version of the commit):
- https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/475
-
4778d7166a02caf793df4f845dc35b6933d87c81: qt: add a qml overlay filter element
Matthew Waters [Wed, 26 Feb 2020 07:29:06 +0000 (18:29 +1100)]
qt: add a qml overlay filter element
It takes a qml scene description and renders it using a possible input
stream.
Currently supported on GLX and WGL.
Matthew Waters [Tue, 25 Feb 2020 10:47:14 +0000 (21:47 +1100)]
qt: don't always activate/deactivate our GstGLContext
Techincally it is enough to activate at the beginning and then forget.
Matthew Waters [Tue, 4 Feb 2020 08:43:52 +0000 (19:43 +1100)]
test/qml: add an dynamically adding qmlglsink element
The example shows how to add qmlglsink to an already running pipeline
with pre-existing OpenGL elements.
Matthew Waters [Tue, 4 Feb 2020 08:40:45 +0000 (19:40 +1100)]
qmlglsink: propagate the context up the the application
Allows the application to be notified of the OpenGL context creation.
Matthew Waters [Mon, 3 Feb 2020 04:59:34 +0000 (15:59 +1100)]
qtitem: fix leak of caps
Niels De Graef [Sun, 15 Mar 2020 18:28:18 +0000 (19:28 +0100)]
wavpack: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:26:18 +0000 (19:26 +0100)]
vpx: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:22:00 +0000 (19:22 +0100)]
twolame: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:20:49 +0000 (19:20 +0100)]
taglib: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:18:39 +0000 (19:18 +0100)]
speex: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:16:22 +0000 (19:16 +0100)]
soup: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:14:17 +0000 (19:14 +0100)]
shout2: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:11:52 +0000 (19:11 +0100)]
raw1394: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:06:50 +0000 (19:06 +0100)]
qt: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 18:00:18 +0000 (19:00 +0100)]
pulse: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 17:54:33 +0000 (18:54 +0100)]
mpg123: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 17:52:57 +0000 (18:52 +0100)]
libpng: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 17:49:53 +0000 (18:49 +0100)]
libcaca: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sun, 15 Mar 2020 17:40:28 +0000 (18:40 +0100)]
lame: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sat, 14 Mar 2020 16:52:38 +0000 (17:52 +0100)]
jack: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Sat, 14 Mar 2020 16:43:50 +0000 (17:43 +0100)]
gtk: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Fri, 13 Mar 2020 17:47:49 +0000 (18:47 +0100)]
gdk_pixbuf: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Fri, 13 Mar 2020 17:42:38 +0000 (18:42 +0100)]
flax: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Fri, 13 Mar 2020 17:39:38 +0000 (18:39 +0100)]
dv: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Thu, 12 Mar 2020 18:24:57 +0000 (19:24 +0100)]
cairo: Use G_DECLARE_FINAL_TYPE
Niels De Graef [Thu, 12 Mar 2020 18:20:42 +0000 (19:20 +0100)]
aalib: Use G_DECLARE_FINAL_TYPE
Tim-Philipp Müller [Thu, 12 Mar 2020 16:55:44 +0000 (16:55 +0000)]
tests: rtp-payloading: add minimal vp8/vp9 rtp payloading/depayloading test
Stian Selnes [Fri, 19 Oct 2018 14:17:17 +0000 (16:17 +0200)]
rtpvp8pay, rtpvp9pay: fix caps leak in set_caps()
Edward Hervey [Thu, 12 Mar 2020 10:22:56 +0000 (11:22 +0100)]
videomixer: Don't leak peer caps
Thibault Saunier [Tue, 11 Feb 2020 19:19:15 +0000 (16:19 -0300)]
imagesequencesrc: Cleanup and add some features
* Implement the GstURIHandlerInterface
* Rework the locking
* Implement backward seeking handling
* Generate documentation
Fabian Orccon [Sun, 10 Apr 2016 02:25:32 +0000 (02:25 +0000)]
Add an imagesequencesrc element to stream sequence of images
See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/121
Gordon Hart [Thu, 5 Mar 2020 16:55:44 +0000 (08:55 -0800)]
v4l2src: decrease gst_v4l2src_create log verbosity
Lower the verbosity of the 'sync' log message emitted
each buffer from gst_v4l2src_create down to LOG(6)
from INFO(4). This brings the logging behavior of
v4l2src closer to the GStreamer guidelines, which
recommend the INFO level be reserved for rare or
one-off messages.
yychao [Tue, 10 Mar 2020 09:19:46 +0000 (17:19 +0800)]
qtdemux: Add support for AC4
The caps received from qtdemux for AC-4 content are audio/x-gst-fourcc-ac_4
Based on patch by: Savinderjit Kaur
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/413
Matthew Waters [Tue, 10 Mar 2020 10:07:12 +0000 (21:07 +1100)]
imagefreeze: handle reconfigure events on the srcpad
Matthew Waters [Thu, 5 Mar 2020 11:47:16 +0000 (22:47 +1100)]
imagefreeze: properly ignore setting caps failures
Ignore the return value of gst_pad_set_caps() so that setcaps will set a
framerate that is usable.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/705
Matthew Waters [Thu, 5 Mar 2020 11:45:32 +0000 (22:45 +1100)]
imagefreeze: don't fail sending sticky events downstream
They will be repropagated anyway.
Markus Ebner [Mon, 9 Mar 2020 22:31:09 +0000 (23:31 +0100)]
videocrop: Add support for Y41B and Y42B
Markus Ebner [Mon, 9 Mar 2020 22:25:03 +0000 (23:25 +0100)]
videocrop: Add support for Y444
- Refactored the planar transform method to support all video formats
that are stored planar, independent of the used subsampling
- Added support for Y444
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.