Commit graph

18552 commits

Author SHA1 Message Date
Tim-Philipp Müller
66296fcae3 tests: rtp-payloading: add minimal vp8/vp9 rtp payloading/depayloading test 2020-03-12 16:55:44 +00:00
Stian Selnes
81a87c26f9 rtpvp8pay, rtpvp9pay: fix caps leak in set_caps() 2020-03-12 16:49:58 +00:00
Edward Hervey
5a893f2a95 videomixer: Don't leak peer caps 2020-03-12 11:22:56 +01:00
Thibault Saunier
21bc0d527b imagesequencesrc: Cleanup and add some features
* Implement the GstURIHandlerInterface
* Rework the locking
* Implement backward seeking handling
* Generate documentation
2020-03-11 15:11:54 +00:00
Fabian Orccon
7511999083 Add an imagesequencesrc element to stream sequence of images
See: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/121
2020-03-11 15:11:54 +00:00
Gordon Hart
05e8dd42e7 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.
2020-03-10 20:33:04 +00:00
yychao
7f89085251 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
2020-03-10 15:28:01 +00:00
Matthew Waters
dacdc74043 imagefreeze: handle reconfigure events on the srcpad 2020-03-10 21:22:20 +11:00
Matthew Waters
07a8a1c484 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
2020-03-10 21:22:03 +11:00
Matthew Waters
28f49e1fd5 imagefreeze: don't fail sending sticky events downstream
They will be repropagated anyway.
2020-03-10 21:08:45 +11:00
Markus Ebner
5dcbb6b0d8 videocrop: Add support for Y41B and Y42B 2020-03-10 08:24:56 +00:00
Markus Ebner
b562235283 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
2020-03-10 08:24:56 +00:00
Markus Ebner
4a9e5bbf8b videocrop: Use G_VALUE_INIT to initialize GValues 2020-03-10 08:24:56 +00:00
Sebastian Dröge
fc3bb4c549 jpegdec: Configure JPEG chroma-siting for YUV formats 2020-03-09 21:34:08 +02:00
Ognyan Tonchev
a78a74bff0 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.
2020-03-06 10:44:16 +00:00
Milian Wolff
3e0d557744 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,
 ^~~~~~~~~~~~~~~~~~~
```
2020-03-05 03:21:48 +00:00
Havard Graff
4046970b01 rtptwcc: make RTPTWCCManager a GObject 2020-03-04 16:48:04 +01:00
Havard Graff
026223cde2 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
2020-03-04 12:55:52 +01:00
Jan Schmidt
f490c38416 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.
2020-03-04 05:42:21 +00:00
Jan Schmidt
54f68ff36b 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.
2020-03-04 04:14:40 +00:00
Guillaume Desmottes
d43ad6e029 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
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
b3d96e06c6 deinterlace: add wrapper to get field lines from history
No semantic change so far, will be used to implement alternate support.
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
f0eb1419f6 deinterlace: stop checking line index boundaries
The LINE2() macro already prevents out of bound indexes using CLAMP_HI()
and CLAMP_LOW().
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
cca8008779 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.
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
6dde6038cc 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.
2020-03-03 17:15:00 +00:00
Jennifer Berringer
3287f1cb3f 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
2020-03-03 08:03:32 +00:00
Yeongjin Jeong
830db205f6 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.
2020-03-02 01:37:27 +09:00
Sebastian Dröge
885d330ee6 qtdemux: Try to infer useful header values for raw audio if the sound sample descriptions contain zero values 2020-02-28 13:52:40 +00:00
Sebastian Dröge
9e9af6711d qtdemux: Also use the enda atom for determining endianess of in32, fl32 and fl64 formats
Previously it was only used for in24.
2020-02-28 13:52:40 +00:00
Sebastian Dröge
67be373221 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.
2020-02-28 13:52:40 +00:00
Sebastian Dröge
2c5f6e508c qtdemux: Don't print "unhandled type" warnings for various other raw audio fourccs 2020-02-28 13:52:40 +00:00
Sebastian Dröge
65b30ecce6 qtdemux: Add some more raw audio fourccs to the header instead of duplicating them 2020-02-28 13:52:40 +00:00
Nirbheek Chauhan
42e7864e90 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
2020-02-26 19:05:24 +05:30
Sebastian Dröge
35a1cedb97 qtmux: Add support for 8k resolutions in prefill mode with ProRes 2020-02-25 15:46:44 +02:00
Sebastian Dröge
3998b7cb4c rtpjitterbuffer: Include string.h for memcpy() / memset()
Usually something else is pulling it in somehow already, but not on
Windows.
2020-02-25 09:07:47 +00:00
Håvard Graff
fdf002d069 rtpsession: fix crash when no extension-header present for twcc 2020-02-24 13:06:27 +00:00
Johan Bjäreholt
ce802f033c 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
2020-02-21 12:49:28 +00:00
Jake Barnes
2ffb52499f 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.
2020-02-21 08:56:10 +00:00
Stefano Buora
2d3dccdba7 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.
2020-02-20 08:27:35 +00:00
Matthew Waters
1326fcdbcc 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.
2020-02-20 13:46:06 +11:00
Seungha Yang
f286f30640 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.
2020-02-19 23:18:51 +09:00
Yeongjin Jeong
9feb35638a tests: flvmux: Add test for rollover timestamp
The timestamps that exceed uint32 maximum value should be handled to rollover.
2020-02-18 18:39:34 +09:00
Yeongjin Jeong
e836640bd5 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.
2020-02-18 18:39:31 +09:00
Havard Graff
63ae338c24 rtpjitterbuffer: don't use the timer-object after JBUF_UNLOCK
It could have been freed (rtp_timer_free) in the meantime.
2020-02-17 15:04:45 +01:00
Havard Graff
1df706448c 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
2020-02-14 10:09:02 +00:00
Håvard Graff
9ba9837058 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)
2020-02-14 10:08:05 +00:00
Sebastian Dröge
9593a3679e 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.
2020-02-14 08:48:01 +00:00
Sebastian Dröge
be1c97d3c9 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
2020-02-14 08:48:01 +00:00
Mathieu Duponchelle
1471100f37 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 5f1a732bc7,
this commit restores the behaviour for that case.
2020-02-12 05:47:54 +00:00
Mikhail Fludkov
57b0522cd7 rtpptdemux: set payload to caps inside gst_rtp_pt_demux_get_caps
Refactoring to remove duplicate code and add test
2020-02-11 18:39:22 +00:00