Jan Schmidt [Fri, 16 Oct 2020 13:05:15 +0000 (00:05 +1100)]
rtph264depay: Warn when max SPS/PPS are collected in AVC mode.
The AVC codec_data has a flaw that it can only accomodate
31 SPS headers, even though H.264 can have 32, and 255 PPS,
when there can be 256 in H.264. When streaming RTP some
clients like to cycle through SPS/PPS ids when changing
configuration and can eventually accumulate a full set.
In that case, we have no choice but to discard one (oldest)
entry, or else the count written into the codec_data is wrong
and downstream decoding failures ensue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/775>
Havard Graff [Tue, 27 Oct 2020 23:29:05 +0000 (00:29 +0100)]
rtpjitterbuffer: don't send multiple instant RTX for the same packet
Due to us not properly acknowleding the time when the last RTX was sent
when scheduling a new one, it can easily happen that due to the packet
you are requesting have a PTS that is slightly old (but not too old when
adding the latency of the jitterbuffer), both its calculated second and
third (etc.) timeout could already have passed. This would lead to a burst
of RTX requests, which acts completely against its purpose, potentially
spending a lot more bandwidth than needed.
This has been properly reproduced in the test:
test_rtx_not_bursting_requests
The good news is that slightly re-thinking the logic concerning
re-requesting RTX, made it a lot simpler to understand, and allows us
to remove two members of the RtpTimer which no longer serves any purpose
due to the refactoring. If desirable the whole "delay" concept can actually
be removed completely from the timers, and simply just added to the timeout
by the caller of the API. But that can be a change for a another time.
The only external change (other than the improved behavior around bursting
RTX) is that the "delay" field now stricly represents the delay between
the PTS of the RTX-requested packet and the time it is requested on,
whereas before this calculation was more about the theoretical calculated
delay. This is visible in three other RTX-tests where the delay had
to be adjusted slightly. I am confident however that this change is
correct.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/789>
Jan Schmidt [Tue, 27 Oct 2020 12:43:49 +0000 (23:43 +1100)]
matroska-mux: Fix sparse stream crash
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/656
introduced an invalid memory access when debug is enabled, by casting
the wrong pointer to a GstCollectPad. Fixing that showed the original
change was incorrect and leads to an infinite loop in the
testsuite. This patch fixes both problems.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/788>
Thibault Saunier [Thu, 22 Oct 2020 18:29:01 +0000 (15:29 -0300)]
vpx: Fix the check to unfixed/unknown framerate to set bitrate
0/1 means unknown framerate not X/0 (which is illegal).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/782>
Arun Raghavan [Thu, 22 Oct 2020 13:17:26 +0000 (09:17 -0400)]
rtputils: Count metas with an empty tag list for copying/keeping
The GstMetaInfos registered in core do not set their tags to NULL, but
instead use an empty list (non-NULL list with a single NULL value).
Let's check explicitly for that so as to not miss some metas.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/779>
Bastien Reboulet [Fri, 16 Oct 2020 23:05:45 +0000 (16:05 -0700)]
qmlglsink: fix crash when created/destroyed in quick succession
The crash is caused by a race condition where the render thread
calls a method on the QtGLVideoItem instance that was
previously destroyed by the main thread.
Also, less frequently, QtGLVideoItem::onSceneGraphInitialized
is called when QQuickItem::window is null, also causing a crash.
Fixes #798
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/777>
Sebastian Dröge [Mon, 19 Oct 2020 15:23:25 +0000 (18:23 +0300)]
v4l2codec: Garbage collect old frames if they accumulate because of codec bugs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>
Sebastian Dröge [Mon, 19 Oct 2020 14:56:04 +0000 (17:56 +0300)]
v4l2codec: Pass system frame number as timestamp and use it to retrieve back frames reliably
System frame numbers are supposed to be unique and correct drivers are
passing through timestamps without modification from the output/sink to the
capture/src side.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/778>
Nicolas Dufresne [Thu, 24 Sep 2020 17:13:00 +0000 (13:13 -0400)]
rtpbin: Add clear-ssrc action
This action signal will delegate to clear-ssrc onto the rtpssrcdemux element
associated with the session. This allow rtpbin users to clear pads and
elements for a specific ssrc that is known to no longer be in use. This
happens when a pad is reused in rtpsrc or ristsrc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/736>
John-Mark Bell [Fri, 8 Sep 2017 19:02:13 +0000 (20:02 +0100)]
rtpvp8pay: payload temporally scaled bitstreams.
Co-Authored-By: Vincent Sanders <vince@pexip.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
Stian Selnes [Fri, 17 Nov 2017 14:11:41 +0000 (15:11 +0100)]
rtpvp8pay: Add picture-id-offset property
Add property to set the initial value for picture-id. RFC7741 says
that picture-id MAY be initialized to a random value, thus it's also
valid to simply set it to a fixed initial value. A fixed value is very
useful for testing.
Default behavior is not changed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
Mikhail Fludkov [Thu, 16 Mar 2017 14:23:28 +0000 (15:23 +0100)]
rtpvp8pay: move duplicate code to separate functions
Two new functions to modify picture id:
gst_rtp_vp8_pay_picture_id_reset - picks random picture id of
appropriate bitsize
gst_rtp_vp8_pay_picture_id_increment - increments picture id taking
care of wrapping
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
John-Mark Bell [Fri, 8 Sep 2017 07:13:05 +0000 (08:13 +0100)]
vp8enc: expect bps for temporal-scalability-target-bitrate.
Consistency with target-bitrate is less surprising and with
modern libvpx additional configuration is required to make
temporal scaling work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
John-Mark Bell [Fri, 8 Sep 2017 07:19:20 +0000 (08:19 +0100)]
vp8enc: finish support for temporally scaled encoding
- introduce two new properties:
* temporal-scalability-layer-flags:
Provide fine-grained control of layer encoding to the
outside world. The flags sequence should be a multiple of
the periodicity and is indexed by a running count of encoded
frames modulo the sequence length.
* temporal-scalability-layer-sync-flags:
Specify the pattern of inter-layer synchronisation (i.e.
which of the frames generated by the layer encoding
specification represent an inter-layer synchronisation).
There must be one entry per entry in
temporal-scalability-layer-flags.
- apply temporal scalability settings and expose as buffer
metadata.
This allows the codec to allocate a given frame to the correct
internal bitrate allocator. Additionally, all the
non-bitstream metadata needed to payload a temporally scaled
stream is now attached to each output buffer as a
GstVideoVP8Meta.
- add unit test for temporally scaled encoding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/728>
Stéphane Cerveau [Thu, 15 Oct 2020 16:21:54 +0000 (18:21 +0200)]
meson: update glib minimum version to 2.56
In order to support the symbol g_enum_to_string in various
project using GStreamer ( gst-validate etc.), the glib minimum
version should be 2.56.0.
Remove compat code as glib requirement
is now > 2.56
Version used by Ubuntu 18.04 LTS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/774>
Mathieu Duponchelle [Wed, 14 Oct 2020 12:30:34 +0000 (14:30 +0200)]
rtpst2022-1-fecenc: fix input seqnum check
We need to cast the incremented last seqnum to guint16 for
consistent checks on wraparound
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/770>
Jan Alexander Steffens (heftig) [Sat, 12 Sep 2020 07:02:30 +0000 (09:02 +0200)]
flvmux: Correct time types
- last_dts is in milliseconds, not nanoseconds as expected for
GstClockTime. Make it a generic guint64.
- Use GstClockTime for the fields that actually contain nanoseconds.
None of them should become negative.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/766>
Sebastian Dröge [Fri, 9 Oct 2020 06:31:27 +0000 (09:31 +0300)]
rtpst2022-1-fecenc: Don't unconditionally use GLib 2.60 APIs
g_queue_clear_full() in this case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/768>
Mathieu Duponchelle [Thu, 8 Oct 2020 16:54:55 +0000 (18:54 +0200)]
rtpulpfec: fix potential alignment issue in xor function
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753#note_646453
for context
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
Mathieu Duponchelle [Tue, 6 Oct 2020 01:03:13 +0000 (03:03 +0200)]
rtpmanager: implement SMPTE 2022-1 FEC encoder
+ improve integration of FEC encoders in rtpbin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
Mathieu Duponchelle [Tue, 6 Oct 2020 01:13:30 +0000 (03:13 +0200)]
rtpmanager: implement SMPTE 2022-1 FEC decoder
+ improve integration of FEC decoders in rtpbin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/753>
Olivier Crête [Wed, 8 Jul 2020 21:28:31 +0000 (17:28 -0400)]
rtpfunnel: Also forward custom sticky event
This is useful to track metadata about each group of packets
Also include a unit test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/666>
Thibault Saunier [Tue, 29 Sep 2020 12:44:54 +0000 (09:44 -0300)]
isomp4: Rename GstQTMux to GstBaseQTMux to avoid breaking API
Since
52b63de19ada283c1180c8fc00cacb1465fdf10f the qtmux GType was
renamed GstQTMuxElement which breaks presets, revert that change.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/755>
Sebastian Dröge [Mon, 28 Sep 2020 15:25:21 +0000 (18:25 +0300)]
rtp: Fix allocations to support source-info property
Use gst_rtp_base_payload_allocate_output_buffer() instead of
gst_rtp_buffer_new_allocate() in order to allocate RTP buffer with
correct number of CSRCs according to the meta.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
Stian Selnes [Fri, 23 Oct 2015 09:08:56 +0000 (11:08 +0200)]
rtpvp8pay: Fix allocation to support source-info property
Use gst_rtp_base_payload_allocate_output_buffer() in order to allocate
RTP buffer with correct number of CSRCs according to the meta.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/314
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/612>
Matthew Waters [Mon, 28 Sep 2020 05:36:00 +0000 (15:36 +1000)]
qtmux: output the correct limits in error messages
Having the current bytes being less than the limit was confusing!
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
Matthew Waters [Fri, 31 Jul 2020 06:47:37 +0000 (16:47 +1000)]
qtmux: properly support initial caps nego failure
Scenario:
- gap event causes h264parse to push made up caps that may fail checks
inside qtmux (e.g missing codec_data).
- the caps event has already been marked as received and is sticky on
the sink pad
- gst_qt_mux_pad_can_renegotiate() will retrieve the failed caps event
using gst_pad_get_current_caps() and reject the correct updated caps
with codec_data.
- Failure!
Keep track of the configured caps ourselves instead of relying on the
sticky event on the pad.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
Matthew Waters [Wed, 22 Jul 2020 05:34:44 +0000 (15:34 +1000)]
qtmux: support non-seekable downstream mode
Write an mdat per buffer in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/732>
Nicolas Dufresne [Wed, 23 Sep 2020 19:25:36 +0000 (15:25 -0400)]
rtpbin: Remove the rtpjitterbuffer with the stream
Since !348, the jitterbuffer was only removed with the session. This restores
the original behaviour and removes the jitterbuffer when the stream is
removed. This avoid accumulating jitterbuffer objects into the bin when a
session is reused.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
Nicolas Dufresne [Wed, 23 Sep 2020 17:26:51 +0000 (13:26 -0400)]
rtpbin: Cleanup dead code
The rtpjitterbuffer is now part of the session elements, we no longer need
to do the ref_sink dance when signalling it. It is already owned by the bin
when signalled. Also, the code that handles generic session elements already
handle the ref_sink() calls since:
03dc22951bacb6fdc3868c8f801e6a52c33a745f
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/735>
Matthew Waters [Fri, 18 Sep 2020 06:09:20 +0000 (16:09 +1000)]
rtph26*depay: drop FU's without a corresponding start bit
If we have not received a FU with a start bit set, any subsequent FU
data is not useful at all and would result in an invalid stream.
This case is constructed from multiple requirements in
RFC 3984 Section 5.8 and RFC 7798 Section 4.4.3. Following are excerpts
from RFC 3984 but RFC 7798 contains similar language.
The FU in a single FU case is forbidden:
A fragmented NAL unit MUST NOT be transmitted in one FU; i.e., the
Start bit and End bit MUST NOT both be set to one in the same FU
header.
and dropping is possible:
If a fragmentation unit is lost, the receiver SHOULD discard all
following fragmentation units in transmission order corresponding to
the same fragmented NAL unit.
The jump in seqnum case is supported by this from the specification
instead of implementing the forbidden_zero_bit mangling:
If a fragmentation unit is lost, the receiver SHOULD discard all
following fragmentation units in transmission order corresponding to
the same fragmented NAL unit.
A receiver in an endpoint or in a MANE MAY aggregate the first n-1
fragments of a NAL unit to an (incomplete) NAL unit, even if fragment
n of that NAL unit is not received. In this case, the
forbidden_zero_bit of the NAL unit MUST be set to one to indicate a
syntax violation.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/730>
Seungha Yang [Sun, 20 Sep 2020 12:06:19 +0000 (21:06 +0900)]
imagefreeze: Response caps query from srcpad
... and chain up to default query handler for unhandled query types.
Unhandled query shouldn't be returned with FALSE if there's no special needs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/731>
Matthew Waters [Wed, 16 Sep 2020 02:15:09 +0000 (12:15 +1000)]
qtmux: make documentation happy
introduce a base qtmux class that we can install documentation snippets
on instead of duplicating across alll the isomp4 elements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 28 May 2020 09:40:24 +0000 (19:40 +1000)]
isomp4/mux: add a fragment mode for initial moov with data
Used by some proprietary software for their fragmented files.
Adds some support for multi-stream fragmented files
Flow is as follows.
1. The first 'fragment' is written as a self-contained fragmented
mdat+moov complete with an edit list and durations, tags, etc.
2. Subsequent fragments are written with a mdat+moof and each stream is
interleaved as data arrives (currently ignoring the interleave-*
properties). data-offsets in both the traf and the trun ensure
data is read from the correct place on demuxing. Data/chunk offsets
are also kept for writing out the final moov.
3. On finalisation, the initial moov is invalidated to a hoov and the
size of the first mdat is extended to cover the entire file contents.
Then a moov is written as regularly would in moov-at-end mode (the
default).
This results in a file that is playable throughout while leaving a
finalised file on completion for players that do not understand
fragmented mp4.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 25 Jun 2020 06:37:56 +0000 (16:37 +1000)]
qtdemux: increase some logging on streams and sample parsing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 25 Jun 2020 06:35:45 +0000 (16:35 +1000)]
qtdemux: bail out when encountering an atom with a size of 0
A size 0 atom means the atom extends to the end of the file. No further
valid atoms will ever follow. Avoids a subsequent scan for an atom from
one byte earlier after encountering a size 0 atom.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 25 Jun 2020 06:33:04 +0000 (16:33 +1000)]
qtdemux: fix subsequent moof parsing after moov with valid samples
reset the moof_offset back to its original value like is done in the
error case just before.
Fixes subsequent parsing of a moof following a moov that contains valid
samples in a non-streaming fragmented mp4.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Matthew Waters [Thu, 25 Jun 2020 06:30:28 +0000 (16:30 +1000)]
qtdemux: extend edit list when fragmented
When we are fragmented, the edit list may only refer to the portion of
the media that is in the moov. Extend the edit list stop time when we
if there is only one qt segment and we are reading a fragmented file.
Fixes playback of some fragmented mp4 files generated by proprietary
programs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/643>
Nicolas Dufresne [Tue, 15 Sep 2020 18:22:13 +0000 (14:22 -0400)]
meson: Allow overriding qt5 feature
This will allow controlling that feature from gst-build
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/729>
Olivier Crête [Wed, 18 Nov 2015 00:14:01 +0000 (19:14 -0500)]
splitmuxsrc: Implement segment query
Fixes #239
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/713>
Sebastian Dröge [Mon, 14 Sep 2020 07:15:35 +0000 (10:15 +0300)]
rtpmp4gdepay: Allow lower-case "aac-hbr" instead of correct "AAC-hbr"
Various live555 based products are using the wrong "mode" string or
seem to assume case-insensitive matching, which is wrong.
Examples for this are the Yuan SC6C0N1 mini and the Kiloview E2.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/727>
Stefan Brüns [Sat, 2 May 2020 00:21:00 +0000 (02:21 +0200)]
qtdemux: Add support for AAX encrypted audio streams
This is modelled after the DASH Common Encryption scheme, but is somewhat
simpler as more parts are fixed, i.e. just one encryption scheme.
The output caps are fixed to 'application/x-aavd'. All information
required for decryption are part of the 'adrm' atom, which is passed
on as a property. The property is attached to the buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
Stefan Brüns [Sat, 2 May 2020 00:20:44 +0000 (02:20 +0200)]
qtdemux: Add 'aavd' and related fourcc codes for AAX encrypted audio
The 'aavd' box is contained in the 'stsd' sample description. The 'aavd'
box follows the layout of an 'mp4a' entry, i.e. it contains a single
standard 'esds' extension box, and the two proprietary 'adrm' and 'aabd'
extension boxes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/577>
Haakon Sporsheim [Mon, 23 Jun 2014 06:46:37 +0000 (08:46 +0200)]
vpxdec: request a sync point on decoder errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/725>
Camilo Celis Guzman [Sun, 13 Sep 2020 16:31:57 +0000 (18:31 +0200)]
rtp/vrawpay: use alloc_output_buffer from base class
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/726>
Ricky Tang [Mon, 7 Sep 2020 15:20:58 +0000 (23:20 +0800)]
rtspsrc: Fix push-backchannel-buffer parameter mismatch
When using python, signal parameter must match with function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/724>
Jérôme Laheurte [Thu, 10 Sep 2020 09:24:32 +0000 (11:24 +0200)]
jpegdec: check buffer size before dereferencing. Fixes #541
Some cameras (Panacast) have buggy drivers/firmware which send
invalid JPEG frames, containing no data, which makes jpegdec
crash because it assumes the frame is at least 2 bytes long.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/723>
Jan Alexander Steffens (heftig) [Thu, 10 Sep 2020 09:11:00 +0000 (11:11 +0200)]
flvmux: Improve logging of gst_flv_mux_buffer_to_tag_internal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
Jan Alexander Steffens (heftig) [Wed, 9 Sep 2020 13:12:53 +0000 (15:12 +0200)]
flvmux: Move stream skipping to GstAggregatorPadClass.skip_buffer
Besides looking like the correct place to put this, it allows us to drop
the entire aggregator queue. The old implementation only dropped at most
one buffer for each call of aggregate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/722>
Havard Graff [Tue, 8 Sep 2020 15:35:50 +0000 (17:35 +0200)]
v4l2object: plug memory-leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/718>
Mathieu Duponchelle [Fri, 28 Aug 2020 16:09:15 +0000 (18:09 +0200)]
vp9enc: expose row-mt property
With recent libvpx versions, multithreading can be enabled on
a per-tile basis, instead of on a per tile-column basis.
In combination with the new tile-rows property, this allows the
encoder to make much better use of the available CPU power.
Bump minimum libvpx version to 1.7.0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
Mathieu Duponchelle [Fri, 28 Aug 2020 15:45:48 +0000 (17:45 +0200)]
vpxenc: change default for deadline to good quality
Having the deadline set to best quality causes the encoder
to be absurdly slow, most real-life users will want the good
quality tradeoff instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
Mathieu Duponchelle [Fri, 28 Aug 2020 15:39:47 +0000 (17:39 +0200)]
vp9enc: expose tile-columns and tile-rows properties
Based on patch by Stian Selnes <stian@pexip.com>.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
Mathieu Duponchelle [Fri, 28 Aug 2020 15:35:26 +0000 (17:35 +0200)]
vpxenc: add configure_encoder virtual method
For subclasses to expose format-specific properties
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/707>
Mathieu Duponchelle [Tue, 8 Sep 2020 18:57:33 +0000 (20:57 +0200)]
splitmuxsink: fix sink pad release while PLAYING
- Release the split mux lock while removing the probes
- Flush the sinkpad to unblock other pads
- Turn check_completed_gop into a do while statement, when
waking up we want to recheck whether the current GOP is
ready for sending
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/719>
John-Mark Bell [Tue, 31 Oct 2017 09:40:33 +0000 (09:40 +0000)]
vp8enc: improve unit tests
- make test_encode_simple cope with libvpx built with
CONFIG_REALTIME_ONLY. Sadly, there's no way to detect this at
runtime beyond trying to set lag-in-frames to >0, pushing a
buffer and catching the GST_FLOW_NOT_NEGOTIATED return.
- fix bitrot in test_encode_simple_when_bitrate_set_to_zero.
- port test_encode_simple to GstHarness and introduce a separate
test for the lag-in-frames property.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/708>
Jakub Adam [Fri, 21 Aug 2020 14:03:09 +0000 (16:03 +0200)]
docs: Update plugin cache
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
Jakub Adam [Tue, 24 Mar 2020 18:35:07 +0000 (19:35 +0100)]
vpx: Support GST_VIDEO_FORMAT_I422_10LE
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
Jakub Adam [Tue, 24 Mar 2020 16:16:59 +0000 (17:16 +0100)]
vpx: Support GST_VIDEO_FORMAT_I420_10LE
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
Jakub Adam [Mon, 23 Mar 2020 20:44:30 +0000 (21:44 +0100)]
vp9enc: support GST_VIDEO_FORMAT_Y444
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/705>
Tim-Philipp Müller [Tue, 8 Sep 2020 16:30:35 +0000 (17:30 +0100)]
ci: include template from gst-ci master branch again
Tim-Philipp Müller [Tue, 8 Sep 2020 15:58:37 +0000 (16:58 +0100)]
Back to development
Tim-Philipp Müller [Mon, 7 Sep 2020 23:05:14 +0000 (00:05 +0100)]
Release 1.18.0
Tim-Philipp Müller [Mon, 7 Sep 2020 21:39:02 +0000 (22:39 +0100)]
meson: dist pot file in tarballs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/716>
Sebastian Dröge [Mon, 7 Sep 2020 09:13:18 +0000 (12:13 +0300)]
gst: Update for gst_video_transfer_function_*() function renaming
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/715>
Jan Alexander Steffens (heftig) [Mon, 31 Aug 2020 13:01:32 +0000 (15:01 +0200)]
flvmux: Avoid crash when best pad gets flushed
The 'best' pad might receive a flush event between us picking it and us
popping the buffer. In this case, the buffer will be missing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/711>
Jan Alexander Steffens (heftig) [Mon, 31 Aug 2020 11:43:42 +0000 (13:43 +0200)]
flvmux: Correct breaks in gst_flv_mux_find_best_pad
The code seems to use `continue` and `break` as if both refer to the
surrounding `while` loop. But because `break` breaks out of the
`switch`, they actually have the same effect.
This may have caused the loop not to terminate when it should. E.g. when
`skip_backwards_streams` drops a buffer we should abort the aggregation
and wait for all pads to be filled again. Instead, we might have just
selected a subsequent pad as our new "best".
Replace `break` with `done = TRUE; break`, and `continue` with `break`.
Then simplify the code a bit.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/710>
Dmitriy Purgin [Wed, 13 May 2020 09:31:38 +0000 (11:31 +0200)]
gstqmlgl: build on Windows with qmake without pkgconfig; update instructions on building for Windows
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/591>
Philipp Zabel [Fri, 21 Aug 2020 10:12:48 +0000 (12:12 +0200)]
meson: fix build failure if orc is enabled but none of its users are
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/778
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/704>
Zeid Bekli [Thu, 20 Aug 2020 12:26:04 +0000 (14:26 +0200)]
rtpL16depay: unref buffer on error
gst_rtp_L16_depay_process to unref buffer on wrong payload size or
reorder failure.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/702>
Tim-Philipp Müller [Thu, 20 Aug 2020 15:11:58 +0000 (16:11 +0100)]
Release 1.17.90
Sebastian Dröge [Tue, 18 Aug 2020 07:27:52 +0000 (10:27 +0300)]
rtputils: Don't call NULL GstMeta transform function
It's optional and if it does not exist then no transformation is
possible.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/701>
Julian Bouzas [Thu, 13 Aug 2020 19:27:25 +0000 (15:27 -0400)]
rtp: Do not register rtpreddec and rtpredenc twice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/699>
Sebastian Dröge [Wed, 12 Aug 2020 09:21:43 +0000 (12:21 +0300)]
rtpmanager: Improve readability of "stats" docs by making the fields an actual list
Otherwise they end up all in the same line one after another.
Also add docs for the "avg-jitter" stats field of the jitterbuffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/698>
Nicolas Dufresne [Tue, 11 Aug 2020 21:24:11 +0000 (17:24 -0400)]
v4l2h264codec: Map newly defined profile/levels
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/697>
Nicolas Dufresne [Tue, 11 Aug 2020 21:18:42 +0000 (17:18 -0400)]
v4l2: Sync headers with kernel 5.9
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/697>
Víctor Manuel Jáquez Leal [Thu, 6 Aug 2020 11:15:10 +0000 (13:15 +0200)]
v4l2: use GstV4l2Error in gst_v4l2_open()
gst_v4l2_open() is called by gst_v4l2_device_provider_probe_device(),
where the GstV4l2Object is created without an associated GstElement.
If gst_v4l2_open() fails, it raises a bus message, but without an
element, a precondition check fails on
gst_element_message_full_with_details() generating a crash if running
with fatal-warnings debug mode.
GstV4l2Error is a helper to raise error bus messages when it is
appropiated. This patch changes the direct bus messages to this
helper, and the elements will actually send the error message.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/694>
Vivia Nikolaidou [Mon, 10 Aug 2020 17:20:53 +0000 (20:20 +0300)]
flvmux: Return NEED_DATA when we drop a buffer
When we are dropping a buffer in find_best_pad (e.g. waiting for a
keyframe, or skipping backwards timestamp), return
GST_AGGREGATOR_FLOW_NEED_DATA to make sure we have enough data at the
next run. Otherwise, a stream that accidentally fell behind (e.g.
relinking race, or just waiting for a keyframe) will never get the
opportunity to catch up to the other one, because the other one will
always keep advancing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>
Vivia Nikolaidou [Mon, 10 Aug 2020 17:20:04 +0000 (20:20 +0300)]
flvmux: Return NEED_DATA when no best pad is found
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>
Vivia Nikolaidou [Mon, 10 Aug 2020 17:17:38 +0000 (20:17 +0300)]
flvmux: Fix possible crash on GST_ITERATOR_RESYNC
Wrong pointer type
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/696>
Matthew Waters [Mon, 10 Aug 2020 05:49:55 +0000 (15:49 +1000)]
qmlgloverlay: fix multiple elements with Qt 5.15
With Qt 5.15 multiple qmlgloverlay elements would produce:
ASSERT: "!m_gl->property(QSG_RENDERCONTEXT_PROPERTY).isValid()" in file /path/to/qt5/qtdeclarative/src/quick/scenegraph/qsgdefaultrendercontext.cpp, line 121
Workaround by setting the (seeminigly unused) property before
initialization.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/695>
Sebastian Dröge [Wed, 5 Aug 2020 07:41:33 +0000 (10:41 +0300)]
rtph26[45]pay: Change default aggregate-mode to "none" for backwards compatibility
We didn't aggregate at all in previous versions and there are apparently
various RTP implementations that don't handle aggregation well at all.
As part of this also document that for RTSP it is recommended to keep it
set to "none" while for WebRTC it should be set to "zero-latency".
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/749
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/692>
Matthew Waters [Fri, 24 Jul 2020 06:58:34 +0000 (16:58 +1000)]
build: update for gl pkg-config file split
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/680>
Jan Alexander Steffens (heftig) [Fri, 31 Jul 2020 11:50:13 +0000 (13:50 +0200)]
splitmuxsink: Make sure flushing doesn't block
* Trying to disconnect a stream from a running splitmuxsink by flushing
it results in the FLUSH_START blocking in the stream queue's
gst_pad_pause_task because the flush did not unblock
complete_or_wait_on_out, so add a check for ctx->flushing there.
* Add a GST_SPLITMUX_BROADCAST_INPUT so check_completed_gop notices
flushing changed and the incoming push is unblocked.
* Pass the FLUSH_STOP along to the muxer without waiting.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/687>
Vivia Nikolaidou [Tue, 4 Aug 2020 12:49:43 +0000 (15:49 +0300)]
imagefreeze: Wait until we have a clock
Otherwise it can happen that it tries to get the clock in PAUSED state
in live mode, which does not exist.
Thanks to Sebastian Dröge for helping debugging.
Fixes #775
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/691>
Tim-Philipp Müller [Fri, 31 Jul 2020 10:05:02 +0000 (11:05 +0100)]
qtdemux: extract bit depth from codec data for ALAC
The info in the sound sample description might not be
accurate if it's an older version atom.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/771
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/686>
Jordan Petridis [Tue, 28 Jul 2020 15:46:30 +0000 (18:46 +0300)]
auparse: fix compiler warnings
GCC 10 was complaining like following. It really is complaining about default cases returning
with potentially unitialized *desval, but those cases in the switch should never be hit.
```
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c: In function 'gst_au_parse_chain':
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:481:37: error: 'timestamp' may be used uninitialized in this function [-Werror=maybe-uninitialized]
481 | GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:482:36: error: 'duration' may be used uninitialized in this function [-Werror=maybe-uninitialized]
482 | GST_BUFFER_DURATION (outbuf) = duration;
../subprojects/gst-plugins-good/gst/auparse/gstauparse.c:480:34: error: 'offset' may be used uninitialized in this function [-Werror=maybe-uninitialized]
480 | GST_BUFFER_OFFSET (outbuf) = offset;
cc1: all warnings being treated as errors
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/671>
George Kiagiadakis [Wed, 29 Jul 2020 11:06:55 +0000 (14:06 +0300)]
rtspsrc: drop stream-start message posted by the internal udp sink(s)
See #1368
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/685>
Hosang Lee [Wed, 22 Jul 2020 07:24:15 +0000 (16:24 +0900)]
tests: qtdemux: test correct pad names are created
Test correct pad names are created in accordance to their media type
in mss mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/628>
Hosang Lee [Tue, 16 Jun 2020 08:23:44 +0000 (17:23 +0900)]
qtdemux: create correct pad names in encrypted streams
Refer to "original-media-type" when setting stream's subtype
for encrypted streams in mss mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/628>
Thibault Saunier [Wed, 22 Jul 2020 18:31:13 +0000 (14:31 -0400)]
matroskamux: Do caps renegotiation when it only adds fields
Matroskamux can accept caps renegotiation if the new caps is a
superset of the old one, meaning upstream added new info to
the caps.
Same logic as
a5f22f03aa25b04726f78ae619f40b3b648f7d48 in qtmux.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/678>
Tim-Philipp Müller [Fri, 24 Jul 2020 13:02:26 +0000 (14:02 +0100)]
rtpfunnel: protect internal srccaps with lock
These are modified from sink pad event handlers, so
could be accessed from multiple threads at the same
time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/681>
Havard Graff [Sun, 23 Feb 2020 22:44:16 +0000 (23:44 +0100)]
rtpfunnel: copy caps before sending them in a caps-event
Reason being we don't want downstream to own a ref to our
internal caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/681>
Mathieu Duponchelle [Mon, 27 Jul 2020 13:41:26 +0000 (15:41 +0200)]
rtpmanager: fix various documentation issues
Improper naming of properties, improper links, misc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/684>
Tim-Philipp Müller [Fri, 24 Jul 2020 16:13:04 +0000 (17:13 +0100)]
rpicamsrc: hypothetical fix for data pointer calculation
mmal buffer header docs say data is valid for length bytes
from offset. In practice offset always seems to be 0 so
far though.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>
Tim-Philipp Müller [Fri, 24 Jul 2020 15:35:43 +0000 (16:35 +0100)]
rpicamsrc: mark buffers as header and keyframe/delta-unit
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>
Tim-Philipp Müller [Fri, 24 Jul 2020 15:14:00 +0000 (16:14 +0100)]
rpicamsrc: fix nal alignment of output buffers
We claim output buffers are nal-aligned, but that wasn't
actually true: We would push out a partial nal in case
the nal doesn't fit into the max encoder-selected output
buffer size, and then the next buffer would not start
with a sync marker. That's not right and makes h264parse
unhappy.
Instead accumulate buffers until we have a full frame
(we can't rely on the NAL_END flag, it's always set).
Fixes #768
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>
Tim-Philipp Müller [Mon, 13 Jul 2020 22:43:48 +0000 (23:43 +0100)]
rpicamsrc: fix "Could not find component vc.ril.camera" on recent raspios
Make extra sure all the required mmal libs such as libmmal_vc_client.so
actually get linked and stay linked. Otherwise the above error happens
it seems.
buster (10.4) with meson 0.55 and pi ref 2020-05-27
pi-gen,
825107f04027269db77426046f5085475b1ea22f, stage5
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>
Tim-Philipp Müller [Mon, 13 Jul 2020 16:01:42 +0000 (17:01 +0100)]
rpicamsrc: deviceprovider: hook up i18n properly
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>
Tim-Philipp Müller [Mon, 13 Jul 2020 15:55:48 +0000 (16:55 +0100)]
rpicamsrc: deviceprovider: advertise (M)JPEG as well
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/682>