False positive for the three variables but some warnings like:
../tests/check/elements/matroskamux.c:875:10:
warning: 'chapters_offset' may be used uninitialized in this function [-Wmaybe-uninitialized]
*index = chapters_offset;
~~~~~~~^~~~~~~~~~~~~~~~~
The above is false positive as there is a gboolean to check if it was
initialized or not (found_chapters_declaration).
This reverts commit dcd3ce9751.
This functionality was implemented for gstopenwebrtc, but it
turned out this was not actually needed for webrtc bundling
support, as shown in webrtcbin. It also doesn't correspond
to any standards.
This is an API break, but nothing should actually depend on
this, at least not for its initial purpose.
Changes in rtpbin.c were reverted manually, to preserve some
refactoring that had occurred in the original commit.
Fixes#537
gstreamer!55 makes some changes to how the `error-after` counter works
which breaks this test. This change makes the test not rely on the
ability to alter `error-after` at runtime and explicitly stops and
starts the harness before pushing data.
An alternative would be to add another argument to
`harness_rtpulpfecdec` to set `error-after` on construction but that's
slightly more long-winded. so I went for this approach instead.
Fixes#532, even though that's already closed.
The initial mission statement for this test was:
* demonstrate usage of the request-aux-* signals in rtpbin
* test the rtx elements
We have examples that serve the first use case, and better
(harnessed) tests for the second use case.
This test is slow and racy, it served its purpose but can now
be removed.
Fixes#533
When the EOS event is received, run all timers immediately and avoid
pushing the EOS downstream before this has been run. This ensures that
the lost packet statistics are accurate.
The teardown of the pads checks the refcount, but there are timers
inside the jitterbuffer that can push things, so if we're not lucky,
things could be pushed while the pads are being shut down. Putting the
jitterbuffer to NULL first avoids this.
Always wait with starting the RTCP thread until either a RTP or RTCP
packet is sent or received. Special handling is needed to make sure the
RTCP thread is started when requesting an early RTCP packet.
We want to wait with starting the RTCP thread until it's needed in order
to not send RTCP packets for an inactive source.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
This mode is useful for muxers that can take a long time to finalize a
file. Instead of blocking the whole upstream pipeline while the muxer is
doing its stuff, we can unlink it and spawn a new muxer+sink combination
to continue running normally.
This requires us to receive the muxer and sink (if needed) as factories,
optionally accompanied by their respective properties structures. Also
added the muxer-added and sink-added signals, in case custom code has to
be called for them.
https://bugzilla.gnome.org/show_bug.cgi?id=783754
If obtain_internal_source() returns a source that is not internal it
means there exists a non-internal source with the same ssrc. Such an
ssrc collision should be handled by sending a GstRTPCollision event
upstream and choose a new ssrc, but for now we simply drop the packet.
Trying to process the packet further will cause it to be pushed
usptream (!) since the source is not internal (see source_push_rtp()).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
If there is an external source which is about to timeout and be removed
from the source hashtable and we receive feedback RTCP packet with the
media ssrc of the source, we unlock the session in
rtp_session_process_feedback before emitting 'on-feedback-rtcp' signal
allowing rtcp timer to kick in and grab the lock. It will get rid of
the source and rtp_session_process_feedback will be left with RTPSource
with ref count 0.
The fix is to grab the ref to the RTPSource object in
rtp_session_process_feedback.
https://bugzilla.gnome.org/show_bug.cgi?id=795139
These are the sources we send from, so there is no reason to
report receive statistics for them (as we do not receive on them,
and the remote side has no knowledge of them).
https://bugzilla.gnome.org/show_bug.cgi?id=795139
ULP FEC, as defined in RFC 5109, has the protected and protection
packets sharing the same ssrc, and a different payload type, and
implies rewriting the seqnums of the protected stream when encoding
the protection packets. This has the unfortunate drawback of not
being able to tell whether a lost packet was a protection packet.
rtpbasedepayload relies on gaps in the seqnums to set the DISCONT
flag on buffers it outputs. Before that commit, this created two
problems:
* The protection packets don't make it as far as the depayloader,
which means it will mark buffers as DISCONT every time the previous
packets were protected
* While we could work around the previous issue by looking at
the protection packets ignored and dropped in rtpptdemux, we
would still mark buffers as DISCONT when a FEC packet was lost,
as we cannot know that it was indeed a FEC packet, even though
this should have no impact on the decoding of the stream
With this commit, we consider that when using ULPFEC, gaps in
the seqnums are not a reliable indicator of whether buffers should
be marked as DISCONT or not, and thus rewrite the seqnums on
the decoding side as well to form a perfect sequence, this
obviously doesn't prevent the jitterbuffer from doing its job
as the ulpfec decoder is downstream from it.
https://bugzilla.gnome.org/show_bug.cgi?id=794909
Looping the test 500 times to only execute the test once every
33 times means we inited and deinited gstreamer 467 times
for no reason at all, which was annoying when running the test
with valgrind.
We expose a set of new elements:
* ULPFEC encoder / decoder
* A storage element, which should be placed before jitterbuffers,
and is used to store packets in order to attempt reconstruction
after the jitterbuffer has sent PacketLost events
* RED encoder / decoder (RFC 2198), these are necessary to
use FEC in webrtc, as browsers will propose and expect ulpfec
packets to be wrapped in red packets
With contributions from:
Mathieu Duponchelle <mathieu@centricular.com>
Sebastian Dröge <sebastian@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792696
When the signal returns a floating reference, as its return type
is transfer full, we need to sink it ourselves before passing
it to gst_bin_add (which is transfer floating).
This allows us to unref it in bin_remove_element later on, and
thus to also release the reference we now own if the signal
returns a non-floating reference as well.
As we now still hold a reference to the element when removing it,
we also need to lock its state and setting it to NULL before
unreffing it
Also update the request_aux_sender test.
https://bugzilla.gnome.org/show_bug.cgi?id=792543
TOC support in mastroskamux has been deactivated for a couple of years. This commit updates it to recent GstToc evolutions and introduces toc unit tests for both matroska-mux and matroska-demux.
There are two UIDs for Chapters in Matroska's specifications:
- The ChapterUID is a mandatory unsigned integer which internally refers to a given chapter. Except for title & language which use dedicated fields, this UID can also be used to add tags to the Chapter. The tags come in a separate section of the container.
- The ChapterStringUID is an optional UTF-8 string which also uniquely refers to a chapter but from an external perspective. It can act as a "WebVTT cue identifier" which "can be used to reference a specific cue, for example from script or CSS".
During muxing, the ChapterUID is generated and checked for unicity, while the ChapterStringUID receives the user defined UID. In order to be able to refer to chapters from the tags section, we maintain an internal Toc tree with the generated ChapterUID.
When demuxing, the ChapterStringUIDs (if available) are assigned to the GstTocEntries UIDs and an internal toc mimicking the toc is used to keep track of the ChapterUIDs and match the tags with the appropriate GstTocEntries.
https://bugzilla.gnome.org/show_bug.cgi?id=790686
Sometimes all the buffers are received before the time we lock the
check_mutex, in which case g_cond_wait will wait forever for another
one. Just check if this is the case before waiting.
https://bugzilla.gnome.org/attachment.cgi?id=358397
This patch simplifies the tests (44% less code) and
makes them much more readable.
The provided SessionHarness also makes it much easier
to write new tests for rtpsession.
https://bugzilla.gnome.org/show_bug.cgi?id=791070
If the use-robust-muxing property is set, check if the
assigned muxer has reserved-max-duration and
reserved-duration-remaining properties, and if so set
the configured maximum duration to the reserved-max-duration
property, and monitor the remaining space to start
a new file if the reserved header space is about to run out -
even though it never ought to.
Switching to a new fragment because the input caps have
changed didn't properly end the previous file. Use the normal
EOS sequence to ensure that happens. Add a test that it works.
SoupSession's ssl-ca-file property is deprecated. Use the recommended
tls-database property.
This is a bit more complex as it requires creating a GTlsFileDatabase
object for an absolute (!) path to the CA certificates file.
https://bugzilla.gnome.org/show_bug.cgi?id=784005
Even though hooked up to the build system, it's clear that no one
has ever built or used this with GStreamer 1.x. It wants to link
against libgstinterfaces, which no longer exists. And uses 0.10-style
raw audio caps. And the last meaningful change was done in 2009.
Let's just remove it.
streamheader and codec_data buffers fields are only meant to be
in the negotiated caps, not the template caps.
Fixes false-positive leaks of those buffers detected by the leaks
tracer, as template caps are static, and we decided to not include
code in gstreamer core to handle this unusual case of template caps
having buffers in them.
https://bugzilla.gnome.org/show_bug.cgi?id=768762
Some radio streams uses StreamTitle='' to reset the title after a
track stopped playing, e.g. while the host talks between tracks or
during news segments.
This change forces an empty tag object to be distributed if
StreamTitle or StreamUrl is received with empty value, thus allowing
downstream elements to get notified about this.
https://bugzilla.gnome.org/show_bug.cgi?id=778437
Add a new signal for formatting the filename, which receives
a GstSample containing the first buffer from the reference
stream that will be muxed into that file.
Useful for creating filenames that are based on the
running time or other attributes of the buffer.
To make it work, opening of files and setting filenames is
now deferred until there is some data to write to it,
which also requires some changes to how async state changes
and gap events are handled.
Now matroskamux mark all packets of audio-only streams as keyframes so
in test_block_group after pushing the test audio data 4 buffers are produced
and not more 2. The last buffer is the original data and must match with what
pushed. The remaining ones are matroskamux headers
https://bugzilla.gnome.org/show_bug.cgi?id=754696
* Changed PCMU->TEST for common macros
* Changed verify-functions (lost & rtx) into macros.
* Remove option to add marker-bit for test-buffers (not used anywhere)
* Add new push_test_buffer function that makes sure there are correlation
between dts and the time on the clock. (classic test-mistake)
* Established a generic starting-point for tests with the
construct_deterministic_initial_state function and use it where
applicable, which removes lots of "boilerplate" everywhere.
* Add basic lost-event test
* Remove as much "magic constants" as possible.
* Remove 3 tests that no longer are testing anything that others don't,
and was completely unmaintainable.
* Remove unnecessary use of the testclock
* Verify each test is testing what it actually says it does (and modify
where it doesn't)
In general, make the tests much smaller, better, more maintainable and
readable.
https://bugzilla.gnome.org/show_bug.cgi?id=774409
A new signal named on-bundled-ssrc is provided and can be
used by the application to redirect a stream to a different
GstRtpSession or to keep the RTX stream grouped within the
GstRtpSession of the same media type.
https://bugzilla.gnome.org/show_bug.cgi?id=772740
When doing rtx, the jitterbuffer will always add an rtx-timer for the next
sequence number.
In the case of the packet corresponding to that sequence number arriving,
that same timer will be reused, and simply moved on to wait for the
following sequence number etc.
Once an rtx-timer expires (after all retries), it will be rescheduled as
a lost-timer instead for the same sequence number.
Now, if this particular sequence-number now arrives (after the timer has
become a lost-timer), the reuse mechanism *should* now set a new
rtx-timer for the next sequence number, but the bug is that it does
not change the timer-type, and hence schedules a lost-timer for that
following sequence number, with the result that you will have a very
early lost-event for a packet that might still arrive, and you will
never be able to send any rtx for this packet.
Found by Erlend Graff - erlend@pexip.comhttps://bugzilla.gnome.org/show_bug.cgi?id=773891
The lost-event was using a different time-domain (dts) than the outgoing
buffers (pts). Given certain network-conditions these two would become
sufficiently different and the lost-event contained timestamp/duration
that was really wrong. As an example GstAudioDecoder could produce
a stream that jumps back and forth in time after receiving a lost-event.
The previous behavior calculated the pts (based on the rtptime) inside the
rtp_jitter_buffer_insert function, but now this functionality has been
refactored into a new function rtp_jitter_buffer_calculate_pts that is
called much earlier in the _chain function to make pts available to
various calculations that wrongly used dts previously
(like the lost-event).
There are however two calculations where using dts is the right thing to
do: calculating the receive-jitter and the rtx-round-trip-time, where the
arrival time of the buffer from the network is the right metric
(and is what dts in fact is today).
The patch also adds two tests regarding B-frames or the
“rtptime-going-backwards”-scenario, as there were some concerns that this
patch might break this behavior (which the tests shows it does not).
The new timeout is always going to be (timeout + delay), however, the
old behavior compared the current timeout to just (timeout), basically
being (delay) off.
This would happen if rtx-delay == rtx-retry-timeout, with the result that
a second rtx attempt for any buffers would be scheduled immediately instead
of after rtx-delay ms.
Simply calculate (new_timeout = timeout + delay) and then use that instead.
https://bugzilla.gnome.org/show_bug.cgi?id=773905
It's been broken for years, and it's unlikely it will ever
be fixed for collectpads/videomixer now that there's compositor
which works fine. So let's disable it, since all it does
is that it creates noise that distracts from other failures.
Also see the corresponding adder bug as it failed in the same way:
https://bugzilla.gnome.org/show_bug.cgi?id=708891
It seems that the forked processes all attempt to handle the listening
socket from the server, and only one has to shutdown the socket to break
the server completely.
Create a new server inside each test to avoid this.
https://bugzilla.gnome.org/show_bug.cgi?id=772656
The tests accumulate buffers in GstCheck's buffers list, and the list is
not (consistently) reset between tests. Do that and remove the now
conflicting unrefs for outbuffers.
https://bugzilla.gnome.org/show_bug.cgi?id=772644
The basic idea is this:
1. For *larger* rtx-rtt, weigh a new measurement as before
2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less
3. For very large measurements, consider them "outliers"
and count them a lot less
The idea being that reducing the rtx-rtt is much more harmful then
increasing it, since we don't want to be underestimating the rtt of the
network, and when using this number to estimate the latency you need for
you jitterbuffer, you would rather want it to be a bit larger then a bit
smaller, potentially losing rtx-packets. The "outlier-detector" is there
to prevent a single skewed measurement to affect the outcome too much.
On wireless networks, these are surprisingly common.
https://bugzilla.gnome.org/show_bug.cgi?id=769768
Assuming equidistant packet spacing when that's not true leads to more
loss than necessary in the case of reordering and jitter. Typically this
is true for video where one frame often consists of multiple packets
with the same rtp timestamp. In this case it's better to assume that the
missing packets have the same timestamp as the last received packet, so
that the scheduled lost timer does not time out too early causing the
packets to be considered lost even though they may arrive in time.
https://bugzilla.gnome.org/show_bug.cgi?id=769768
There is no need to schedule another EXPECTED timer if we're already
past the retry period. Under normal operation this won't happen, but if
there are more timers than the jitterbuffer is able to process in
real-time, scheduling more timers will just make the situation worse.
Instead, consider this packet as lost and move on. This scenario can
occur with high loss rate, low rtt and high configured latency.
https://bugzilla.gnome.org/show_bug.cgi?id=769768
This patch fixes an issue with the estimated gap duration when there is
a gap immediately after a lost timer has been processed. Previously
there was a discrepancy beteen the gap in seqnum and gap in dts which
would cause wrong calculated duration. The issue would only be seen with
retranmission enabled since when it's disabled lost timers are only
created when a packet is received and the actual gap length and last dts
is known.
https://bugzilla.gnome.org/show_bug.cgi?id=769768
Stats should also be collected for unsuccessful packets.
rtx-rtt is very important for determining the necessary configured
latency on the jitterbuffer. It's especially important to be able to
increase the latency when retransmitted packets arrive too late and are
considered lost. This patch includes these late packets in the
calculation of the various rtx stats, making them more correct and
useful.
Also in the case where the original packet arrives after a NACK is sent,
the received RTX packet should update the stats since it provides useful
information about RTT.
The RTT is only updated if and only if all requested retranmissions are
received. That way the RTT is guaranteed to make sense. If not we don't
know which request the packet is a response to and the RTT may be bogus.
A consequence of this patch is that RTT is not updated for a request
when one of the RTX packets for that seqnum is lost, but that since
measured RTT will be more accurate.
The implementation store the RTX information from the timed out timers
and use this when the retransmitted packet arrives. For performance
these timers are stored separately from the "normal" timers in order to
not impact performance (see attached performance test).
https://bugzilla.gnome.org/show_bug.cgi?id=769768
Under certain conditions gst_rtp_buffer_get_payload() returns a copy of
the payload. In this case the payload modifications will not affect the
rtp buffer. So instead of modifying the payload buffer directly we
should modify the buffer that actually gets pushed on the adapter.
The functionality of all the tests was kept exactly the same. Some tests
were renamed:
test_push_forward_seq -> test_rtxsend_rtxreceive
test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss
test_rtxreceive_data_reconstruction was testing that retransmitted
buffer produced by rtxsend was correctly transformed to the original
buffer by rtxreceive. Now we are checking for this in all the tests
where both rtxsend & rtxreceive are involved. That's why the test was
removed.
Need to set max-misorder-time and max-dropout-time to 0 so the
jitterbuffer does not base them on packet rate calculations.
If it does, out gap is big enough to be considered a new stream and
we wait for a few consecutive packets just to be sure
https://bugzilla.gnome.org/show_bug.cgi?id=751311
When parsing NAL unit type in codec_data, check the 6bits of
NAL_unit_type only and do not require the array_completeness bit to be
0, since the default and mandatory value of array_completeness is 1 for
hvc1.
https://bugzilla.gnome.org/show_bug.cgi?id=768653
Handle sprop-vps, sprop-sps and sprop-pps in caps instead of
sprop-parameter-sets.
rtph265pay works with byte-stream and hvc1 formats but not hev1 yet. It
handles profile-id, tier-flag and level-id in caps query.
https://bugzilla.gnome.org/show_bug.cgi?id=753760
This is to handle cases where upstream handles the fragmented streaming in TIME
segments and sends us data with gaps within fragments. This would happen when dealing
with trick-modes.
When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
it must obey the following rules:
* The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
* The buffers containing the first sample after a gap:
* MUST start at the beginning of a sample,
* MUST have the DISCONT flag set,
* MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.
https://bugzilla.gnome.org/show_bug.cgi?id=767354
Some endpoints (like Tandberg E20) can send BYE packet containing our
internal SSRC. I this case we would detect SSRC collision and get rid
of the source at some point. But because we are still sending packets
with that SSRC the source will be recreated immediately.
This brand new internal source will not have some variables incorrectly
set in its state. For example 'seqnum-base` and `clock-rate` values will be
-1.
The fix is not to act on BYE RTCP if it contains internal or unknown
SSRC.
https://bugzilla.gnome.org/show_bug.cgi?id=762219
Keeping the lock while emitting the stats signal introduces potential
deadlock in those situations when the signal callback wants the access
to rtpsession's properties which also requre the lock.
https://bugzilla.gnome.org/show_bug.cgi?id=762216
Avoid using soup_server_run_async and old get_port() APIs,
replace with me soup_server_listen and get the port through the
URIs list returned from the server.
When a packet arrives that has already been considered lost as part of a
large gap the "lost timer" for this will be cancelled. If the remaining
packets of this large gap never arrives, there will be missing entries
in the queue and the loop function will keep waiting for these packets
to arrive and never push another packet, effectively stalling the
pipeline.
The proposed fix conciders parts of a large gap definitely lost (since
they are calculated from latency) and ignores the late arrivals.
In practice the issue is rare since large gaps are scheduled immediately,
and for the stall to happen the late arrival needs to be processed
before this times out.
https://bugzilla.gnome.org/show_bug.cgi?id=765933
