Commit graph

969 commits

Author SHA1 Message Date
Thomas Bluemel
8d955fc32b rtpjitterbuffer: Only calculate skew or reset if no gap.
In the case of reordered packets, calculating skew would cause
pts values to be off. Only calculate skew when packets come
in as expected. Also, late RTX packets should not trigger
clock skew adjustments.

Fixes #612
2019-07-03 06:23:07 -06:00
Jan Alexander Steffens (heftig)
91e858dcbe
test: flvmux: Test changing caps with one sinkpad
These tests segfault without the preceding crash fix.
2019-06-19 14:36:21 +02:00
Jan Alexander Steffens (heftig)
daafce54ac
test: flvmux: Use gst_harness_sink_push_many
And check its return value.
2019-06-19 14:36:21 +02:00
Mathieu Duponchelle
ebe2756434 jitterbuffer: unset DTS on output buffers 2019-06-14 16:02:59 +02:00
Mikhail Fludkov
ec5fa49631 rtpjitterbuffer: late packets shouldn't affect PTS of the following packet
If, say, a rtx-packet arrives really late, this can have a dramatic
effect on the jitterbuffer clock-skew logic, having it being reset
and losing track of the current dts-to-pts calculations, directly affecting
the packets that arrive later.

This is demonstrated in the test, where a RTX packet is pushed in really
late, and without this patch the last packet will have its PTS affected
by this, where as a late RTX packet should be redundant information, and
not affect anything.
2019-06-13 11:55:10 +02:00
Mikhail Fludkov
b9c3e354ee rtpjitterbuffer: fix rtx delay calulation when large packet spacing 2019-06-12 11:39:32 +02:00
Stian Selnes
6269ed49ab rtpjitterbuffer: Fix delay for EXPECTED timers added by gaps
This patch corrects the delay set on EXPECTED timers that are added when
processing gaps. Previously the delay could be too small so that
'timout + delay' was much less than 'now', causing the following retries
to be scheduled too early. (They were sent earlier than
rtx-retry-timeout after the previous timeout.)
2019-06-12 11:39:32 +02:00
Havard Graff
8ed7ab178b rtpjitterbuffer: don't try and calculate packet-rate if seqnum are jumping
Turns out that the "big-gap"-logic of the jitterbuffer has been horribly
broken.

For people using lost-events, an RTP-stream with a gap in sequencenumbers,
would produce exactly that many lost-events immediately.
So if your sequence-numbers jumped 20000, you would get 20000 lost-events
in your pipeline...

The test that looks after this logic "test_push_big_gap", basically
incremented the DTS of the buffer equal to the gap that was introduced,
so that in fact this would be more of a "large pause" test, than an
actual gap/discontinuity in the sequencenumbers.

Once the test was modified to not increment DTS (buffer arrival time) with
a similar gap, all sorts of crazy started happening, including adding
thousands of timers, and the logic that should have kicked in, the
"handle_big_gap_buffer"-logic, was not called at all, why?

Because the number max_dropout is calculated using the packet-rate, and
the packet-rate logic would, in this particular test, report that
the new packet rate was over 400000 packets per second!!!

I believe the right fix is to don't try and update the packet-rate if
there is any jumps in the sequence-numbers, and only do these calculations
for nice, sequential streams.
2019-06-12 11:39:31 +02:00
Havard Graff
dd422f0b7f rtpjitterbuffer: fix unused variables 2019-06-12 11:39:31 +02:00
Nicolas Dufresne
f7c712d0b8 rtpssrcdemux: Avoid taking streamlock out-of-band
In this change we now protect the internal srcpads list using the
stream lock and limit usage of the internal stream lock to
preventing data flowing on the other src pad type while creating
and signalling the new pad.

This fixes a deadlock with RTPBin shutdown lock. These two locks would
end up being taken in two different order, which caused a deadlock. More
generally, we should not rely on a streamlock when handling out-of-band
data, so as a side effect, we should not take a stream lock when
iterating internal links.
2019-06-04 09:26:06 -04:00
Nicolas Dufresne
947a37f3c8 rtpsession: Always keep at least one NACK on early RTCP
We recently added code to remove outdate NACK to avoid using bandwidth
for packet that have no chance of arriving on time. Though, this had a
side effect, which is that it was to get an early RTCP packet with no
feedback into it. This was pretty useless but also had a side effect,
which is that the RTX RTT value would never be updated. So we we stared
having late RTX request due to high RTT, we'd never manage to recover.

This fixes the regression by making sure we keep at least one NACK in
this situation. This is really light on the bandwidth and allow for
quick recover after the RTT have spiked higher then the jitterbuffer
capacity.
2019-05-17 19:13:22 +00:00
Nicolas Dufresne
84c102b6fe rtpsession: Call on-new-ssrc earlier
Right now, we may call on-new-ssrc after we have processed the first
RTP packet. This prevents properly configuring the source as some
property like "probation" are copied internally for use as a
decreasing counter. For this specific property, it prevents the
application from disabling probation on auxiliary sparse stream.

Probation is harmful on sparse streams since the probation algorithm
assume frequent and contiguous RTP packets.
2019-05-02 14:44:58 -04:00
Nicolas Dufresne
a8ea9f0d05 test: rtpsession: Verify on-sending-nacks callback 2019-04-07 12:00:49 -04:00
Mathieu Duponchelle
280d86a841 rtpsession: Add disable-sr-timestamp property
The Onvif Streaming Spec, in section 6.11, mandates that when
Rate-Control is disabled potential RTCP packets shall have
their timestamps set to 0.

<https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf>
2019-04-05 20:23:08 +02:00
Nicolas Dufresne
464ada3f29 test: rtpsession: Test FB Nack packing
We used to split the NACK if a smaller seqnum of a range of seqnum was
submited. This test also make sure that the three operations (append,
prepend, update) works properly.
2019-04-05 14:53:09 +00:00
Nicolas Dufresne
a31569713f test: rtpsession: Test handling of NACK surplus
This test verify that NACKs that didn't fit in one packet are properly
filtered and inserted into the following pipeline.
2019-04-05 14:53:09 +00:00
John Bassett
74a74bfc99 rtpsession: Fix race when sending PLI, FIR and NACK packets
Calling rtp_session_send_rtcp before marking the source as requiring a
pli/fir/nack meant the rtcp_thread could be scheduled and start running
before the source was updated. This meant the request would not be sent
early but instead was transmitted with the next regular RTCP packet.

Add test for nack generation.
2019-04-05 14:53:09 +00:00
Antonio Ospite
6f12f1cecc test: rtpbin_buffer_list: add test to verify that stats are correct
Add a test to verify that stats about sent and received packets are
correct even when using buffer lists.

NOTE: the newly introduced get_session_source_stats() selects the
desired source (sender or receiver) by filtering them by type (using the
get_sender parameter) rather than by ssrc because this simplifies the
code and it's good enough for testing purposes as there is usually one
source per type in the test setup.

Filtering by ssrc would have required handling asynchronous signals like
"on-new-sender-ssrc", with the relative locking, just to retrieve the
actual ssrc of the sender.
2019-04-02 13:02:28 +02:00
Antonio Ospite
a330012d6a test: rtpbin_buffer_list: move buffer list creation next to its validation
The tests create a buffer list and then use the chain_list callback to
verify that the correct packets have been pushed.

Move the creation and validation code next to each other so that the
reader can more easily understand what is going on.

While at it add some comments to introduce the two related functions.
2019-04-01 18:42:32 +00:00
Antonio Ospite
af8698e656 test: rtpbin_buffer_list: set the chain_list function directly in the test
The helper function set_chain_function does not really do anything useful, remove it.
2019-04-01 18:42:32 +00:00
Antonio Ospite
5567066bff test: rtpbin_buffer_list: make check_packet more flexible
Make it possible to differentiate between the position in the list and
the packet index in the global structures in check_packet, in some
future case the list may change, in case some element removes a buffer
from the list, and the two indices may not coincide.
2019-04-01 18:42:32 +00:00
Antonio Ospite
5807d79a9d test: rtpbin_buffer_list: factor out a function to create packets buffers 2019-04-01 18:42:32 +00:00
Antonio Ospite
6ab13c19b6 test: rtpbin_buffer_list: check if the chain_list function has been called
Make the test more useful to verify that the chain list function has
actually been called.
2019-04-01 18:42:32 +00:00
Antonio Ospite
623d3b930e test: rtpbin_buffer_list: port to GStreamer 1.0
Port the rtpbin_buffer_list test to GStreamer 1.0 and re-enable it.

Some other changes include:
  - the check on the caps has been moved from the buffer level to the
    pad level;
  - remove underscore prefix from static functions names, this is not
    idiomatic in C and rarely used in the other tests;
  - the unused header_buffer variable has been removed;
  - check_group() has been renamed to check_packet() because in
    GStreamer 1.0 there is no concept of "group" anymore, the comments
    have also been updated to reflect this.
2019-04-01 18:42:32 +00:00
Tim-Philipp Müller
abef4e4ea3 tests: jpegdec: bump discoverer timeout for valgrind
Tests might take a bit longer, esp. when run under valgrind
and/or they're running on the CI with other things going on,
so let's just bump the timeout to something higher and let
the test runner time us out if needed.
2019-04-01 18:20:53 +01:00
Olivier Crête
785219a317 rtpstorage: Make debug category available to sub objects 2019-03-26 19:41:06 -04:00
Antonio Ospite
2513edf229 test: imagefreeze: add test for the num-buffers property 2019-03-14 09:12:28 +01:00
Tim-Philipp Müller
a2d01b3a8b tests: rtpulpfec: fix buffer leak in unit test
This freed wrapped memory instead of the GstMemory or buffer.
2019-03-06 19:40:10 +00:00
Tim-Philipp Müller
081da67444 tests: rtpjitterbuffer: fix leaks in new test_push_eos() test 2019-03-06 17:28:57 +00:00
Tim-Philipp Müller
6b68b73341 tests: .gitignore more test and example binaries 2019-03-06 17:26:03 +00:00
Olivier Crête
6530fa53f2 rtp jitterbuffer test: Test for queue filling 2019-02-11 23:41:14 +00:00
Nirbheek Chauhan
062f2c46fa misc: Fix warnings on Cerbero's mingw (gcc 4.7)
error: this decimal constant is unsigned only in ISO C90 [-Werror]
2019-02-06 14:28:54 +00:00
Nicolas Dufresne
0725e54d6c test: h265depay: Add todo for testing aggregate packets with marker
We are missing a sample to test this, but a fix has been made, so add a
todo.
2019-01-31 19:30:14 +00:00
Nicolas Dufresne
cf3da6a443 test: rtph264depay: Check handling of STAP-A marker
Related to #557
2019-01-31 19:30:14 +00:00
Victor Toso
4a33b083f1 tests: rtp-payloading avoid -Wmaybe-uninitialized
More false positives as both of them are initialized in the line
before they are used, wrapped with fail_unless() check.
2019-01-18 13:53:18 +00:00
Victor Toso
2f77d877c3 tests: matroskamux avoid -Wmaybe-uninitialized
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).
2019-01-18 13:53:18 +00:00
Jan Alexander Steffens (heftig)
8f8de410c5 test: rtph265pay: Verify we only mark the last fragment 2019-01-09 15:36:40 +00:00
Jan Alexander Steffens (heftig)
03d138985f test: rtph265pay: Use a bigger test frame
The existing frame's last slice is too small to be used for
fragmentation tests.
2019-01-09 15:36:40 +00:00
Jan Alexander Steffens (heftig)
791711f9be test: rtph264pay: Verify we only mark the last fragment 2019-01-09 15:36:40 +00:00
Seungha Yang
cc5ee5f673 tests: Remove pointless unistd.h include 2018-12-30 21:54:44 +09:00
Mathieu Duponchelle
f52e16ceb8 Revert "rtpbin: receive bundle support"
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
2018-12-20 13:25:10 +00:00
Nicolas Dufresne
6941079d8d test: rtph265: Copy and port tests from rtph264
This copy and port all the relevant tests from rtph264.
2018-12-18 13:39:54 -05:00
Nicolas Dufresne
a0c58a77dc test: rtph264depay: Check the marker is converted to flag 2018-12-18 13:39:54 -05:00
Nicolas Dufresne
6b89144c9c test: rtph264depay: Check that EOS drains the depayloaded
In AU mode, the depayloader may have accumulated NALs, test that
these NALs are drained and not dropped.
2018-12-18 13:39:54 -05:00
Nicolas Dufresne
aa7e78b8e4 test: rtph264pay: Add tests for marker bit
Test that marker bit is transferred when input buffer has the
marker flag set but also that it's set whenever the payloader
receives complete AU.
2018-12-18 13:39:54 -05:00
Nicolas Dufresne
73ee9cdea2 test: rtph264pay: Verify slices timestamp
This test make sure that timestamps are properly transfered
to each NALU.
2018-12-18 13:39:54 -05:00
Nicolas Dufresne
cd09a3103f test: rtph264pay: Add reserved nals test 2018-12-18 13:39:54 -05:00
Jonny Lamb
9a3e8ad2d7 rtpulpfec: stop and start the harness when setting error-after
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.
2018-12-18 12:32:48 +00:00
Mathieu Duponchelle
306d5021e5 tests: remove rtpaux test
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
2018-12-18 11:08:50 +00:00
Olivier Crête
59d398b66c rtpjitterbuffer tests: Validate the number of buffers 2018-12-14 12:10:16 +00:00
Olivier Crête
d857522237 rtpjitterbuffer: Run all timers immediately on EOS
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.
2018-12-14 12:10:16 +00:00
Olivier Crête
c6e8325945 rtpjitterbuffer test: Stop jitterbuffer before pads to avoid race
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.
2018-12-14 12:10:16 +00:00
Nicolas Dufresne
c596bdda38 test: rtpssrcdemux: Test event forwarding
This the first unit test of this element. It adds a test that verify
that events are forwarded correctly.
2018-11-29 15:19:17 -05:00
Jordan Petridis
515ada7e22
Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 05:52:16 +02:00
Linus Svensson
ac94c706da rtpsession: test: Plug memory leak 2018-11-13 12:30:35 +00:00
Havard Graff
65a7d39bd4 flvmux: Test that timestamps are always increasing
Decreasing timestamps break rtmpsink.

With contributions from Olivier Crête.

https://bugzilla.gnome.org/show_bug.cgi?id=796382
2018-11-05 18:17:04 -05:00
Olivier Crête
cc69c876fe rtpsession: Allow changing the SDES at runtime
Make it possible to modify the SDES in a packet at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=763502
2018-10-28 12:10:36 +00:00
Yeongjin Jeong
301142604e tests: flvmux: Fix pushing invalid audio caps in tests
Previous commit created caps with incorrect aac codec data
that did not match the audio channel.

https://bugzilla.gnome.org/show_bug.cgi?id=797256
2018-10-21 02:44:24 -04:00
Havard Graff
13aa805943 rtpsession: fix up GHashTable-behavior dependent tests
GHashTable iteration order changed in recent GLib,
and tests were relying on that.

https://mail.gnome.org/archives/desktop-devel-list/2018-October/msg00016.html
2018-10-20 12:32:44 +01:00
Havard Graff
53a45b1222 Initial commit of GstRtpFunnel
For funneling together rtp-streams into a single session.
Use-cases include multiplexing and bundle.
2018-10-15 14:20:58 +02:00
Yeongjin Jeong
afa4be4b3b tests: flvdemux: Add new test for channel detect using aac codec-data
https://bugzilla.gnome.org/show_bug.cgi?id=797275
2018-10-12 14:35:37 -04:00
Yeongjin Jeong
7b5f7249e8 tests: flvmux: Add new test for caps change after starting to write headers
https://bugzilla.gnome.org/show_bug.cgi?id=797256
2018-10-11 15:35:24 -04:00
Havard Graff
6c05180dc5 rtpmux: respect downstream "timestamp-offset" in caps.
https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:39:02 -04:00
Havard Graff
6f37bd8f19 rtpmux: cleanup ssrc-handling code a bit
And add some better logging.

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:38:57 -04:00
Havard Graff
7cd36d2914 rtpmux: property should overrule both upstream and downstream
https://bugzilla.gnome.org/show_bug.cgi?id=762213

https://bugzilla.gnome.org/show_bug.cgi?id=795162
2018-10-10 15:35:31 -04:00
Havard Graff
ac6e77acad rtpsession: Don't start the RTCP thread until it's needed
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
2018-07-12 18:37:33 +02:00
Seungha Yang
aecc17251d tests: qtdemux: Add checking exposed segment event
https://bugzilla.gnome.org/show_bug.cgi?id=796480
2018-06-06 11:19:25 -04:00
Thiago Santos
0de143fa3e tests: qtdemux: Avoid using data beyond array and improve error msg
Makes it easier to debug the failures as well as prevents problems
reading out of bounds data.
2018-05-28 11:25:13 -07:00
Tim-Philipp Müller
48dd93662d tests: rtpstorage: fix potential crashes / test failures on 32-bit
Pass 64 bits to g_object_set() for 64-bit integer properties like
rtpstorage's "size-time" property.

https://bugzilla.gnome.org/show_bug.cgi?id=796429
2018-05-27 20:30:46 +01:00
Vivia Nikolaidou
d11339d616 splitmuxsink: Added new async-finalize mode
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
2018-05-24 12:47:24 +03:00
Havard Graff
77f3ce2e45 rtpsession: Add tests for PLI and FIR
https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-15 11:52:45 +01:00
Stian Selnes
457fdf95c4 rtpsession: Drop packet if trying to send from non-internal source
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
2018-05-15 10:34:29 +01:00
Havard Graff
b43ee8f5b1 rtpsession: Try media_ssrc if no src can be found for PLI sender_ssrc
Some RTP stacks out there does not set the sender_ssrc. In order to be
more robust, try to lookup the media_ssrc before dropping the PLI.

https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 20:41:39 +01:00
Mikhail Fludkov
386ca1d378 rtpsession: Fix on-feedback-rtcp race
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
2018-05-13 20:33:56 +01:00
John-Mark Bell
0a2b55ac3c rtpsession: do not emit RBs for internal senders.
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
2018-05-13 19:16:59 +01:00
Havard Graff
cd8c12f240 tests: rtpsession: fix indentation
https://bugzilla.gnome.org/show_bug.cgi?id=795139
2018-05-13 19:09:29 +01:00
Seungha Yang
3f090be2d1 tests: qtdemux: Add test for stream change
Add test case to verify track-id change and stream change

https://bugzilla.gnome.org/show_bug.cgi?id=684790
2018-05-10 08:09:20 +02:00
Olivier Crête
168fae813b flvmux: Wait for caps from both srcs before writing header
Wait for caps on all pads to start writing data even when source is live.

Includes unit test by Havard Graff that simulates it.

https://bugzilla.gnome.org/show_bug.cgi?id=794722
2018-04-26 15:41:54 -04:00
Mathieu Duponchelle
90f5ae8f45 ulpfecdec: output perfect seqnums
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
2018-04-19 18:17:39 +02:00
Mathieu Duponchelle
9b1aec0f79 flvmux test: refactor looped test.
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.
2018-04-13 23:02:26 +02:00
Mathieu Duponchelle
ec3c49e958 souphttpsrc test: free g_get_current_dir return 2018-04-13 20:35:24 +02:00
Mathieu Duponchelle
cc9fe814d6 rtpulpfec tests: Fix leaks 2018-04-13 17:37:47 +02:00
Sebastian Dröge
ed2ccb1a60 rtp: Fix compilation with non-C99 compilers
By moving variable declarations out of loop headers.
2018-03-20 12:08:28 +02:00
Olivier Crête
96261ce220 flvmux: Duration & unit tests
The muxed buffers will not carry the duration of the
incoming buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=793457
2018-03-01 18:25:02 -05:00
Mathieu Duponchelle
b0dd092ea6 tests: fix redenc tests
The default of the allow-no-red-blocks property was changed in a
previous commit, thus breaking the test assumptions
2018-02-27 16:34:51 +01:00
Tim-Philipp Müller
7f6aa7c344 .gitignore more test binaries 2018-02-22 10:54:02 +00:00
Mikhail Fludkov
d5ad50bd61 rtp: Implement ULPFEC (RFC 5109)
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
2018-02-21 14:15:22 +01:00
Jan Alexander Steffens (heftig)
54f312644e tests: aacparser: Test that short raw frames don't get concatenated
https://bugzilla.gnome.org/show_bug.cgi?id=792644
2018-01-18 19:09:25 +00:00
Mathieu Duponchelle
03dc22951b rtpbin: fix leak of elements requested by signals
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
2018-01-18 15:26:43 +01:00
Sebastian Rasmussen
0d57709d38 tests: udpsink: add check that sets QoS on IPv4/6 sockets
https://bugzilla.gnome.org/show_bug.cgi?id=757449
2017-12-23 12:45:11 +01:00
fengalin
3464aac3c9 matroska: fix memory leaks due to toc related updates
https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
Sebastian Dröge
c55824e4fa matroskamux: Fix various memory leaks in the unit test
https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
fengalin
694c07fe63 matroska-mux: migrate test to gst_harness
... following the guide lines from Håvard Graff (see https://gstconf.ubicast.tv/videos/moar-better-tests/).

https://bugzilla.gnome.org/show_bug.cgi?id=790686
2017-12-15 16:14:43 +02:00
fengalin
a6702a76d5 matroska: re-activate and update TOC support
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
2017-12-15 16:14:43 +02:00
Tim-Philipp Müller
c6b686624a tests: ignore rtph264 test binary 2017-12-09 16:15:24 +00:00
George Kiagiadakis
33bddfe321 tests: udpsrc: verify the correct amount of bytes is sent to the socket
https://bugzilla.gnome.org/show_bug.cgi?id=786799
2017-12-09 16:08:49 +00:00
George Kiagiadakis
ea7d2a0257 tests: udpsrc: ensure test won't timeout if the buffers are already received
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
2017-12-09 16:08:38 +00:00
George Kiagiadakis
45c82ee798 tests: udpsrc: fix test_udpsrc to actually run and fix locking
Previously this would silently be skipped because 1600 != 1400
and there is no assertion on this call.

Also unlock check_mutex after use.

https://bugzilla.gnome.org/show_bug.cgi?id=786799
2017-12-09 16:05:28 +00:00
Haakon Sporsheim
3c0d006c03 rtpsession: Handle zero length feedback packets
https://bugzilla.gnome.org/show_bug.cgi?id=791074
2017-12-02 13:58:34 +00:00
Havard Graff
96d837b301 tests: rtpsession: refactor tests to use GstHarness
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
2017-12-02 13:05:01 +00:00
Jan Schmidt
76e458a119 splitmuxsink: Use muxer reserved space properties if present.
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.
2017-11-25 00:56:11 +11:00
Jan Schmidt
3a813a0dcc splitmux: Fix file switch-on-caps-change.
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.
2017-11-24 16:56:03 +11:00
Tim-Philipp Müller
bca8ac2cf0 tests: rtp-payloading: add unit test for rtph264pay codec_data
Make sure no trailing zero bytes sneak into our SPS or PPS.

https://bugzilla.gnome.org/show_bug.cgi?id=732758
2017-11-23 09:36:15 +01:00
Tim-Philipp Müller
a9e57f3608 tests: rtph264depay: add test for using downstream memory allocator 2017-11-23 09:36:00 +01:00
Tim-Philipp Müller
5547901a37 mpg123: hook up to build system
https://bugzilla.gnome.org/show_bug.cgi?id=774252
2017-08-20 15:50:22 +01:00
Tim-Philipp Müller
4b1f43ebe3 Moving mpg123 plugin from -ugly 2017-08-20 13:48:48 +01:00
Sebastian Dröge
7e718d6039 Revert "matroskamux: adjust unit test to modified behaviour"
This reverts commit 8fe478c8a7.

We're back to previous behaviour
2017-07-18 10:08:33 +03:00
Olivier Crête
96e71b0286 rtpsession: Send EOS if all internal sources sent bye
The ones which are not internal should not matter, and we should
wait for all sources to have sent their BYEs.

And add unit test

https://bugzilla.gnome.org/show_bug.cgi?id=773218
2017-07-04 21:14:10 -04:00
Jan Alexander Steffens (heftig)
aa8ac28d86 tests: souphttpsrc: Avoid deprecated ssl-ca-file property
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
2017-06-29 15:32:30 -04:00
Jan Alexander Steffens (heftig)
9922091f1b tests: souphttpsrc: Avoid deprecated server ssl properties
The ssl-cert-file and ssl-key-file properties are deprecated. Use the
soup_server_set_ssl_cert_file function to load the files.

https://bugzilla.gnome.org/show_bug.cgi?id=784005
2017-06-29 15:32:30 -04:00
Jan Alexander Steffens (heftig)
27a0ea8cf5 tests: souphttpsrc: Make ssl_cert/key_file static
Just a bit of cleanup.

https://bugzilla.gnome.org/show_bug.cgi?id=784005
2017-06-29 15:32:30 -04:00
Tim-Philipp Müller
dd23afb6d4 sys: remove sunaudio plugin
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.
2017-06-23 20:02:43 +01:00
Tim-Philipp Müller
c35292505b meson: add options to set package name and origin
https://bugzilla.gnome.org/show_bug.cgi?id=782172
2017-05-20 14:53:42 +01:00
Tim-Philipp Müller
4df3669c0c tests: rtp-payloading: add test for rtph264depay avc/byte-stream output
Make sure avc output doesn't contain SPS/PPS inline, but
byte-stream output does.
2017-04-24 17:31:04 +01:00
Edward Hervey
7e9b7658e5 tests: Add vp9enc to gitignore 2017-04-12 11:33:05 +02:00
George Kiagiadakis
21f532f1c6 tests/check/rtprtx: add checks for rtprtxqueue's max-size-{time,packets} properties
https://bugzilla.gnome.org/show_bug.cgi?id=780867
2017-04-11 09:44:33 +03:00
Vincent Penquerc'h
d7212dac2e tests: fix leak in splitmux test
https://bugzilla.gnome.org/show_bug.cgi?id=781025
2017-04-09 11:19:56 +03:00
Jan Schmidt
57939fd98a splitmux test: Use passed first/last timestamps
Don't hard-code the expected timestamp range, use the
values the caller is passing in.
2017-03-14 15:48:08 +11:00
Nicolas Dufresne
27303b5904 tests: Add missing LDADD for libm in tests using math.h
Also, remove the math.h include for the one that just prentend to need
it.
2017-03-08 22:55:09 -05:00
Jan Schmidt
4335c4c160 splitmux: Add unit test for reverse playback
Ensure that reverse playback works and generates the range
of timestamps (0-3s) we expect, in monotonically descending order.
2017-03-04 00:35:32 +11:00
Sebastian Dröge
eefcdc9ee1 rtp-payloading: Add new test for Vorbis renegotiation
Check if encoding, payloading, depayloading and decoding works if the
stream configuration (and thus the headers) change.
2017-02-27 19:25:35 +02:00
George Kiagiadakis
e6bd2a5c18 tests: splitmux: add unit test for content with sparse streams
https://bugzilla.gnome.org/show_bug.cgi?id=761086
2017-02-27 12:58:21 +02:00
Guillaume Desmottes
0f719af307 tests: matroskamux, qtmux: don't add codec_data buffers to template caps
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
2017-02-21 15:47:16 +00:00
Søren Juul
1184429e21 icydemux: reset tags on empty value
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
2017-02-14 12:24:13 +02:00
Tim-Philipp Müller
781b5ac781 tests: rtpjitterbuffer: fix compiler warning due to c99-ism
rtpjitterbuffer.c:592:3: error: ‘for’ loop initial declarations are only allowed in C99 mode
2017-01-09 19:04:04 +00:00
Jan Schmidt
f7009eb5d7 splitmuxsink: Add format-location-full signal
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.
2017-01-03 01:34:02 +11:00
Edward Hervey
3a4d4dcd27 check: Remove dead code 2017-01-02 15:06:33 +01:00
Nicola Murino
8fe478c8a7 matroskamux: adjust unit test to modified behaviour
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
2016-12-21 16:58:42 +00:00
Havard Graff
0a81f71df5 tests/jitterbuffer: Major refactoring and cleanups
* 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
2016-12-14 15:00:37 +02:00
Sebastian Dröge
63938ef730 gst: Don't declare variables inside the for loop header
This is a C99 feature.
2016-12-13 22:32:46 +02:00
Philippe Normand
dcd3ce9751 rtpbin: receive bundle support
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
2016-11-16 08:56:34 +01:00
Havard Graff
1a4393fb4d rtpjitterbuffer: fix timer-reuse bug
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.com

https://bugzilla.gnome.org/show_bug.cgi?id=773891
2016-11-04 16:56:56 +02:00
Havard Graff
fb9c75db36 rtpjitterbuffer: fix lost-event using dts instead of pts
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).
2016-11-04 16:51:20 +02:00
Havard Graff
bea35f97c8 rtpjitterbuffer: fix bug in reschedule_timer
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
2016-11-04 16:40:14 +02:00
Tim-Philipp Müller
752dd15c54 tests: wavparse: add test for processing an actual .wav file
https://bugzilla.gnome.org/show_bug.cgi?id=773861
2016-11-03 15:42:29 +02:00
Havard Graff
78ab8cbdcd rtph263ppay: Fix caps leak
Fix leaking caps when downstream has not-fixed caps.

https://bugzilla.gnome.org/show_bug.cgi?id=773515
2016-11-01 20:20:47 +02:00
Tim-Philipp Müller
834339b773 tests: videomixer: disable racy flush_start_flush_stop test
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
2016-10-20 22:08:14 +01:00
Jan Alexander Steffens (heftig)
6deab72e10 tests: Fix souphttpsrc tests without CK_FORK=no
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
2016-10-20 13:29:07 +03:00
Jan Alexander Steffens (heftig)
22ced681af tests: Fix level test in CK_FORK=no mode
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
2016-10-20 13:23:30 +03:00
Tim-Philipp Müller
e6d188967a tests: fix indentation 2016-09-15 09:53:07 +01:00
Havard Graff
f440b074b1 rtpjitterbuffer: improved rtx-rtt averaging
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
2016-09-14 19:37:50 -04:00
Stian Selnes
f8238f0a9f rtpjitterbuffer: Detect whether to assume equidistant spacing when loss
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
2016-09-14 19:37:50 -04:00
Stian Selnes
2eb7383816 rtpjitterbuffer: Don't request rtx if 'now' is past retry period
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
2016-09-14 19:37:50 -04:00
Stian Selnes
ab49dfd0b2 rtpjitterbuffer: Fix lost duration when gap after lost timer
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
2016-09-14 19:37:50 -04:00
Havard Graff
8087a8a31c rtpjitterbuffer: Improved expected-timer handling when gap > 0
https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Stian Selnes
38a7545003 rtpjitterbuffer: Major improvements for RTX stats
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
2016-09-14 19:37:50 -04:00
Havard Graff
1b868cc9b1 rtpjitterbuffer: Add and expose more stats and increase testing of it
Add num-pushed and num-lost.
Expose num-late, num-duplicates and avg-jitter.

https://bugzilla.gnome.org/show_bug.cgi?id=769768
2016-09-14 19:37:50 -04:00
Josep Torra
d40f007d61 gitignore: ignore qtdemux, rtph261 and rtpvp9 tests 2016-08-26 21:32:07 +02:00
Josep Torra
ccc7d7e5a3 tests: remove a wrong 'const' specifier
Fixes "error: duplicate 'const' declaration specifier"
2016-08-26 21:14:47 +02:00
Stian Selnes
8bf77e34f2 rtpvp9depay: Support flexible mode 2016-08-26 11:57:15 -04:00