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.
