This patch introduces the AVTP Audio Format (AAF) payloader element from
the AVTP plugin. The element inputs audio raw data and outputs AVTP
packets (aka AVTPDUs), implementing a typical protocol payloader element
from GStreamer.
AAF is one of the available formats to transport audio data in an AVTP
system. AAF is specified in IEEE 1722-2016 section 7 and provides two
encapsulation mode: PCM and AES3. This patch implements PCM
encapsulation mode only.
The AAF payloader working mechanism consists of building the AAF header,
prepending it to the GstBuffer received on the sink pad, and pushing the
buffer downstream. Payloader parameters such as stream ID, maximum
transit time, time uncertainty, and timestamping mode are passed via
element properties. AAF doesn't support all possible sample format and
sampling rate values so the sink pad caps template from the payloader is
a subset of audio/x-raw. Additionally, this patch implements only
"normal" timestamping mode from AAF. "Sparse" mode should be implemented
in future.
Upcoming patches will introduce other AVTP payloader elements that will
have some common code. For that reason, this patch introduces the
GstAvtpBasePayload abstract class that implements common payloader
functionalities, and the GstAvtpAafPay class that extends the
GstAvtpBasePayload class, implementing AAF-specific functionalities.
The AAF payloader element is most likely to be used with the AVTP sink
element (to be introduced by a later patch) but it could also be used
with UDP sink element to implement AVTP over UDP as described in IEEE
1722-2016 Annex J.
This element was inspired by RTP payloader elements.
This patch introduces the bootstrap code from the AVTP plugin (plugin
definition and init) as well as the build system files. Upcoming patches
will introduce payloaders, source and sink elements provided by the AVTP
plugin. These elements can be utilized by a GStreamer pipeline to
implement TSN audio/video applications.
Regarding the plugin build system files, both autotools and meson files
are introduced. The AVTP plugin is landed in ext/ since it has an
external dependency on libavtp, an opensource AVTP packetization
library. For further information about libavtp check [1].
[1] https://github.com/AVnu/libavtp
The agent itself will take a ref on the property setter, so we'll be
left with two references to the certificate object, when actually there
should be only one
When WPEBackend-fdo >= 1.3.0 is detected, the threaded view now relies on the
wpe_fdo_egl_exported_image API instead of the EGLImageKHR-based API which is
going to be deprecated in 2.26. The GLib sources created by the view now use the
default priority as well, the custom priority is no longer required.
Regression introduced by b4bdcf15b7237eb5c5943c4f746701a477333000
This commit prevents the handshake from reaching dtlsdec when
the receive state of the receive bin is set to DROP (for example
when transceivers are sendonly).
This preserves the intent of the commit, by blocking the bin
at its sinks until the receive state is no longer BLOCK, but
makes sure the handshake still goes through, by only dropping
data at the src pads, as was the case before.
The timestamp/PTS alone is meaningless without the segment and usually
applications care about the running-time or stream-time.
This also keeps the messages in sync with the spectrum and level
elements.
Header data must be forwarded to downstream, but if demux does not finish
to finding type (e.g., ts, mp4 and etc), this header data can be cleared
by _stream_clear_pending_data(). Moreover, although demux finish downloading
header data, still it has fragment date to be downloaded, fragment sequence
shouldn't be advanced yet at that moment.
https://bugzilla.gnome.org/show_bug.cgi?id=776928
Including gl.h from WPEThreadedView.h leads to GST_LEVEL_DEFAULT detected as
redefined. The proposed fix is to include config.h from the CPP implementation
file and disable gl.h inclusion in the header, by using forward declarations.
1. The spec indicates that the notification should occur near the end of
'setting the description' processing
2. The current location with the drop of the lock could cause the 'check
if negotiation is needed' logic to execute and become confused about
the state of the webrtcbin's current local descriptions.
In the bad case, the following assertions could be hit:
g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_local_description->sdp));
g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_remote_description->sdp));
Moving the signalling state change later in the set description task
means that checking for a renegotiation will early abort as the
signalling state is not STABLE before the session description and
transceivers have been updated.
The start of each segment is relative to the Period start, minus
the presentation time offset.
As specified in section 5.3.9.6 of the MPEG DASH specification:
The value of the @t attribute minus the value of the
@presentationTimeOffset specifies the MPD start time of
the first Segment in the series.
dashdemux was not taking account of presentationTimeOffset and in
some methods was not taking into account the Period start time.
This commit modifies the segment->start value to always be
relative to the MPD start time (zero for VOD,
availabilityStartTime for live streams). This makes all uses of
the segment list consistent.
Fixes#841
