This patch introduces the avtpsink elements which implements a typical
network sink. Implementation is pretty straightforward since the burden
is implemented by GstBaseSink class.
The avtpsink element defines three new properties: 1) network interface
from where AVTPDU should be transmitted, 2) destination MAC address
(usually a multicast address), and 3) socket priority (SO_PRIORITY).
Socket setup and teardown are done in start/stop virtual methods while
AVTPDU transmission is carried out by render(). AVTPDUs are encapsulated
into Ethernet frames and transmitted to the network via AF_PACKET socket
domain. Linux requires CAP_NET_RAW capability in order to open an
AF_PACKET socket so the application that utilize this element must have
it. For further info about AF_PACKET socket domain see packet(7).
Finally, AVTPDUs are expected to be transmitted at specific times -
according to the GstBuffer presentation timestamp - so the 'sync'
property from GstBaseSink is set to TRUE by default.
This patch introduces the AAF depayloader element, the counterpart from
the AAF payloader. As expected, this element inputs AVTPDUs and outputs
audio raw data and supports AAF PCM encapsulation only.
The AAF depayloader srcpad produces a fixed format that is encoded
within the AVTPDU. Once the first AVTPDU is received by the element, the
audio features e.g. sample format, rate, number of channels, are decoded
and the srcpad caps are set accordingly. Also, at this point, the
element pushes a SEGMENT event downstream defining the segment according
to the AVTP presentation time.
All AVTP depayloaders will share some common code. For that reason, this
patch introduces the GstAvtpBaseDepayload abstract class that implements
common depayloader functionalities. AAF-specific functionalities are
implemented in the derived class GstAvtpAafDepay.
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
Fix a recently introduced segfault. Don't de-reference a NULL
SPS pointer when attempting to update source caps before SPS
has been seen in the stream.
Interleaved frames can be fragmented between
incoming frames. Thus, we can have multiple
frames within the single input frame, as well as
incomplete frame. Now it preserves parsing
state and handle both situations.
Fixes#991
msdkenc supports CSC implicitly, so it is possible that two VPP
processes are required when a pipeline contains msdkvpp and msdkenc.
Before this fix, msdkvpp and msdkenc may share the same context, hence
the same mfx session, which results in MFX_ERR_UNDEFINED_BEHAVIOR
in MSDK because a mfx session has at most one VPP process only
This fixes the broken pipelines below:
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420 ! msdkh264enc ! \
msdkh264dec ! msdkvpp ! video/x-raw,format=YUY2 ! fakesink
gst-launch-1.0 videotestsrc ! msdkvpp ! video/x-raw,format=YUY2 ! \
msdkh264enc ! fakesink
MSDK supports JPEG YUY2 (422 chroma) output color
format. The color format of input bitstream is
described by JPEGChromaFormat and JPEGColorFormat
fields in the mfxInfoMFX structure which is filled
in by the MFXVideoDECODE_DecodeHeader function.
To obtain lossless decoded output from 422 encoded
JPEGs, we must set the output color format in the
FourCC and ChromaFormat fields in the mfxFrameInfo
structure to the appropriate values at post_configure
so that they are propagated through to the srcpad
caps accordingly.
A post_configure virtual method is added to allow
codec subclasses to adjust the initialized parameters
after MFXVideoDECODE_DecodeHeader is called from the
gstmsdkdec::gst_msdkdec_handle_frame function.
This is useful if codecs want to adjust the output
parameters based on the codec-specific decoding
options that are present in the mfxInfoMFX structure
after MFXVideoDECODE_DecodeHeader initializes them.
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.
The length of the TCP payload is the IP plus TCP header length
subtracted from the IP datagram length specified in the IP header.
Prior to this, the size was calculated incorrectly, considering
all data after TCP header as a payload till the end of a packet.
Fixes#995
If recording is set to FALSE after the last audio or video buffer and
before the EOS event then recording stop is never signalled.
Similarly, we should signal recording stop once both audio and video are
EOS, regardless of the recording property, as there's nothing to be
recorded anymore.
Regression introduced by b4bdcf15b7
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.
