Make the nvenc OpenGL usage rely on the the same condition
that the automake USE_OPENGL conditional checks, as the
USE_OPENGL doesn't actually get set into the configure script,
so it can't check that
Since cuda-tools 9.0, nvcuvid.h is replaced by dynlink_nvcuvid.h.
This patch changes nvdec to use run-time dynamic linking if
cuda-tools version >= 9.
nvenc does not require any change since its necessary headers are
still available.
https://bugzilla.gnome.org/show_bug.cgi?id=791724
Measures the audio latency between the source pad and the sink pad by
outputting period ticks on the source pad and measuring how long they
take to arrive on the sink pad.
Very useful for quantifying latency improvements in audio pipelines.
This plugin was particularly useful during development of the
low-latency features of the wasapi plugin.
https://bugzilla.gnome.org/show_bug.cgi?id=793839
This keep-it-simple plugin is useful when you want to pipe arbitrary
data to a different pipeline within the same process. Some advantages
over appsink/appsrc, the inter elements, etc:
* Ease of use. Buffers, events, and caps are transmitted as-is without
copying or serialization.
* Enables zerocopy (especially DMABUF) transparently without any
special-casing.
* Enables usage with sinks or elements that are unreliable and may
throw errors and need re-initialization, such as a network sink, a
USB device sink (v4l2), etc.
* Transmits arbitrary data, not just audio/video/subs
* Can easily implement 1 producer pipeline -> N dynamic consumer
pipelines within a single process when combined with the `tee`
element.
All queries, events, buffers, and buffer lists are proxied. State
changes, clocks, and base times for the two pipelines are independent
since the upstream and downstreams continue to be different pipelines.
https://bugzilla.gnome.org/show_bug.cgi?id=788200
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/
The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer. In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.
The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.
With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792523
For libsrtp 1, add defines that translate the new namespaced identifiers
to the old unnamespaced ones. Also move the code for setting and getting
a stream's ROC into two compat functions that match libsrtp2's API.
It seems that libsrtp2 properly supports changing the ROC without having
to touch the sequence numbers afterwards, given that srtp_set_stream_roc
sets a pending_roc field, so the entire roc_changed dance should not be
needed anymore. The compat functions for libsrtp 1 just contain our
preexisting hacks, however, so it's still needed there.
libsrtp2 has no means of discovering the streams in the session, so to
create the stats structure we need to iterate over our own set of SSRCs.
For this we also need to re-add the previously removed ssrcs_set to the
encoder.
https://bugzilla.gnome.org/show_bug.cgi?id=776901
This plugin is useful when you want to pipe arbitrary data to
a different pipeline within the same process. Buffers, events, and caps
are transmitted as-is without copying or manipulation.
SRT[0] is an open source transport technology[1] that optimizes
streaming performance across unpredictable networks.
Although SRT is based on UDP, it works like connection-oriented
protocol. However, it doesn't mean that the SRT server or client
is necessarily to link to a receiver or a sender so, here, the
pairs of source and sink elements are introduced.
- srtserversink: SRT server to feed SRT stream
- srtclientsrc: SRT client to get SRT stream from srtserversink
- srtclientsink: SRT client to send SRT stream
- srtserversrc: SRT server to listen from srtclientsink
[0] https://github.com/Haivision/srt
[1] http://www.srtalliance.org/https://bugzilla.gnome.org/show_bug.cgi?id=785730
Add support for parsing linear time code from
an audio source using libltc
https://github.com/x42/libltc
The user can now choose between 3 different and independently
running timecode sources. The old override-existing property
has been replaced by timecode-source.
https://bugzilla.gnome.org/show_bug.cgi?id=784295
