Having an unlimited input queue is very bad if the
encoder can't run at real-time. Eventually it will
consume all RAM. I don't really see any reason to
have more than 1 outstanding encoded frame, so
remove the queue and limit things to 1 pending frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2499>
There was a race when going to PAUSED while pushing a buffer to the
pipeline process (where we weren't even cancelling anything).
This rework base all the cancellation around the GCancellable
"cancelled" signal trying to ensure that the streaming thread will not
block once a cancel operation happens.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2504>
Commit bc09d8cc changed gstmpdparser to put the entire
<ContentProtection> element in the "value" field, so that DRMs
other than PlayReady could make use of the data inside this
element.
However, the data in the "value" field does not include any
XML namespace declarations that are used within the element. This
causes problems for a namespace aware XML parser that wants to
make use of this data.
This commit modifies the way the XML is converted to a string
so that XML namespaces are preserved in the output.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2487>
GstWebRTCSCTPTransport is now made into into an abstract base class
that only contains property specifications matching the
RTCSctpTransport interface of the W3C WebRTC specification, see
https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This
class is put into the WebRTC library to expose it for applications and
to allow for generation of bindings for non-dynamic languages using
GObject introspection.
The actual implementation is moved to the subclass WebRTCSCTPTransport
located in the WebRTC plugin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
Being able to access the SCTP Transport object from the application
means the application can access the associated DTLS Transport object
and its ICE Transport object. This means we can observe the ICE state
also for a data-channel-only session. The collated
ice-connection-state on webrtcbin only includes the ICE Transport
objects that resides on the RTP transceivers (which is exactly how it
is specified in
https://w3c.github.io/webrtc-pc/#rtciceconnectionstate-enum).
For the consent freshness functionality (RFC 7675) to work the ICE
state must be accessible and consequently the SCTP transport must be
accessible for enabling consent freshness checking for a
data-channel-only session.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
This encoder advertises alignment=au as output format, which means
each output frame should contain a full decodable access unit.
The video encoder base class is not aware of our output alignment
and will output spurious buffers with just the SPS/PPS inside when
we call gst_video_encoder_set_headers(), which is broken because
each buffer is supposed to contain a full decodable access unit
in our case.
Just don't tell the base class about our headers, they will be
sent at the beginning of each IDR frame anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2178>
The mapping between an RTP session and the SDP m= line is not always the
same, especially when BUNDLEing is used.
This causes a failure in a specific case where if when bundling,
if mline 0 is a data channel, and mline 1 an audio/video section,
then retrieving the transceiver at mline 0 (rtp session used) will fail
and cause an assertion.
This fix is actually potentially a regression for cases where the remote
part does not provide the a=ssrc: media level SDP attributes as is now
becoming common, especially when simulcast is involved.
The correct fix actually requires reading out header extensions as used
with bundle for signalling in the actual data, what media and therefore
transceiver is being used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2467>
When converting from one framerate to another, counters are
reset periodically, however when not converting they never are
and can_genearte_output ends up making overflow-prone calculations
with large values for input_frames and output_frames.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2465>
The GST_ELEMENT_ERROR will call the gst_object_get_path_string and
use gst_object_get_parent to get the full object path name, which
needs to lock the object. But we are already in a locked context and
so this will cause a deadlock, the pipeline can not exit normally.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2451>
Some manifests use the ContentProtection node to store additional information
such as the license server url. Our MPD parser used to process the
ContentProtection node, extracting Playready PSSH boxes. However for other DRM
systems, only the `value` attribute was passed down to the protection event, so
for example, Widevine data was not parsed at all and "Widevine" was passed to
the event, which is not very useful for decryptors that require a PSSH init
data.
Parsing should now be done by decryptors which will receive the entire
ContentProtection XML node as a string. This gives more "freedom" to the
decryptor which can then detect and parse custom nodes as well.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2400>
if there is no pad name, the representation id
was NULL, causing a crash when writing the mpd file.
gst-launch-1.0 videotestsrc num-buffers=900 ! video/x-raw, width=800,
height=600, framerate=30/1 ! x264enc ! video/x-h264, profile=high !
dashsink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2064>
We can only convert from non-cdp to cdp within the confines of valid cdp
framerates. The existing caps negotiation code was allowing any
framerate to convert to a cdp output which is incorrect and would hit an
assertion later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2372>
If they're not dropped, they can be blocked in the queue even if it is
leaky in the case where there is a buffer being pushed downstream. Since
in webrtc, it's unlikely that there will be a special allocator to
receive RTP packets, there is almost no downside to just ignoring the
queries.
Also drop queries if they get caught in the pad probe after the queue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2363>
By using the clocksync inside the dtlssrtpenc, all streams inside a
bundled are synchronized together. This will cause problems if their
buffers are not already arriving synchronized: clocksync would wait for
a buffer on one stream and then buffers from the other stream(s) with
lower timestamps would all be sent out too late.
Placing the clocksync before the rtpbin and rtpfunnel synchronizes each
stream individually and they will be send out more smoothly as a result.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2355>
