GitLab

We need to mark the codec_data as a keyframe or else downstream decoders might
decide to skip it, waiting for a keyframe.

Fixes #631996

ie. "foo_bar" -> "foo-bar"

Incomming buffer is only pushed on the adapter at the end of the
handle_buffer function. But duration/timestamp of this buffer is already
taken into account for the current data in the adapter. This leads to
wrong rtp timestamps and extra latency.

Mark the rtspsrc element as a source.
Requires 0.10.31.1 now

it's perfectly ok for a video output device to not have overlay capabilities.
this patch removes the need to get/set the overlay parameters if the user
does not explicitely request one of the overlay properties

gst_v4l2sink_change_state() would free the pool without checking whether there
was a valid pool...

From c4a8adc to 5a668bf

From 5e3c9bf to c4a8adc

Both history_count and fields_required count from 1. As per the while loop
condition that follows this code, to perform the deinterlacing method, we need
history_count >= fields_required fields in the history. Therefore if we have
history_count < fields_required (not fields_required + 1), we need more fields.

This fixes the assumption that DecoderSpecificInfo must be 2 bytes long
for AAC files. The specification allows HE-AAC to be explicitly
signalled in a backward compatible way. This is done by means of an
additional information after the regular AAC header. It is expected that
decoders that can play AAC but not HE-AAC will parse the header normally
and ignore extended bits, much as they do for the HE-AAC specific payload
in the actual stream.

https://bugzilla.gnome.org/show_bug.cgi?id=612313

Fixes #629047.

Take the duration of the frames into account so that we don't drop frames that
are only partially past the QoS deadline.

Use the adapter timestamp code to get more accurate timestamps.
Fix latency calculation, we add our own latency in the worst case.

Fixes #629896

https://bugzilla.gnome.org/show_bug.cgi?id=630205

Fixes #631303

Add some defines for the default width/height/fps instead of using different
values in different places.

Implement a latency query and report how much latency we will add to the
stream.
Alse make some defaults for the default width/height/framerate

Fixes #631303

Add a python version of the audio sender pipeline.

Ported by Sp4rc on IRC.

Add a python version of the PCMA client app.

Ported by Sp4rc on IRC.

These warnings are wrong though, the variables are only used in
the cases where they *are* initialized by the bit reader.

The clock rate is always 8000 Hz according to the RFC and
the sampling rate must always be 16000 Hz.

https://bugzilla.gnome.org/show_bug.cgi?id=631082

The functions used to extract this data have been moved to gstpbutils to
facilitate reuse.

https://bugzilla.gnome.org/show_bug.cgi?id=617318

The functions used to extract this data have been moved to gstpbutils to
facilitate reuse.

https://bugzilla.gnome.org/show_bug.cgi?id=617318

This uses gstpbutils to extract the profile and level from the video
object sequence and adds this to stream caps. This can be used as
metadata and for fine-grained decoder selection.

https://bugzilla.gnome.org/show_bug.cgi?id=616521

This exports the AAC profile and level in caps for use as metadata and
(eventually) for more fine-grained selection of decoders at
caps-negotiation time. (Doesn't work for HE-AAC yet though.)

https://bugzilla.gnome.org/show_bug.cgi?id=612313