This is the same code that is in decklinkaudiosrc, audioringbuffer,
audiomixer and various other places. Have it once instead of copying it
everywhere.
https://bugzilla.gnome.org/show_bug.cgi?id=787560
+ Refactor previous constructor to call on that new constructor
+ Reimplement is_passthrough to strictly check whether the matrix
is an identity matrix, comparing channel-masks was incorrect:
the mixer may be remixing from a list of positions to the same
list of positions, but ordered differently, and reciprocally,
the mixer may be remixing from a list of positions to another
list of positions identically ordered
+ Remove unused tmp field, must have been a refactoring leftover
https://bugzilla.gnome.org/show_bug.cgi?id=785471
It is often usefull to make sure that you get a full copy of a profile.
For example you want to let the user modify it in the user interface
but still keep an unchanged version for later use.
API:
gst_encoding_profile_copy
It adds a third argument to pass GstBufferPoolAcquireParams
to gst_buffer_pool_acquire_buffer.
If a user subclasses GstBufferPoolAcquireParams, this allows to
pass an updated param to the underlying buffer pool at each
gst_video_decoder_allocate_output_frame_with_params call.
https://bugzilla.gnome.org/show_bug.cgi?id=773165
Adds "memory:DMABuf" caps feature. Since 1.11 tag.
Useful when the the dma-buf buffer cannot be mapped to CPU for r/w requests.
Example: protected content or platform constraints.
https://bugzilla.gnome.org/show_bug.cgi?id=759358
Usually this information is static for the whole stream, and various
container formats store this information inside the headers for the
whole stream.
Having it inside the caps for these cases simplifies code and makes it
possible to express these requirements more explicitly with the caps.
https://bugzilla.gnome.org/show_bug.cgi?id=771376
The _pull_sample() and _pull_preroll() functions block
until a sample is available, EOS happens or the pipeline
is shut down (returning NULL in the last two cases).
This adds _try_pull_sample() and _try_pull_preroll()
functions with a timeout argument to specify the maximum
amount of time to wait for a new sample.
To avoid code duplication, wait forever if the timeout is
GST_CLOCK_TIME_NONE and use that to implement
_pull_sample/_pull_preroll with the original behavior.
Add also corresponding action signals "try-pull-sample"
and "try-pull-preroll".
https://bugzilla.gnome.org/show_bug.cgi?id=768852
We currently don't log much about channel positions making debugging
harder as it should be. This is the first step in my attempt to improve
this.
https://bugzilla.gnome.org/show_bug.cgi?id=763985
gst_rtsp_connection_connect_with_response accepts a response pointer
which it fills with the response from setup_tunneling if the
connection is configured to be tunneled. The motivation for this is to
allow the caller to inspect the response header to determine if
additional authentication is required so that the connection can be
retried with the appropriate authentication headers.
The function prototype of gst_rtsp_connection_connect has been
preserved for compatability with existing code and wraps
gst_rtsp_connection_connect_with_response.
https://bugzilla.gnome.org/show_bug.cgi?id=749596
Have all sections in alphabetical order. Also make the macro order consistent.
This is a preparation for generating the file. Remove GET_CLASS macro for
some elements, since it is not used and the header is not installed.
Add flags and enums to support multiview signalling in
GstVideoInfo and GstVideoFrame, and the caps serialisation and
deserialisation.
videoencoder: Copy multiview settings from reference input state
Add gst_video_multiview_* support API and GstVideoMultiviewMeta meta
https://bugzilla.gnome.org/show_bug.cgi?id=611157
Summary:
So that the user can easily use the same encoding profile to render
with/without audio/video stream.
API:
gst_encoding_profile_is_disabled
gst_encoding_pofile_set_enabled
https://bugzilla.gnome.org/show_bug.cgi?id=749056
[API] gst_discoverer_info_to_variant
[API] gst_discoverer_info_from_variant
[API] GstDiscovererSerializeFlags
+ Serializes as a GVariant
+ Adds a test
+ Does not serialize potential GstToc (s)
https://bugzilla.gnome.org/show_bug.cgi?id=748814
All those where super straight forward from the warnings gtkdoc prints. It kind
of makes sense to apply them before the list of warnings is >100 and people
complain that gtkdoc is noisy.
`socketsrc` can be considered a source counterpart to `multisocketsink`.
It can be considered a generalization of `tcpclientsrc` and
`tcpserversrc`: it contains all the logic required to communicate over
the socket but none of the logic for creating the sockets/establishing
the connection in the first place, allowing the user to accomplish this
externally in whatever manner they wish making it applicable to other
types of sockets besides TCP.
This commit essentially copies the implementation directly from
tcpserversrc. Later patches will tidy the implementation up and
re-implement `tcpclientsrc` and `tcpserversrc` in terms of `socketsrc`.
See https://bugzilla.gnome.org/show_bug.cgi?id=739546
Add API to add and get custom headers that are not
covered by our header fields enum. This is backwards
compatible in that it will also work for our defined
fields, so if we ever add a new header field to the
enum, get_header_by_name() for the same header string
will still work.
API: gst_rtsp_message_add_header_by_name()
API: gst_rtsp_message_take_header_by_name()
API: gst_rtsp_message_remove_header_by_name()
API: gst_rtsp_message_get_header_by_name()
In some cases, the user might want the stream outputted by encodebin to
be in the exact same format during all the stream. We should let the
user specify when this is the case. This commit add some API in the
GstEncodingProfile to determine whether the format can be renegotiated
after the encoding started or not.
API:
gst_encoding_profile_set_allow_dynamic_output
gst_encoding_profile_get_allow_dynamic_output
https://bugzilla.gnome.org/show_bug.cgi?id=740214
