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
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.
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
Move the conversion code used in videoconvert to the video library
and expose a simple but generic API to do arbitrary conversion. It can
currently do colorspace conversion but the plan is to add videoscale to
it as well.
See https://bugzilla.gnome.org/show_bug.cgi?id=732415
Currently the API is far from optimal and the user has to work around
our badly defined API to simply install missing plugins.
API:
new:
gst_discoverer_info_get_missing_elements_installer_details
deprecated:
gst_discoverer_info_get_misc
gst_discoverer_stream_info_get_misc
https://bugzilla.gnome.org/show_bug.cgi?id=720596
allows configuration of whether GstVideoGLTextureUploadMeta is
added to buffers resulting from a buffer pool. This is sperate
to the caps feature in that an element may want to add the upload
meta itself rather than allowing the buffer pool to.
https://bugzilla.gnome.org/show_bug.cgi?id=712798
It was possible to decide only what #GstElement implementing #GstPreset
to use during the encoding, we can now let the user select a specific preset previously
saved using #gst_preset_save_preset specifying the name chosen when it was saved
in the gst_encoding_profile_set_preset_name.
Actually loading a preset with %NULL as a name would have always failed, so
in the current state of the API that feature is unusable
API:
gst_encoding_profile_set_preset_name
gst_encoding_profile_get_preset_name
And only return the proportion. The earliest time already can be
retrieved from get_max_decode_time() and by renaming we allow this
to be more extensible in the future.
Add a getter for the QoS proportion and earliest_time to help
subclasses do better estimations based on the proportion.
API: gst_video_decoder_get_qos_info()
https://bugzilla.gnome.org/show_bug.cgi?id=687991
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
Video base classes and theora plugin still needs to be ported again
Conflicts:
docs/libs/gst-plugins-base-libs-docs.sgml
docs/libs/gst-plugins-base-libs-sections.txt
docs/libs/gst-plugins-base-libs.types
ext/theora/gsttheoradec.c
ext/theora/gsttheoradec.h
ext/theora/gsttheoraenc.c
ext/theora/gsttheoraenc.h
gst-libs/gst/video/Makefile.am
gst-libs/gst/video/video.c
gst-libs/gst/video/video.h
gst/playback/gsturidecodebin.c
tests/check/libs/video.c
tests/check/pipelines/theoraenc.c
win32/common/libgstvideo.def
