The old API would only assert or return an invalid timecode, the new API
returns a boolean or NULL. We can't change the existing API
unfortunately but can at least deprecate it.
audioconvert's passthrough status can no longer be determined
strictly from input / output caps equality, as a mix-matrix can
now be specified.
We now call gst_base_transform_set_passthrough dynamically, based
on the return from the new gst_audio_converter_is_passthrough()
API, which takes the mix matrix into account.
According to RFC3611, the extended report blocks in XR packet can
have variable length. To visit each block, the iterator should look
into block header. Once XR type is extracted, users can parse the
detailed information by given functions.
Loss/Duplicate RLE
The Loss RLE and the Duplicate RLE have same format so
they can share parsers. For unit test, randomly generated
pseudo packet is used.
Packet Receipt Times
The packet receipt times report block has a list of receipt
times which are in [begin_seq, end_seq).
Receiver Reference Time paser for XR packet
The receiver reference time has ntptime which is 64 bit type.
DLRR
The DLRR report block consists of sub-blocks which has ssrc, last RR,
and delay since last RR. The number of sub-blocks should be calculated
from block length.
Statistics Summary
The Statistics Summary report block provides fixed length
information.
VoIP Metrics
VoIP Metrics consists of several metrics even though they are in
a report block. Data retrieving functions are added per metrics.
https://bugzilla.gnome.org/show_bug.cgi?id=789822
This patch adds API in the audio decoder base class for setting the arbitrary
caps on the source pad. Previously only caps converted from audio info were
possible. This is particularly useful when subclass wants to set caps features
for audio decoder producing metadata.
Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.
A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.
RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=761947
Helper function for bindings, in python for example
users can now replace:
res, msg = GstSdp.SDPMessage.new()
GstSdp.sdp_message_parse_buffer(bytes(text.encode()), msg)
with:
res, msg = GstSdp.SDPMessage.new_from_text(text)
https://bugzilla.gnome.org/show_bug.cgi?id=796563
If we can guarantee the lifetime of the fd is longer than
the memory, we can use DONT_CLOSE flag not to close when release.
But it's not provided in gstdmabuf yet while gstfdmemory does.
For example, in case of using VA-API or MSDK, we would need this api.
Otherwise we should call dup to duplicate the fd.
https://bugzilla.gnome.org/show_bug.cgi?id=794829
This commits add common elements for Ancillary Data and Closed
Caption support in GStreamer:
* A VBI (Video Blanking Interval) parser that supports detection
and extraction of Ancillary data according to the SMPTE S291M
specification. Currently supports the v210 and UYVY video
formats.
* A new GstMeta for Closed Caption : GstVideoCaptionMeta. This
supports the two types of CC : CEA-608 and CEA-708, along with
the 4 different ways they can be transported (other systems
are super-set of those).
https://bugzilla.gnome.org/show_bug.cgi?id=794901
* Explicitely specify which headers aren't to be included in gtkdoc-scan
This is essentially all the headers that are not installed and only
for internal/local usage. This also includes the orc-generated headers.
* Remove all symbols/sections that are no longer present (due to accurately
scanning only the headers we need).
* Add or expose sections which weren't previously exposed
* Make sure the "unified" library headers (ex: gst/video/video.h) are used
everywhere applicable. Only use the specific headers where applicable
(such as the GL-implementation-specific objects)
* Add all documentation which was not previously exposed in the right sections
* Update 'types' file to get as many runtime information as possible
This brings down the number of unused symbols to 15 (from over 300).
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
