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Update docs
Add another flag for the quantizer
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Add docs for the internal audioconvert object before moving it to the
audio library.
Remove get_sizes and implement the trivial logic in the element.
Remove some unused orc functions
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Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
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Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.
API: gst_audio_channel_get_fallback_mask()
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sample
In some conditions we might process empty buffers, calling
gst_control_binding_get_value_array in that case will lead
to the assertion:
(lt-ges-launch-1.0:18859): GStreamer-CRITICAL **: gst_control_binding_get_value_array: assertion 'values' failed
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Add a new method to get the default channel-mask.
Use the new method on audiodecoder and audioconvert.
API: gst_audio_channel_get_default_mask()
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And fix the test_overlay_blend test where we blend over a
transparent frame and where expecting wrong results
https://bugzilla.gnome.org/show_bug.cgi?id=681447
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Also support all premultiplied/non-premultiplied source/destination
configurations
https://bugzilla.gnome.org/show_bug.cgi?id=681447
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https://bugzilla.gnome.org/show_bug.cgi?id=757152
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https://bugzilla.gnome.org/show_bug.cgi?id=757152
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... instead of relying on the segment. For the clipping at the start we assume
a proper value in the OpusHead, as generated by opusparse or opusenc.
Transmuxing in general is not guaranteed to produce the correct values, or
even have a OpusHead (e.g. when having RTP input).
https://bugzilla.gnome.org/show_bug.cgi?id=757153
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https://bugzilla.gnome.org/show_bug.cgi?id=757153
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https://bugzilla.gnome.org/show_bug.cgi?id=757153
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The granulepos does not have the pre-skip subtracted while timestamps do,
and the last granulepos will be shorter by the number of samples that should
be dropped because of padding in the end.
As such, extrapolating the granule of the beginning of the first frame will
lead to a negative value, which is not a problem but intentional.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
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GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
handle negative values better.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
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Add a TRUNCATE_RANGE flag for unpack functions to fill the least
significate bits with 0 (as did the old code). Also add functions
that don't truncate. Use the TRUNC flag in audioconvert for
backwards compatibility for now.
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Avoid shifts by using convh functions.
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Use (1 << 31) as the multiplier for int<->float conversions. This makes
sure that int->float conversions always end up with floats between
[-1.0, 1.0].
For the conversion from float to int, this multiplier will give the complete
int range after we perform clipping.
Change the unit test to take this into consideration.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755301
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No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
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Use GstClockTimeDiff and Clock macros to print signed integer time
differences in the debug logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
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GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
handle negative values better.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
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Raise the frequency limit and try to negotiate to a samplerate of 4*freq
when larger then the default samplerate.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=754450
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Raise the channel limit and set the channel-mask for > 2 channels.
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Use the pack functions to also support the other audio formats we
have.
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To ensure the subtraction of two GstClockTime values (which are guint64)
can be negative. Use GST_CLOCK_DIFF which returns a gint64.
CID 1338049
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And use the profile called `default` if none provided.
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Otherwise the loop is useless!
Fixes CID 1338051
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https://bugzilla.gnome.org/show_bug.cgi?id=757068
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https://bugzilla.gnome.org/show_bug.cgi?id=757068
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https://bugzilla.gnome.org/show_bug.cgi?id=757068
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bindings
The client-removed signal used G_INT_TYPE instead of G_SOCKET_TYPE
in its definition leading to problems on platforms where the size
of a pointer is larger than the size of an integer, It would also
not work at all with dynamic language bindings.
https://bugzilla.gnome.org/show_bug.cgi?id=757155
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Due to a typo, videotestsrc did not handle the Bayer
format 'gbrg' properly and reported it as invalid,
causing negotiation errors.
https://bugzilla.gnome.org/show_bug.cgi?id=757264
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Rewrite audioconvert to try to make it more clear what steps are
executed during conversion.
Add passthrough step that just does a memcpy when possible.
Add ORC optimized dither and quantization functions.
Implement noise-shaping on S32 samples only and allow for arbitrary
noise shaping coefficients if we want this later.
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don't use gpointer * for something that should be gpointer.
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huge unsigned one
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The removed GList link needs to be freed too, and
the G_OPTION_REMAINING arguments need to be freed.
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There was already some code to handle that, but the support was not
complete in those code paths.
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encoding target
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Turn the quantizer into a reusable object.
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