When an empty mix matrix is passed, audio-channel-mixer
will now generate a (potentially truncated) identity matrix,
this replicates the behaviour of audiomixmatrix in first-channels
mode.
https://bugzilla.gnome.org/show_bug.cgi?id=788833
remove_format_info was a bit confusing to read, this removes
it in favor of standard gst_caps_map_in_place calls.
This no longer simplifies the resulting caps, but I
consider this should be the job of basetransform.
https://bugzilla.gnome.org/show_bug.cgi?id=785471
channels=1 is always mono, having it 'unpositioned' does not make
sense.
This fixes pipeline such as:
gst-validate-1.0 audiotestsrc ! audio/x-raw,channels=2,rate=44100,layout=interleaved ! audioconvert ! audioresample ! audio/x-raw, rate=44100, channels=1 ! avenc_mp2 ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=785407
Optimize LE<->BE conversion by adding a dedicated fast path instead of
using the generic converter. Implement transform_ip function in order to do the
endian swap in place.
This saves buffer allocation for the intermediate format, can be done in place
and also performs the conversion in one step instead of unpack-convert-pack.
For all bit widths the naive algorithm is implemented, which provides the best
performance when compiled with -O3. ORC was considered but eventually removed
as it requires a dedicated function for in-place conversion (due to the
"restrict" parameters).
A more complex algorithm for the 24-bit conversion with unrolled loop and
32-bit processing is implemented in the #if 0 section. It performs better if
compiled with -O2. With -O3 however the naive algorithm performs better.
https://bugzilla.gnome.org/show_bug.cgi?id=773073
find_suitable_mask() had complexity O(n^2) on the number of bits.
For common case like 2-channel audio the mask was calculated in about 4k loop
cycles.
Optimize both n_bits_set() and find_suitable_mask() to O(n) where n is the
number of bits set in the mask.
https://bugzilla.gnome.org/show_bug.cgi?id=772864
Pass flags in _converter_new() so that we can configure ourselves
differently depending on some options.
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
Simplify the API, we don't need the consumed and produced output
arguments. The caller needs to use the _get_in_frames/get_out_frames API
to check how much input is needed and how much output will be produced.
In this specific case it wouldn't cause problems as we only ever access the
first array element, but let's make explicit what is happening here.
CID 1346530 and 1346529
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
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.
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()
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
Previously we could've chosen another format with the same
depth even if the input format was possible.
Also make sure to chose according to the order in the
caps.
Enhance current code to prefer an exact match on sample depth if
possible. Also ignore GST_AUDIO_FORMAT_FLAG_UNPACK when checking
equality on the flags.
