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* commit 'a24a252709dd38f12aa4929ce4981f87091a5113':
aarch64: NEON optimized FIR audio resampling
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Optimized for the default filter length 16.
30% faster opus silk decoding.
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* commit 'cae8df78759c2e69257f7fe58842f34c0d98a7ec':
lavr: define ResampleContext in resample.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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Required for arch optimized resampling.
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* commit '21d8f4da91673051ad7055cf63d989488a56368e':
resample: remove an unneeded context variable
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* commit 'f7c5fd81516fb7b186c64c66dc080e1139586226':
resample: implement flushing
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* commit '254c95cdd1b669a722204a4a65974fc2523c8f83':
resample: split linear into its own function
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* commit 'be394968c81019887ef996a78a526bdd85d1e216':
resample: add initial padding explicitly
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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This simplifies the code, since we do not have to deal with a possibly
negative source index anymore.
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* commit 'f20892eb67a7b4b5a3c080388a35a3380516a805':
resample: split the nearest neighbour path into a separate function pointer
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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* commit 'b9dea23766f52b8e059e72d34980bb7b456efe8f':
resample: fix avresample_get_delay() return value
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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The correct "next" input sample is not the first sample of the
resampling buffer, but the center sample of the filter_length-sized
block at the beginning.
CC:libav-stable@libav.org
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Simplifies header dependencies by not including all other internal headers
in internal.h.
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It adds unnecessary complication for insignificant usability improvement.
The user really should know if they'll need resampling compensation before
opening the context.
Note that only the documentation has changed. The current functionality will
still work until the next major bump.
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Since the resampler handles buffering of unconsumed samples internally, the
caller does not need this information.
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Based partially on implementation by Michael Niedermayer <michaelni@gmx.at> in
libswresample in FFmpeg. See commits:
7f1ae79d38c4edba9dbd31d7bf797e525298ac55
24ab1abfb6d55bf330022df4b10d7aec80b3f116
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This is a new library for audio sample format, channel layout, and sample rate
conversion.
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