| Commit message (Collapse) | Author | Age |
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Signed-off-by: Diego Biurrun <diego@biurrun.de>
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This fixes a -Wunused-variable warning.
Signed-off-by: Ganesh Ajjanagadde <gajjanagadde@gmail.com>
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Bug-Id: CID 1323180
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Deprecated in 11/2012.
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Fixes the corner case in which the internal buffer size
is larger than input buffer provided and resizing it
before moving the left over samples would make it write
to now unallocated memory.
Bug-Id: 825
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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This applies to every library where performance is not critical.
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modelled after aarch64 code
on Cortex-A8, s16 and s32 code is about 2x faster,
float code about 7x faster
Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net>
Signed-off-by: Martin Storsjö <martin@martin.st>
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Optimized for the default filter length 16.
30% faster opus silk decoding.
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Required for arch optimized resampling.
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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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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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