| Commit message (Collapse) | Author | Age |
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This matches the name of the codec tested.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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These tests include adpcm-ima-dk3 audio which is tested elsewhere.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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These files contain mace6 audio which is tested elsewhere.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This file has eamad video which is tested elsewhere.
Also rename the test to reflect this change.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This test contains raw rgb565le video. Converting to rgb24 serves
no useful purpose here.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Fix some orthography, wording and grammar issues; update the SDL section
with more current instructions; simplify lib.exe example command line;
drop outdated comments about libnut.
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Signed-off-by: Diego Elio Pettenò <flameeyes@flameeyes.eu>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Change the size specifiers to match the actual element sizes
of the data. This makes no practical difference with strict
alignment checking disabled (the default) other than somewhat
documenting the code. With strict alignment checking on, it
avoids trapping the unaligned loads.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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Converting the double to float for lrintf() loses precision when
the value is not exactly representable as a single-precision float.
Apart from being inaccurate, this causes discrepancies in some
configurations due to differences in rounding.
Note that the changed timestamp in the vc1-ism test is a bogus,
made-up value.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Code mostly inspired by vp8's MC, however:
- its MMX2 horizontal filter is worse because it can't take advantage of
the coefficient redundancy
- that same coefficient redundancy allows better code for non-SSSE3 versions
Benchmark (rounded to tens of unit):
V8x8 H8x8 2D8x8 V16x16 H16x16 2D16x16
C 445 358 985 1785 1559 3280
MMX* 219 271 478 714 929 1443
SSE2 131 158 294 425 515 892
SSSE3 120 122 248 387 390 763
End result is overall around a 15% speedup for SSSE3 version (on 6 sequences);
all loop filter functions now take around 55% of decoding time, while luma MC
dsp functions are around 6%, chroma ones are 1.3% and biweight around 2.3%.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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This involves a division that should be a shift.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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The vertically interpolating variants of these functions read
ahead one line to optimise the loop. On the last line processed,
this might be outside the buffer. Fix these invalid reads by
processing the last line outside the loop.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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According to the behaviour of librtmp, it is recommended to send this
message to the server after receiving the 'onBWDone' callback in order
to do bandwidth checking and improve compatibility with some servers.
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player.
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Flash plugin.
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stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This fixes the warning:
cmdutils.c:897: warning: ‘type’ may be used uninitialized in this function
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Replace SOL_SCTP by the more portable IPPROTO_SCTP.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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This is needed to expose some networking APIs.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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The current SSE version is slower than the MMX version on Athlon64 and Sandy
Bridge, but the SSE4 and AVX versions are faster on Sandy Bridge.
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This is easier to follow than having them randomly scattered in
avfilter.c and defaults.c.
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Those functions are only useful inside filters. It is better to not
support user filters until the API is more stable.
This breaks audio filtering API and ABI in theory, but since it's
unusable right now this shouldn't be a problem.
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There is no real reason for it to be 64bit, it's just a plain int in the
rest of Libav.
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There's no reason for it to be explicitly 32 bits. It's declared as a
plain int in all other places in Libav.
This breaks audio filtering API and ABI in theory, but since it's
unusable right now this shouldn't be a problem.
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The additional parameters are just complicating the function interface.
Assume that a requested samples buffer will *always* have the format
specified in the requested link.
This breaks audio filtering API and ABI in theory, but since it's
unusable right now this shouldn't be a problem.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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avfilter_default_get_audio_buffer
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Remove AVFilterBufferRefAudioProps.size, and use nb_samples in its place
everywhere.
This is required as the size in the audio buffer may be aligned, so it
may not contain a well defined number of samples.
Also remove the useless planar parameter, which can be deduced from the
sample format.
This is technically an API and ABI break, but since the audio part of
lavfi is not usable now, this should not be a problem in practice.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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This allows compiling the Xvid rate control code without the encoder.
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libavcodec/jpeglsdec.c:54:9: warning: variable ‘len’ set but not used
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