This reduces code duplication and differences with the fork.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Initialise VC1DSPContext for parser as well as for decoder.
Note, the VC-1 code doesn't actually use the function pointer yet.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* commit '7fb993d338d88f2f62e0a358b6c9f3eb9a3a08ac':
qpeldsp: Mark source pointer in qpel_mc_func function pointer const
Conflicts:
libavcodec/h264qpel_template.c
libavcodec/x86/cavsdsp.c
libavcodec/x86/rv40dsp_init.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6869612f5c7d4d2f20f69a5658328a761deadb1c':
arm: Macroize the test for 'setend' CPU instruction support
Conflicts:
libavcodec/arm/h264dsp_init_arm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4de8b60684ce13dff3e3d372dae4f49b9e53f755':
idct: Move arm-specific declarations to a header in the arm directory
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7e18a727d2c2a19f22fcf68875d1b05fd2eafcef':
arm: cosmetics: Consistently use lowercase for shift operators
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '87552d54d3337c3241e8a9e1a05df16eaa821496':
armv6: Accelerate ff_fft_calc for general case (nbits != 4)
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous implementation targeted DTS Coherent Acoustics, which only
requires nbits == 4 (fft16()). This case was (and still is) linked directly
rather than being indirected through ff_fft_calc_vfp(), but now the full
range from radix-4 up to radix-65536 is available. This benefits other codecs
such as AAC and AC3.
The implementaion is based upon the C version, with each routine larger than
radix-16 calling a hierarchy of smaller FFT functions, then performing a
post-processing pass. This pass benefits a lot from loop unrolling to
counter the long pipelines in the VFP. A relaxed calling standard also
reduces the overhead of the call hierarchy, and avoiding the excessive
inlining performed by GCC probably helps with I-cache utilisation too.
I benchmarked the result by measuring the number of gperftools samples that
hit anywhere in the AAC decoder (starting from aac_decode_frame()) or
specifically in the FFT routines (fft4() to fft512() and pass()) for the
same sample AAC stream:
Before After
Mean StdDev Mean StdDev Confidence Change
Audio decode 2245.5 53.1 1599.6 43.8 100.0% +40.4%
FFT routines 940.6 22.0 348.1 20.8 100.0% +170.2%
Signed-off-by: Martin Storsjö <martin@martin.st>
The previous implementation targeted DTS Coherent Acoustics, which only
requires mdct_bits == 6. This relatively small size lent itself to
unrolling the loops a small number of times, and encoding offsets
calculated at assembly time within the load/store instructions of each
iteration.
In the more general case (codecs such as AAC and AC3) much larger arrays
are used - mdct_bits == [8, 9, 11]. The old method does not scale for
these cases, so more integer registers are used with non-unrolled versions
of the loops (and with some stack spillage). The postrotation filter loop
is still unrolled by a factor of 2 to permit the double-buffering of some
VFP registers to facilitate overlap of neighbouring iterations.
I benchmarked the result by measuring the number of gperftools samples
that hit anywhere in the AAC decoder (starting from aac_decode_frame())
or specifically in ff_imdct_half_c / ff_imdct_half_vfp, for the same
example AAC stream:
Before After
Mean StdDev Mean StdDev Confidence Change
aac_decode_frame 2368.1 35.8 2117.2 35.3 100.0% +11.8%
ff_imdct_half_* 457.5 22.4 251.2 16.2 100.0% +82.1%
Signed-off-by: Martin Storsjö <martin@martin.st>
The previous implementation targeted DTS Coherent Acoustics, which only
requires mdct_bits == 6. This relatively small size lent itself to
unrolling the loops a small number of times, and encoding offsets
calculated at assembly time within the load/store instructions of each
iteration.
In the more general case (codecs such as AAC and AC3) much larger arrays
are used - mdct_bits == [8, 9, 11]. The old method does not scale for
these cases, so more integer registers are used with non-unrolled versions
of the loops (and with some stack spillage). The postrotation filter loop
is still unrolled by a factor of 2 to permit the double-buffering of some
VFP registers to facilitate overlap of neighbouring iterations.
I benchmarked the result by measuring the number of gperftools samples
that hit anywhere in the AAC decoder (starting from aac_decode_frame())
or specifically in ff_imdct_half_c / ff_imdct_half_vfp, for the same
example AAC stream:
Before After
Mean StdDev Mean StdDev Confidence Change
aac_decode_frame 2368.1 35.8 2117.2 35.3 100.0% +11.8%
ff_imdct_half_* 457.5 22.4 251.2 16.2 100.0% +82.1%
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f46bb608d9d76c543e4929dc8cffe36b84bd789e':
dsputil: Split off pixel block routines into their own context
Conflicts:
configure
libavcodec/dsputil.c
libavcodec/mpegvideo_enc.c
libavcodec/pixblockdsp_template.c
libavcodec/x86/dsputilenc.asm
libavcodec/x86/dsputilenc_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '79fce1ec8abd017593c003917fc123f7119a78d6':
arm: Avoid using the 'setend' instruction on ARMv7 and newer
Conflicts:
libavcodec/arm/h264dsp_init_arm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f23d26a6864128001b03876b0b92fffe131f2060':
h264: avoid using uninitialized memory in NEON chroma mc
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9a9e2f1c8aa4539a261625145e5c1f46a8106ac2':
dsputil: Split audio operations off into a separate context
Conflicts:
configure
libavcodec/takdec.c
libavcodec/x86/Makefile
libavcodec/x86/dsputil.asm
libavcodec/x86/dsputil_init.c
libavcodec/x86/dsputil_mmx.c
libavcodec/x86/dsputil_x86.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
