dsputil: Drop unused bit_depth parameter from all init functions
examples/output: Remove unused variable

doc/examples/output.c:460:9: warning: unused variable ‘i’
ppc: dsputil: Coalesce all init files
x86: dsputil: Avoid pointless CONFIG_ENCODERS indirection

The remaining dsputil bits are encoding-specific anyway.
fate: Use the correct, local path to samples for opus reference files

This fixes running fate in configs where the samples are located
in a different path on the target.

Signed-off-by: Martin Storsjö <martin@martin.st>
g2meet: allow size changes within original sizes
arm: dsputil: Coalesce all init files
h264: Remove some commented-out, broken cruft
h264: K&R formatting cosmetics

Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
avpacket: Check for and return errors in ff_interleave_add_packet()

Signed-off-by: Diego Biurrun <diego@biurrun.de>
x86: dsputil: Coalesce all init files

This makes the init files match the structure of the dsputil split.
configure: Assume runtime cpu detection on arm on --target-os=android as well

Signed-off-by: Martin Storsjö <martin@martin.st>
dsputil: Split motion estimation compare bits off into their own context
armv6: Accelerate ff_imdct_half for general case (mdct_bits != 6)

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>
armv6: Accelerate ff_fft_calc for general case (nbits != 4)

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>
armv6: Accelerate vector_fmul_window

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 vector_fmul_window_c() / ff_vector_fmul_window_vfp() for the
same sample AAC stream:

                    Before          After
                    Mean   StdDev   Mean   StdDev  Confidence  Change
Audio decode        1598.2 47.4     1529.2 25.4    100.0%      +4.5%
vector_fmul_window  244.0  22.1     188.9  22.3    100.0%      +29.2%

Signed-off-by: Martin Storsjö <martin@martin.st>
armv6: Accelerate butterflies_float

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 butterflies_float_c() / ff_butterflies_float_vfp() for the
same sample AAC stream:

                   Before          After
                   Mean   StdDev   Mean   StdDev  Confidence  Change
Audio decode       1542.8 43.7     1470.5 41.5    100.0%      +4.9%
butterflies_float  130.0  11.9     70.2   12.1    100.0%      +85.2%

Signed-off-by: Martin Storsjö <martin@martin.st>
arm: cosmetics: Fix a misaligned asm operand

Signed-off-by: Martin Storsjö <martin@martin.st>
arm: cosmetics: Consistently use lowercase for shift operators

Signed-off-by: Martin Storsjö <martin@martin.st>
dct-test: Move cpu_flags variable out of global scope