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

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

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>

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>

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>

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

This makes the init files match the structure of the dsputil split.

Signed-off-by: Diego Biurrun <diego@biurrun.de>

Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>

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>

The remaining dsputil bits are encoding-specific anyway.

doc/examples/output.c:460:9: warning: unused variable ‘i’

The specification requires at most 1 track enabled per alternate group.

Sample-Id: OPFLAG_B_Qualcomm_1.bit, OPFLAG_C_Qualcomm_1.bit
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>

Signed-off-by: Diego Biurrun <diego@biurrun.de>

AVFormatContext->priv_data is not always a MpegTSContext, it can be
RTSPState when decoding a RTP stream. So it is necessary to pass
MpegTSContext pointer explicitly.

Within libav, the write_section_data function doesn't actually use
the MpegTSContext at all, so this doesn't change anything at the
moment (no memory was corrupted before), but it reduces the risk of
anybody trying to touch the MpegTSContext via AVFormatContext->priv_data
in the future.
Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>

Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>

Use it for logging, instead of NULL or the stream codec context.

Its contents are meaningful only if the stream codec context is the one
actually used for encoding, which is often not the case (and is
discouraged).

Use AVCodecContext.field_order instead.