Ben Avison [Thu, 10 Jul 2014 23:12:34 +0000 (00:12 +0100)]
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%
Ben Avison [Thu, 10 Jul 2014 23:12:33 +0000 (00:12 +0100)]
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%
Ben Avison [Wed, 16 Jul 2014 15:02:01 +0000 (16:02 +0100)]
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%
Ben Avison [Thu, 10 Jul 2014 23:12:31 +0000 (00:12 +0100)]
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%
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.
Andrew Kelley [Sun, 6 Jul 2014 05:24:43 +0000 (22:24 -0700)]
af_compand: make sure request_frame always outputs at least one frame
This fixes a segmentation fault because request_frame in fifo.c assumes
that the call to ff_request_frame will populate fifo->root.next.
Before, it was possible for request_frame in af_compand to not do this,
resulting in a null pointer access. Now, request_frame in af_compand
always will return at least one frame or an error, as per the API
specifications in avfilter.h for request_frame.
Omer Osman [Mon, 7 Jul 2014 08:28:12 +0000 (10:28 +0200)]
libfdk-aacdec: Decode the first AAC frame to reliably identify the bitstream
For implicit signaling cases (as possible for Spectral Band Replication
and Parametric Stereo Tools), the decoder must decode the first frame to
correctly identify the stream configuration (as called from
avformat_find_stream_info). The mechanism for this is built-in and only
requires adding CODEC_CAP_CHANNEL_CONF to the libfdk-aacdec AVCodec
struct.
Signed-off-by: Omer Osman <omer.osman@iis.fraunhofer.de> Signed-off-by: Martin Storsjö <martin@martin.st>
Omer Osman [Fri, 4 Jul 2014 09:05:44 +0000 (11:05 +0200)]
libfdk-aacdec: Reduce the default decoder delay by one frame
The default error concealment method if none is set via
aacDecoder_SetParam(AAC_CONCEAL_METHOD) is set in
CConcealment_InitCommonData within the fdk-aac library
and is set to Energy Interpolation. This method requires one frame
delay to the output. To reduce the default decoder output delay and
avoid missing the last frame in file based decoding, use Noise
Substitution as the default concealment method.
Signed-off-by: Omer Osman <omer.osman@iis.fraunhofer.de> Signed-off-by: Martin Storsjö <martin@martin.st>
projects / ffmpeg / log