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Lynne 2d85e6e723
ac3enc_fixed: convert to 32-bit sample format 
The AC3 encoder used to be a separate library called "Aften", which
got merged into libavcodec (literally, SVN commits and all).
The merge preserved as much features from the library as possible.

The code had two versions - a fixed point version and a floating
point version. FFmpeg had floating point DSP code used by other
codecs, the AC3 decoder including, so the floating-point DSP was
simply replaced with FFmpeg's own functions.
However, FFmpeg had no fixed-point audio code at that point. So
the encoder brought along its own fixed-point DSP functions,
including a fixed-point MDCT.

The fixed-point MDCT itself is trivially just a float MDCT with a
different type and each multiply being a fixed-point multiply.
So over time, it got refactored, and the FFT used for all other codecs
was templated.

Due to design decisions at the time, the fixed-point version of the
encoder operates at 16-bits of precision. Although convenient, this,
even at the time, was inadequate and inefficient. The encoder is noisy,
does not produce output comparable to the float encoder, and even
rings at higher frequencies due to the badly approximated winow function.

Enter MIPS (owned by Imagination Technologies at the time). They wanted
quick fixed-point decoding on their FPUless cores. So they contributed
patches to template the AC3 decoder so it had both a fixed-point
and a floating-point version. They also did the same for the AAC decoder.
They however, used 32-bit samples. Not 16-bits. And we did not have
32-bit fixed-point DSP functions, including an MDCT. But instead of
templating our MDCT to output 3 versions (float, 32-bit fixed and 16-bit fixed),
they simply copy-pasted their own MDCT into ours, and completely
ifdeffed our own MDCT code out if a 32-bit fixed point MDCT was selected.

This is also the status quo nowadays - 2 separate MDCTs, one which
produces floating point and 16-bit fixed point versions, and one
sort-of integrated which produces 32-bit MDCT.

MIPS weren't all that interested in encoding, so they left the encoder
as-is, and they didn't care much about the ifdeffery, mess or quality - it's
not their problem.

So the MDCT/FFT code has always been a thorn in anyone looking to clean up
code's eye.

Backstory over. Internally AC3 operates on 25-bit fixed-point coefficients.
So for the floating point version, the encoder simply runs the float MDCT,
and converts the resulting coefficients to 25-bit fixed-point, as AC3 is inherently
a fixed-point codec. For the fixed-point version, the input is 16-bit samples,
so to maximize precision the frame samples are analyzed and the highest set
bit is detected via ac3_max_msb_abs_int16(), and the coefficients are then
scaled up via ac3_lshift_int16(), so the input for the FFT is always at least 14 bits,
computed in normalize_samples(). After FFT, the coefficients are scaled up to 25 bits.

This patch simply changes the encoder to accept 32-bit samples, reusing
the already well-optimized 32-bit MDCT code, allowing us to clean up and drop
a large part of a very messy code of ours, as well as prepare for the future lavu/tx
conversion. The coefficients are simply scaled down to 25 bits during windowing,
skipping 2 separate scalings, as the hacks to extend precision are simply no longer
necessary. There's no point in running the MDCT always at 32 bits when you're
going to drop 6 bits off anyway, the headroom is plenty, and the MDCT rounds
properly.

This also makes the encoder even slightly more accurate over the float version,
as there's no coefficient conversion step necessary.

SIZE SAVINGS:
ARM32:
HARDCODED TABLES:
BASE           - 10709590
DROP  DSP      - 10702872 - diff:   -6.56KiB
DROP  MDCT     - 10667932 - diff:  -34.12KiB - both:   -40.68KiB
DROP  FFT      - 10336652 - diff: -323.52KiB - all:   -364.20KiB
SOFTCODED TABLES:
BASE           -  9685096
DROP  DSP      -  9678378 - diff:   -6.56KiB
DROP  MDCT     -  9643466 - diff:  -34.09KiB - both:   -40.65KiB
DROP  FFT      -  9573918 - diff:  -67.92KiB - all:   -108.57KiB

ARM64:
HARDCODED TABLES:
BASE           - 14641112
DROP  DSP      - 14633806 - diff:   -7.13KiB
DROP  MDCT     - 14604812 - diff:  -28.31KiB - both:   -35.45KiB
DROP  FFT      - 14286826 - diff: -310.53KiB - all:   -345.98KiB
SOFTCODED TABLES:
BASE           - 13636238
DROP  DSP      - 13628932 - diff:   -7.13KiB
DROP  MDCT     - 13599866 - diff:  -28.38KiB - both:   -35.52KiB
DROP  FFT      - 13542080 - diff:  -56.43KiB - all:    -91.95KiB

x86:
HARDCODED TABLES:
BASE           - 12367336
DROP  DSP      - 12354698 - diff:  -12.34KiB
DROP  MDCT     - 12331024 - diff:  -23.12KiB - both:   -35.46KiB
DROP  FFT      - 12029788 - diff: -294.18KiB - all:   -329.64KiB
SOFTCODED TABLES:
BASE           - 11358094
DROP  DSP      - 11345456 - diff:  -12.34KiB
DROP  MDCT     - 11321742 - diff:  -23.16KiB - both:   -35.50KiB
DROP  FFT      - 11276946 - diff:  -43.75KiB - all:    -79.25KiB

PERFORMANCE (10min random s32le):
ARM32 - before -  39.9x - 0m15.046s
ARM32 - after  -  28.2x - 0m21.525s
                       Speed:  -30%

ARM64 - before -  36.1x - 0m16.637s
ARM64 - after  -  36.0x - 0m16.727s
                       Speed: -0.5%

x86   - before - 184x -    0m3.277s
x86   - after  - 190x -    0m3.187s
                       Speed:   +3%
 		2021-01-14 01:44:12 +01:00
compat avfilter/scale_cuda: add lanczos algorithm 2020-11-04 01:43:21 +01:00
doc ac3enc_fixed: convert to 32-bit sample format 2021-01-14 01:44:12 +01:00
ffbuild ffbuild: Refine MIPS handling 2020-07-23 16:30:02 +02:00
fftools doc/ffmpeg: document max_error_rate 2021-01-10 19:14:37 +05:30
libavcodec ac3enc_fixed: convert to 32-bit sample format 2021-01-14 01:44:12 +01:00
libavdevice kmsgrab: Do not require the modifier to stay constant. 2021-01-13 23:08:02 +00:00
libavfilter avfilter/vf_convolution: use correct stride variable 2021-01-10 17:27:27 +01:00
libavformat avformat/allformats: test pointer to be used 2021-01-12 00:17:30 -03:00
libavresample
…
libavutil lavu: support arbitrary-point FFTs and all even (i)MDCT transforms 2021-01-13 17:34:13 +01:00
libpostproc lavu/mem: move the DECLARE_ALIGNED macro family to mem_internal on next+1 bump 2021-01-01 14:14:57 +01:00
libswresample swresample/audioconvert: Fix left shift of negative value 2020-09-30 10:50:45 +02:00
libswscale lavu/mem: move the DECLARE_ALIGNED macro family to mem_internal on next+1 bump 2021-01-01 14:14:57 +01:00
presets
…
tests ac3enc_fixed: convert to 32-bit sample format 2021-01-14 01:44:12 +01:00
tools tools/target_dem_fuzzer.c: Decrease maxblocks 2021-01-05 02:00:05 +01:00
.gitattributes
…
.gitignore
…
.mailmap mailmap: add entry for myself 2020-07-13 11:24:04 +08:00
.travis.yml
…
CONTRIBUTING.md
…
COPYING.GPLv2
…
COPYING.GPLv3
…
COPYING.LGPLv2.1
…
COPYING.LGPLv3
…
CREDITS
…
Changelog avfilter: add temporal midway equalizer filter 2021-01-01 12:43:42 +01:00
INSTALL.md
…
LICENSE.md
…
MAINTAINERS smvjpegdec: merge into mjpegdec 2020-12-10 10:07:09 +01:00
Makefile tools/enum_options: fix build and add to Makefile 2020-11-20 15:20:24 +01:00
README.md
…
RELEASE RELEASE: We are after the 4.3 branch point, update for that 2020-06-10 00:20:24 +02:00
configure configure: update copyright year 2021-01-01 00:38:22 +01:00

README.md

FFmpeg README

FFmpeg is a collection of libraries and tools to process multimedia content such as audio, video, subtitles and related metadata.

Libraries

  • libavcodec provides implementation of a wider range of codecs.
  • libavformat implements streaming protocols, container formats and basic I/O access.
  • libavutil includes hashers, decompressors and miscellaneous utility functions.
  • libavfilter provides a mean to alter decoded Audio and Video through chain of filters.
  • libavdevice provides an abstraction to access capture and playback devices.
  • libswresample implements audio mixing and resampling routines.
  • libswscale implements color conversion and scaling routines.

Tools

  • ffmpeg is a command line toolbox to manipulate, convert and stream multimedia content.
  • ffplay is a minimalistic multimedia player.
  • ffprobe is a simple analysis tool to inspect multimedia content.
  • Additional small tools such as aviocat, ismindex and qt-faststart.

Documentation

The offline documentation is available in the doc/ directory.

The online documentation is available in the main website and in the wiki.

Examples

Coding examples are available in the doc/examples directory.

License

FFmpeg codebase is mainly LGPL-licensed with optional components licensed under GPL. Please refer to the LICENSE file for detailed information.

Contributing

Patches should be submitted to the ffmpeg-devel mailing list using git format-patch or git send-email. Github pull requests should be avoided because they are not part of our review process and will be ignored.