The sample rates have already been checked generically
via AVCodec.supported_samplerates.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The LTP profile of AAC is... terrible.
It was an early 90's attempt at bridging the gap between speech
codecs and general purpose codecs. It did so by trying to exploit the fact
that most speech patterns are regular.
Unfortunately, it went about it the same way as AAC Main, by taking
the previous frame's samples, modifying them through an LPC filter,
transforming them back using a forward MDCT, putting the output
coefficients back into the current frame, and using delta coding.
But once again, they ignored basic mathematics and MDCT leakage.
Thankfully, because AAC LTP is meant to operate at very low bitrates,
the extreme quantization results in most leakage being irrelevant.
Unfortunately, the result is that the output sounds pretty much
terrible regardless of whether LTP is enabled or not.
This was the first attempt at trying to couple speech coding into AAC.
No, the second attempt did not succeed either.
Nnnneither did the third. Or fourth.
For the fifth one, they literally just jammed a speech codec into AAC
with USAC once they saw Opus do it.
Just drop support for encoding AAC LTP. It was always experimental
to begin with.
The Main profile of AAC is... terrible.
It enables the use of delta coding across coefficients of two frames
to try to increase compression, and it enabled one more pole for TNS
filters.
What the AAC authors failed to take into account were basic
mathematics, as MDCT leakage (e.g. the spread of each frequency when
represented in a discrete spectrum) is significant in most audio codecs.
This leads to huge variations between each frame, basically rendering
prediction completely pointless.
In fact, enabling AAC-Main prediction does not, in general, even recoup
the metadata losses from signalling the profile and prediction properties
in the first place. So you lose efficiency by using AAC Main.
The rumor is that it was put in the AAC spec for patent reasons, though
patent-wise, it has about as much use as a patent for a bicycle designed
for use by snakes.
The only other thing AAC Main changes is it permits 3-pole TNS filters.
When AAC's bands are absolutely tiny, except for very high frequency bands,
where you're likely to use PNS instead.
Just get rid of it.
The AAC decoders share no common code with the AAC encoder,
so they are not restricted to using the same structures.
This implies that one can use different structs for each
component and remove elements not used by the decoders/
the encoder. This leads to quite sizeable savings:
sizeof(ChannelElement) for the encoder went down to 134432B
here from 547552B; for the decoder it went down to 512800B.
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
These defines are also used in other contexts than just AVCodecContext
ones, e.g. in libavformat. Furthermore, given that these defines are
public, the AV-prefix is the right one, so deprecate (and not just move)
the FF-macros.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
avpriv_mpeg4audio_sample_rates has a size of 64B and it is currently
avpriv; a clone of it exists in aacenctab.h and from there it is inlined
in aacenc.c (which also uses the avpriv version) and in the FLV muxer.
This means that despite it being avpriv both libavformat as well as
libavcodec have copies already.
This situation is clearly suboptimal. Given the overhead of exporting
symbols (for x64 Elf/Linux/GNU: 2x2B version, 2x24B .dynsym, 24B .rela.dyn,
8B .got, 4B hash + twice the size of the name (here 31B)) the object is
unavprived, i.e. duplicated into libavformat when creating a shared
build; but the duplicates in the AAC encoder and FLV muxer are removed.
This involves splitting of the sample rate table into a file of its own;
this allowed to break some spurious dependencies (e.g. both the AAC
encoder as well as the Matroska demuxer actually don't need the
mpeg4audio_get_config stuff).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The spec is correct, it does list these layouts as having rear speakers.
Questionable how many decoders correctly interpret those correctly since
side is way more popular.
Also fixes fate-aac-yoraw-encode.
Reported-by: pkviet <pkv.stream@gmail.com>
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit implements support for PCE (Program Configuration Elements) in the
AAC encoder, and as such allows for encoding of channel layouts not present
in the presets defined by the spec (which only lists the 8 most common ones).
This has been a highly requested feature and is also the first open source encoder
to support this many layouts.
Many thanks to pkviet <pkv.stream@gmail.com> who implemented support for and
verified all channel layouts.
It didn't work out because of the exceptions that needed to be made
for the "-1" cases and was overall more confusing that just manually
checking and setting options for each profile.
This commit adds the ability for a profile to set the default
options, as well as for the user to override such options
by simply stating them in the command line while still keeping
the same profile, as long as those options are still permitted by
the profile.
Example: setting the profile to aac_low (the default) will turn
PNS and IS on. They can be disabled by -aac_pns 0 and -aac_is 0,
respectively. Turning on -aac_pred 1 will cause the profile to be
elevated to aac_main, as long as no options forbidding aac_main
have been entered (like AAC-LTP, which will be pushed soon).
A useful feature is that by setting the profile to mpeg2_aac_low,
all MPEG4 features will be disabled and if the user tries to enable
them then the program will exit with an error. This profile is
signalled with the same bitstream as aac_low (MPEG4) but some devices
and decoders will fail if any MPEG4 features have been enabled.
This commit implements support for 7.1 channel audio. There's no
more predefined bitstream channel mappings so going beyond 8 channels
(and 7 channels exactly) will require programmable channel elements,
which is already underway.
Needed for following commits. Contains the starting sfb for
every samplerate and window type.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit moves any tables specific to the encoder from aacenc
and aaccoder to a separate file called 'aacenctab.c/.h'.
This was done as a clean up attempt as the encoder was filled with
tables pasted in between functions which made it confusing to follow
and track where each table and definition had been used.
This commit solves this by simply exporting the smaller tables out to
the aacenctab.h while the larger ones are compiled using aacenctab.c
and are referenced from the header file.
Signed-off-by: Claudio Freire <klaussfreire@gmail.com>
