Both the fixed as well as the floating point mpegaudio decoders use
LUTs of type int8_t and uint32_t with 32K entries each; these tables
are completely the same, yet they are not shared. This commit makes
them shared. When both fixed as well as floating point decoders are
enabled, this saves 160KiB from the bss segment for a normal build
(translating into 160KiB less memory usage if both a shared as well as
a floating point decoder have actually been used) and 160KiB from the
binary for a build with hardcoded tables.
It also means that the code to create said LUTs is no longer duplicated
(for a normal build).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The csa_tables (which always consist of 32 entries of four byte each,
but the type depends upon whether the decoder is fixed or
floating-point) are currently initialized once during decoder
initialization; yet it turns out that this is actually no benefit: The
code used to initialize these tables takes up 153 (fixed point) and 122
(floating point) bytes when compiled with GCC 9.3 with -O3 on x64, so it
is better to just hardcode these tables.
Essentially the same applies to the is_tables: They have a size of 128B
each and the code to initialize them occupies 149 (fixed point) resp.
140 (floating point) bytes. So hardcode them, too.
To make the origin of the tables clear, references to the code used to
create them have been added.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Up until now, there were several indiviual tables which were accessed
via pointers to them; by combining the tables, one can avoid said
pointers, saving space.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
By switching from ff_init_vlc_sparse() to ff_init_vlc_from_lengths() one
can replace tables of codes of type uint16_t by tables of symbols of
type uint8_t; this saves about 1.3KB for both the fixed and floating
point decoders (if enabled).
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
These arrays are used by the Musepack decoders, the MPEG audio decoders
as well as qdm2 and up until now, these arrays might be initialized more
than once, leading to potential data races as well as unnecessary
initializations. Therefore this commit ensures that each array will only
be initialized once.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Do this by only keeping the only function pointer from the
AVFloatDSPContext that is needed lateron. This also allows to remove the
decoders' close function.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
This differs from the MPEG specification as the actual real world
files do compute their CRC over variable areas and not the fixed
ones listed in the specification. This is also the reason for
the complexity of this code and the need to perform the CRC
check for layer2 in the middle of layer2 decoding.
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Layers 1 and 2 use lengths in bits which are not a multiple of 8,
and our CRC works on a per-byte basis.
Based on b48397e7b8
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: James Almer <jamrial@gmail.com>
This reverts commit b48397e7b8.
The change did not disable crc checks for layer 1 & 2, it removed reading
the CRC field.
Fixes decoding some mp2 samples and FATE test failures.
Signed-off-by: James Almer <jamrial@gmail.com>
A lot of files have CRC included.
The CRC only covers 34 bytes at most from the frame but it should still be
enough for some amount of error detection.
Initialize the bit buffer with the correct size (amount of bits that will
be read) instead of relying on the bitstream reader overreading the
correct values.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This can be used to simplify code in a couple of places.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Almost all the places from which this function is called already check
the header manually and in the two that don't (the mp3 muxer) the check
should not cause any problems.
Commits 43bc5cf9 and c5371f77 add code for skipping initial zeros in mp3
packets. This code forgot to report to the user that data was skipped at
all.
Since audio codecs allow partial packet decoding, the user application
has to rely on the return value. It will remove the data reported as
consumed by the decoder, and feed it to the decoder again. This resulted
in the mp3 frame after the zero region to be decoded over and over
again, until the zero region was finally skipped by the application.
Fix this by including the amount of skipped bytes to the number of
consumed bytes returned by the decode call.
Fixes trac ticket #4890.
This fixes a segfault when decoding multi-channel MP3onMP4 files.
This is similar to commit cb72230d for MPADSPContext.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
14496-3 suggests packing main_data of MP3 that is usually scattered
into multiple frames due to bit reservoir.
However, after packing main_data into a access unit, bitrate index
in the MPEG audio frame header doesn't match with actual frame size.
In order to accept this, this patch removes unnecessary frame size
checking on mp3 decoder.
Also, mov demuxer was changed to use MP3 parser only on special cases
(QT MOV with specific sample description) to avoid re-packetizing.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Anton Khirnov
Andreas Rheinhardt
Michael Niedermayer
James Almer
Lynne
Clément Bœsch
Andreas Cadhalpun
Diego Biurrun
Luca Barbato
Vittorio Giovara
Lou Logan
Vicente Olivert Riera
wm4
nu774
Michael Niedermayer