This patch replaces the transform used in AAC with lavu/tx and removes
the limitation on only being able to decode 960-sample files
with the float decoder.
This commit also removes a whole bunch of unnecessary and slow
lifting steps the decoder did to compensate for the poor accuracy
of the old integer transformation code.
Overall float decoder speedup on Zen 3 for 64kbps: 32%
Generalize the checks for channels in all positions, and properly support
the three height groups (normal, top, bottom) instead of manually setting
the relevant channels for the latter two after the normal height tags were
parsed.
Signed-off-by: James Almer <jamrial@gmail.com>
If PCE defines channels not covered by those in the standard configurations
then don't try to come up with some made up layout and just return them in the
coded order.
Fixes al08_44.mp4 from the conformance suite, now reporting and decoding all 48
channels instead of 10.
Signed-off-by: James Almer <jamrial@gmail.com>
Set the correct amount of tags in tags_per_config[].
Also, there are no channels that correspond to a side element in this
configuration, so reflect this in the list of known/supported channel layouts.
Signed-off-by: James Almer <jamrial@gmail.com>
Only used by decoders (encoders have ff_encode_alloc_frame()).
Also clean up the other headers a bit while removing now redundant
internal.h inclusions.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes outputting silence on the second channel when decoding Parametic Stereo
HE-AAC.
Closes ticekt #3361.
Signed-off-by: James Almer <jamrial@gmail.com>
Since this behavior is intentional, use the VERBOSE level instead of WARNING as
it's nothing the user should worry about.
Signed-off-by: James Almer <jamrial@gmail.com>
The amount of padding samples reported by containers take into account the
extended samplerate in HE-AAC.
Fixes ticket #9671.
Signed-off-by: James Almer <jamrial@gmail.com>
Dual mono files report a channel count of 2 with each individual channel in its
own SCE, instead of both in a single CPE as is the case with standard stereo.
This commit handles this non default channel configuration scenario.
Fixes ticket #1614
Signed-off-by: James Almer <jamrial@gmail.com>
In C, qualifiers for arrays are broken:
const VLC_TYPE (*foo)[2] is a pointer to an array of two const VLC_TYPE
elements and unfortunately this is not compatible with a pointer
to a const array of two VLC_TYPE, because the latter does not exist
as array types are never qualified (the qualifier applies to the base
type instead). This is the reason why get_vlc2() doesn't accept
a const VLC table despite not modifying the table at all, as
there is no automatic conversion from VLC_TYPE (*)[2] to
const VLC_TYPE (*)[2].
Fix this by using a structure VLCElem for the VLC table.
This also has the advantage of making it clear which
element is which.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is more spec-compliant because it does not rely
on dead-code elimination by the compiler. Especially
MSVC has problems with this, as can be seen in
https://ffmpeg.org/pipermail/ffmpeg-devel/2022-May/296373.html
or
https://ffmpeg.org/pipermail/ffmpeg-devel/2022-May/297022.html
This commit does not eliminate every instance where we rely
on dead code elimination: It only tackles branching to
the initialization of arch-specific dsp code, not e.g. all
uses of CONFIG_ and HAVE_ checks. But maybe it is already
enough to compile FFmpeg with MSVC with whole-programm-optimizations
enabled (if one does not disable too many components).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This increases type-safety by avoiding conversions from/through void*.
It also avoids the boilerplate "AVFrame *frame = data;" line
for non-subtitle decoders.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Also move FF_CODEC_TAGS_END as well as struct AVCodecDefault.
This reduces the amount of files that have to include internal.h
(which comes with quite a lot of indirect inclusions), as e.g.
most encoders don't need it. It is furthemore in preparation
for moving the private part of AVCodec out of the public codec.h.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes: negation of -2147483648 cannot be represented in type 'INTFLOAT' (aka 'int'); cast to an unsigned type to negate this value to itself
Fixes: 29057/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5642758933053440
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The fixed-point AAC decoder is the only user of the fixed-point sinewin
tables from sinewin; and it only uses a few of them (about 10% when
counting by size). This means that guarding initializing these tables by
an AVOnce (as done in 3719122065) is
unnecessary for them. Furthermore the array of pointers to the
individual arrays is also unneeded.
Therefore this commit moves these tables directly into aacdec_fixed.c;
this is done by ridding the original sinewin.h and sinewin_tablegen.h
headers completely of any fixed-point code at the cost of a bit of
duplicated code (the alternative is an ugly ifdef-mess).
This saves about 58KB from the binary when using hardcoded tables (as
these tables are hardcoded in this scenario); when not using hardcoded
tables, most of these savings only affect the .bss segment, but the rest
(< 1KB) contains relocations (i.e. savings in .data.rel.ro).
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The floating point kbd tables for 120 and 960 samples are only used by
the floating point decoder whereas the fixed point kbd tables for 128
and 1024 samples are only used by the fixed point AAC decoder. So move
these tables to their only users. This ensures that they are not
accidentally used somewhere else without ensuring that initializing
these tables stays thread-safe (as it is now because the only place from
where they are initialized is guarded by an AVOnce).
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The floating point AAC decoder is the only user of these tables, so it
makes sense to move them there. Furthermore, initializing the ordinary
power-of-two sinetables is currently not thread-safe and if the 120- and
960-point sinetables were not moved, one would have to choose whether
to guard initializing these two tables with their own AVOnces or not.
Doing so would add unnecessary AVOnces as the AAC decoder already guards
initializing its static data by an AVOnce; not doing so would be fragile
if a second user of these tables were to be added.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Expressions like array[get_vlc2()] can be optimized by using a symbols
table if the array is always the same for a given VLC. This requirement
is fulfilled for several VLCs used by the AAC decoders.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Validates the set channel layout as well as verifies that the received
layout to the function matches the reference layout, so that it matches
the implemented re-ordering logic.
Fixes#8845
Fixes: signed integer overflow: 2 * 1881153568 cannot be represented in type 'int'
Fixes: 17996/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5687126468853760
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Identical to avpriv_mpeg4audio_get_config() except taking a size argument in
bytes, and featuring a new logging context paremeter.
Schedule avpriv_mpeg4audio_get_config() for removal as soon as major is bumped
as well.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes: signed integer overflow: 2147483645 + 4 cannot be represented in type 'int'
Fixes: 15418/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_AAC_FIXED_fuzzer-5685269069561856
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Such streams are invalid according to
4.5.2.1 Top level payloads for the audio object types AAC main, AAC SSR, AAC LC and AAC LTP
4.5.2.1.1 Definitions
...cIn the raw_data_block(), several instances of the
same syntactic element may occur, but must have a different 4 bit
element_instance_tag, except for data_stream_element()'s and
fill_element()'s.
Fixes: Ticket7477
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
- enable the parsing code
- use the new buffer instead of replacing the context one
- do not push/pop configuration, just discard the exiting one
- propagate errors correctly
This will result in poor quality audio for SSR streams, but they
will at least demux and decode without error; partially fixing
ticket #1693.
This pulls in the decode_gain_control() function from the
ffmpeg summer-of-code repo (original author Maxim Gavrilov) at
svn://svn.mplayerhq.hu/soc/aac/aac.c with some minor modifications
and adds AOT_AAC_SSR to decode_audio_specific_config_gb().
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Co-authored-by: Maxim Gavrilov <maxim.gavrilov@gmail.com>
Fixes: signed integer overflow: -1625276744 + -1041893960 cannot be represented in type 'int'
Fixes: 5948/clusterfuzz-testcase-minimized-5791479856365568
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'b5f19f7478492307e4b4763aeac3180faf50e17f':
aac: Split function to parse ADTS header data into public and private part
Merged-by: James Almer <jamrial@gmail.com>
Lynne
James Almer
Andreas Rheinhardt
Anton Khirnov
Michael Niedermayer
Andreas Rheinhardt
KM
Xiaohan Wang
Jan Ekström
Carl Eugen Hoyos
Vittorio Giovara
Dale Curtis