ac may be NULL and then accessing ac->avctx results in a segmentation fault.
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit replaces the previous hardcoded constants with both new and previously
defined macros from aac.h. This change makes it easy for anyone reading the code
to know how encoding and decoding scalefactors works. It's also possibly
a step in unifying some of the code across both the encoder and decoder.
Reviewed-by: Claudio Freire <klaussfreire@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Although the specification mandates this bit to zero, it may happen
that software tools incorrectly flip it to one, invalidating a possibly
valid stream.
Relax this restriction, by failing only when AV_EF_BITSTREAM is set.
This behaviour is similar to aac decoders in Firefox and Quicktime.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* commit 'd615187f74ddf3413778a8b5b7ae17255b0df88e':
aacdec: Support for ER AAC ELD 480.
Conflicts:
libavcodec/aacdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0ee2573347ecdb9cb5656001f7201d819eec16d8':
aacdec: Support for ER AAC in LATM
Conflicts:
libavcodec/aacdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '77ab341c0c6cdf2bd437bb48d429e797d1e60da2':
aacdec: add default case in channel layout
Conflicts:
libavcodec/aacdec.c
Note, the default case is currently unreachable
See: a48b890392
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '831a1180785a786272cdcefb71566a770bfb879e':
Update dsputil- and SIMD-related comments to match reality more closely
Conflicts:
libavcodec/x86/hpeldsp.asm
libavutil/arm/float_dsp_init_arm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AAC LOAS can have new audio config objects in the stream itself.
Make sure the decoder reconfigures itself when the first one arrives
midstream.
Bug-Id: 644
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
AVFrame.sample_rate is set in ff_get_buffer, but aacdec calls
ff_get_buffer before the samplerate is known. So it needs to be
set again before returning the frame.
Fixes out of array read
Fixes: asan_static-oob_1efed25_1887_cov_2013541199_HeyYa_RA10_AAC_192K_30s.rm
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b2212dec0f011893ec68eecaa990170fa24050d7':
aac: Fix TNS decoding for the 512 sample window family.
also temporarily disable fate-aac-er_ad6000np_44_ep0 as this commit
causes a mismatch with the reference pcm file
The test will be reenabled after all fixes and with a new pcm reference
Merged-by: Michael Niedermayer <michaelni@gmx.at>
AAC specification has 7.1(wide) as a default layout for 8-channel
streams (channel config 7). However, at least Nero AAC encoder encodes
non-wide 7.1 streams using the default channel config 7, mapping the
side channels of the original audio stream to the second
AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD decodes
the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
the incorrect streams as if they were correct (and as the encoder
intended).
FFmpeg currently decodes such files by-the-spec, i.e. after decoding the
original front pair will be in AV_CH_FRONT_x_OF_CENTER and the original
side pair will be in AV_CH_FRONT_x.
As actual intended 7.1(wide) streams are very rare while misencoded 7.1
files actually exist in the wild, default to assuming a 7.1 layout was
intended unless in strict mode.
Fixes playback of e.g. 8_Channel_ID.m4a in samples.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
