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>
