If memory allocation fails, ERROR(ENOMEM) '-12' will be returned.
When resample() is done first, the negative size param would cause buffer-overflow and SEGV in swri_rematrix().
When swri_rematrix() is run first, resample() would not cause an error but Err num as a wrong parameter passing.
Err num should be returned immediately. And remove an unneeded term from an assert.
coredump info:
#0 0x499517 in posix_memalign (/home/r1/ffmpeg/ffmpeg_4.4.1+0x499517)
#1 0x6c1f0b4 in av_malloc /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavutil/mem.c:86:9
#2 0x6c208fe in av_mallocz /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavutil/mem.c:239:17
#3 0x6c207ad in av_mallocz_array /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavutil/mem.c:195:12
#4 0x654b2e5 in swri_realloc_audio /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libswresample/swresample.c:418:14
#5 0x654f9a1 in swr_convert_internal /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libswresample/swresample.c:601:17
#6 0x654d2c0 in swr_convert /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libswresample/swresample.c:766:19
#7 0x186cf56 in flush_frame /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/af_aresample.c:251:13
#8 0x186a454 in request_frame /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/af_aresample.c:288:20
#9 0x787d9c in ff_request_frame_to_filter /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/avfilter.c:459:15
#10 0x7877f1 in forward_status_change /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/avfilter.c:1257:19
#11 0x77ed7e in ff_filter_activate_default /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/avfilter.c:1288:20
#12 0x77e4e1 in ff_filter_activate /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/avfilter.c:1441:11
#13 0x793b3f in ff_filter_graph_run_once /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/avfiltergraph.c:1403:12
#14 0x7a7bee in get_frame_internal /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/buffersink.c:131:19
#15 0x7a7287 in av_buffersink_get_frame_flags /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/buffersink.c:142:12
#16 0x792888 in avfilter_graph_request_oldest /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/libavfilter/avfiltergraph.c:1356:17
#17 0x5d07df in transcode_from_filter /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/fftools/ffmpeg.c:4639:11
#18 0x59e557 in transcode_step /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/fftools/ffmpeg.c:4729:20
#19 0x593970 in transcode /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/fftools/ffmpeg.c:4805:15
#20 0x58f7a4 in main /home/r1/ffmpeg/ffmpeg-4.4.1/build/src/fftools/ffmpeg.c:5010:9
#21 0x7f6fd2dee0b2 in __libc_start_main /build/glibc-eX1tMB/glibc-2.31/csu/../csu/libc-start.c:308:16
SUMMARY: AddressSanitizer: negative-size-param (/home/r1/ffmpeg/ffmpeg_4.4.1+0x497e67) in __asan_memcpy
Reported-by: TOTE Robot <oslab@tsinghua.edu.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
SWR_CH_MAX is internal only and the arrays are therefore not required
to have that many elements (and they typically don't do it). So remove
this potentially confusing hint.
(Newer versions of GCC emit -Warray-parameter= warnings for this,
because the definition with explicit size differs from the declaration
(which leaves the size unspecified); this is IMO a false-positive,
because definition and declaration didn't conflict, but anyway it is
fixed by this commit.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Instead only include libavutil/version.h; including avutil.h is a
remnant from the time in which the version was in it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
These inclusions are not necessary, as cpu.h is already included
wherever it is needed (via direct inclusion or via the arch-specific
headers).
Also remove other unnecessary cpu.h inclusions from ordinary
non-headers.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
It is not used here at all; instead, add it where it is used without
including it or any of the arch-specific CPU headers.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Some files currently rely on libavutil/cpu.h to include it for them;
yet said file won't use include it any more after the currently
deprecated functions are removed, so include attributes.h directly.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This is as far as 22.2 follows the same channel order as
WaveFormatExtensible's channel mask (and the AV_CH_* defines).
After LFE2 the side channels would follow, but that offset of
one stops us from utilizing them without further tweaks.
This change was verified by using swresample to downmix to 5.1,
and then feeding that to WASAPI.
Only this sub-set of channels actually follows the bit mask order
in the official 22.2 channel mapping. Additionally, the 5.1 channels
are there for backwards compatibility with the previous system.
This enables the utilization of 22.2 content until a proper down/up
matrix is added into swresample.
These bits are utilized by channel layouts such as 22.2. If those
are dropped, the returned channel layout is no longer a match
against the AV_CH_LAYOUT define when returned from this function.
Rematrixing supports up to 64 channels. However, there is only a limited number of channel layouts defined. Since the in/out channel count is currently obtained from the channel layout, for undefined layouts (e.g. for 9, 10, 11 channels etc.) the rematrixing fails.
This patch changes rematrix init methods to use in (used) and out channel count directly instead of computing it from channel layout.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Branch to global symbol results in reference to PLT, and when compiling
for THUMB-2 - in a R_ARM_THM_JUMP19 relocation. Some linkers don't
support this relocation (ld.gold), while others can end up truncating
the relocation to fit (ld.bfd).
Convert this branch through PLT into a direct branch that the assembler
can resolve locally.
See https://github.com/android-ndk/ndk/issues/337 for background.
The current workaround is to disable neon during gstreamer build,
which is not optimal and can be reverted after this patch:
41556c4157
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Removed +len1 in call to s->mix_2_1_f() as I found no logical explanation for it. After removal, problem was gone.
Signed-off-by: Hendrik Schreiber <hs@tagtraum.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Prefer direct in/out channel count values over channel layout, when
available. Fixes a pan filter bug (ticket #6790).
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
* commit '07a2b155949eb267cdfc7805f42c7b3375f9c7c5':
Bump major versions of all libraries
A few API deprecated ~2 years ago or more are also postponed here for
varying reasons.
FF_API_LOWRES:
Since this functionality depends on AVStream->codec, i figure the two can
be removed at the same time in the next bump or so.
FF_API_AVCTX_TIMEBASE:
Couldn't get this one to work. Not just libavcodec but apparently also
libavformat and ffmpeg.c expect AVCodecContext->time_base to be set for
decoding. Upon removal some tests report a different generic stream time
base (like 1/25), and others lose packet duration values. I guess it's
somehow tied to the AVStream->codec clusterfuck.
It can be dealt with alongside FF_API_LAVF_AVCTX in the next bump.
FF_API_OLD_FILTER_OPTS_ERROR:
This one is meant to remain after FF_API_OLD_FILTER_OPTS is removed.
Its purpose is displaying the corrected command line using the new syntax
as a suggestion as part of the error message.
Merged-by: James Almer <jamrial@gmail.com>
When 'out' is an AVFrame that does not have buffers preallocated,
swr_convert_frame tries to allocate buffers of the right size. However
in calculating this size it failed to check for whether 'in' is NULL
(requesting that swr's internal buffers are to be flushed).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '92db5083077a8b0f8e1050507671b456fd155125':
build: Generate pkg-config files from Make and not from configure
build: Store library version numbers in .version files
Includes cherry-picked commits 8a34f36593 and
ee164727dd to fix issues.
Changes were also made to retain support for raise_major and build_suffix.
Reviewed-by: ubitux
Merged-by: James Almer <jamrial@gmail.com>
* commit '11a9320de54759340531177c9f2b1e31e6112cc2':
build: Move build-system-related helper files to a separate subdirectory
"ffbuild" directory name is used instead of "avbuild".
Merged-by: Clément Bœsch <u@pkh.me>
use fltp when doing s32 -> s32 resampling
because s32p has no simd optimization
benchmark:
old 17.913s
new 7.584s (use fma3)
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
when set_compensation is called with zero sample_delta,
compensation does not happen (because dst_incr == ideal_dst_incr)
but compensation_distance is set
regression since 01ebb57c03
Found-by: wm4 <nfxjfg@googlemail.com>
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
so tsf option in aresample will have effect
previously tsf/internal_sample_format had no effect
fate is updated
s32p previously used fltp internally
dblp previously used fltp/dblp internally
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
except filter_length == 1
odd filter_length gives worse frequency response,
even when compared with shorter filter_length
also makes build_filter simpler
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
integrate it inside multiple_resample
allow some calculations to be performed outside loop
Suggested-by: Michael Niedermayer <michael@niedermayer.cc>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
This is faster 2871 -> 2189 cycles for int16 matrixbench -> 23456hz
Fixes a integer overflow in a artificial corner case
Fixes part of 668007-media
Found-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes undefined operation
Fixes part of 668007-media
Found-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
give very bad quality for soxr resampler.
linear_interp is intended for using linear interpolation
between filter bank so quality will be better.
i guess this is misunderstood as 'do not use filter bank,
but directly interpolate linearly between samples'.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
separate dsp.resample to dsp.resample_common and dsp.resample_linear
and choose to call faster resample_common even when linear_interp=on
when c->frac and c->dst_incr_mod are both zero
speed up resampling when exact_rational and linear_interp are both
enabled because exact_rational force c->frac and c->dst_incr_mod to
be zero when soft compensation does not happen
benchmark on exact_rational=on:linear_interp=on
old new
real 8.432s 5.097s
user 7.679s 4.989s
sys 0.125s 0.107s
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
high phase_count is only useful when dst_incr_mod is non zero
in other word, it is only useful on soft compensation
on init, it will build filter with low phase_count
but when soft compensation is enabled, rebuild filter
with high phase_count
this approach saves lots of memory
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
because exact_rational does not guarantee
that phase_count is even
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
give high quality resampling
as good as with linear_interp=on
as fast as without linear_interp=on
tested visually with ffplay
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000, showcqt=gamma=5"
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000:linear_interp=on, showcqt=gamma=5"
ffplay -f lavfi "aevalsrc='sin(10000*t*t)', aresample=osr=48000:exact_rational=on, showcqt=gamma=5"
slightly speed improvement
for fair comparison with -cpuflags 0
audio.wav is ~ 1 hour 44100 stereo 16bit wav file
ffmpeg -i audio.wav -af aresample=osr=48000 -f null -
old new
real 13.498s 13.121s
user 13.364s 12.987s
sys 0.131s 0.129s
linear_interp=on
old new
real 23.035s 23.050s
user 22.907s 22.917s
sys 0.119s 0.125s
exact_rational=on
real 12.418s
user 12.298s
sys 0.114s
possibility to decrease memory usage if soft compensation is ignored
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
This fixes the sum of the integer coefficients ending up summing to a value
larger than the value representing unity.
This issue occurs with qN0.dts when converting to stereo
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Yu Yang
Andreas Rheinhardt
Zhao Zhili
Anton Khirnov
Andreas Rheinhardt
Michael Niedermayer
Jan Ekström
Carl Eugen Hoyos
Paul B Mahol
Marcin Gorzel
Rahul Chaudhry
Hendrik Schreiber
Tobias Rapp
James Almer
hexpointer
Diego Biurrun
Clément Bœsch
Muhammad Faiz
Carl Eugen Hoyos
Matthieu Bouron
Vodyannikov Aleksandr
Timothy Gu
Matthieu Bouron
Clément Bœsch
Ganesh Ajjanagadde