Allows to select a codec (encoder or decoder) only if it supports a
specific profile.
Adds ff_AMediaCodecProfile_getProfileFromAVCodecContext to convert an
AVCodecContext profile to a MediaCodec profile. It only supports H264
for now.
The codepath using MediaCodecList.findDecoderForFormat() (Android >= 5.0)
has been dropped as this method does not allow to select a decoder
compatible with a specific profile.
Signed-off-by: Benjamin Steffes <benjaminst123@gmail.com>
(comment by ronald)
prevent the theoretical case where the container type (int)
would be 64 bit on some platforms, which would waste some space
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The stat struct is defined to stati64, which requires using the appropriate wstati/stati functions as well.
Fixes a whole bunch of compiler warnings as well as build breakage with the decklink avdevice.
Fixes trac #5640
We still only support one single layer though, but this allows
receiving streams that have this structure present even for
single layer streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
Docs clearly states that av_write_trailer should only be called if
avformat_write_header was successful, therefore we have to deinit if we return
failure.
Signed-off-by: Marton Balint <cus@passwd.hu>
When checking pix_fmt mapping, some bitstreams are mapped to an
incorrect pix_fmt instead of being rejected (ENOSYS).
Actually, such bitstreams are not supported (FFmpeg encoder does not
produce such bitstream, such bitstream may come only from another
encoder for the moment).
- JPEG 2000 RCT 11/13/15/16 bit depths are mapped to a 8-bit FFmpeg
pix_fmt (e.g. bgr0), which is not expected.
- JPEG 2000 RCT 9/10/12/14 bit depths with alpha are mapped to a
FFmpeg pix_fmt without alpha (e.g. AV_PIX_FMT_GBRP9 for 9-bit with
alpha), which is not expected.
The order for choosing the pix_fmt is changed to the one used by YCbCr
selection (<=8 bit first).
" && !f->transparency" is added to the other lines.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
this allow a filter to be written like this:
aformat =
sample_fmts = fltp|flt:
sample_rates = 44100|44800
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>
