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author | Michael Niedermayer <michaelni@gmx.at> | 2007-03-22 14:27:16 +0000 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2007-03-22 14:27:16 +0000 |
commit | 8ec04d344ad6c832f1c060e7b62148bd0dc29ed1 (patch) | |
tree | 389b73b208eca5e8890a0c957adcaaca540ac7e5 /libavcodec | |
parent | 4ba67eeed52e84465c0bcba510fcc7e496d203c4 (diff) | |
download | ffmpeg-8ec04d344ad6c832f1c060e7b62148bd0dc29ed1.tar.gz |
set more reasonable cutoff for audio rate conversion
Originally committed as revision 8486 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/resample.c | 3 |
1 files changed, 2 insertions, 1 deletions
diff --git a/libavcodec/resample.c b/libavcodec/resample.c index 043e812c8f..ea5c6d61c8 100644 --- a/libavcodec/resample.c +++ b/libavcodec/resample.c @@ -161,7 +161,8 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels, if(s->filter_channels>2) s->filter_channels = 2; - s->resample_context= av_resample_init(output_rate, input_rate, 16, 10, 0, 1.0); +#define TAPS 16 + s->resample_context= av_resample_init(output_rate, input_rate, TAPS, 10, 0, 0.8); return s; } |