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author | Paul B Mahol <onemda@gmail.com> | 2022-12-12 01:55:06 +0100 |
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committer | Paul B Mahol <onemda@gmail.com> | 2022-12-12 20:29:07 +0100 |
commit | 91abbb9d024817f6e4cf064a0db21c4979912840 (patch) | |
tree | 20128d9f7b71ce0a9a0d0da33632be52797fee42 /libavfilter/afir_template.c | |
parent | c3e20f78b05d78ed1d214ba40fa5133c16af6415 (diff) | |
download | ffmpeg-91abbb9d024817f6e4cf064a0db21c4979912840.tar.gz |
avfilter/af_afir: reduce output gain with default parameters
It was unreasonably high. Also change scaling to reduce
rare quantization errors.
Diffstat (limited to 'libavfilter/afir_template.c')
-rw-r--r-- | libavfilter/afir_template.c | 3 |
1 files changed, 1 insertions, 2 deletions
diff --git a/libavfilter/afir_template.c b/libavfilter/afir_template.c index 45055bba87..712f4b7417 100644 --- a/libavfilter/afir_template.c +++ b/libavfilter/afir_template.c @@ -159,7 +159,7 @@ static void fn(convert_channels)(AVFilterContext *ctx, AudioFIRContext *s) memset(blockin, 0, sizeof(*blockin) * seg->fft_length); memcpy(blockin, time + toffset, size * sizeof(*blockin)); - seg->tx_fn(seg->tx[0], blockout, blockin, sizeof(ftype)); + seg->ctx_fn(seg->ctx, blockout, blockin, sizeof(ftype)); for (int n = 0; n < seg->part_size + 1; n++) { coeff[coffset + n].re = blockout[2 * n]; @@ -316,7 +316,6 @@ static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int offse #else s->afirdsp.dcmul_add(sumin, blockout, (const ftype *)coeff, part_size); #endif - if (j == 0) j = seg->nb_partitions; j--; |