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authorPaul B Mahol <onemda@gmail.com>2022-12-12 01:55:06 +0100
committerPaul B Mahol <onemda@gmail.com>2022-12-12 20:29:07 +0100
commit91abbb9d024817f6e4cf064a0db21c4979912840 (patch)
tree20128d9f7b71ce0a9a0d0da33632be52797fee42 /libavfilter/afir_template.c
parentc3e20f78b05d78ed1d214ba40fa5133c16af6415 (diff)
downloadffmpeg-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.c3
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--;