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avfilter/af_afir: reduce output gain with default parameters

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It was unreasonably high. Also change scaling to reduce
rare quantization errors.
This commit is contained in:
Paul B Mahol 2022-12-12 01:55:06 +01:00
parent c3e20f78b0
commit 91abbb9d02
3 changed files with 28 additions and 20 deletions
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41
libavfilter/af_afir.c
View File

@ -160,6 +160,8 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
{
AudioFIRContext *s = ctx->priv;
const size_t cpu_align = av_cpu_max_align();
union { double d; float f; } cscale, scale, iscale;
enum AVTXType tx_type;
int ret;
seg->tx = av_calloc(ctx->inputs[0]->ch_layout.nb_channels, sizeof(*seg->tx));
@ -167,7 +169,7 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
if (!seg->tx || !seg->itx)
return AVERROR(ENOMEM);
seg->fft_length = part_size * 2 + 1;
seg->fft_length = part_size * 2 + 2;
seg->part_size = part_size;
seg->block_size = FFALIGN(seg->fft_length, cpu_align);
seg->coeff_size = FFALIGN(seg->part_size + 1, cpu_align);
@ -180,22 +182,27 @@ static int init_segment(AVFilterContext *ctx, AudioFIRSegment *seg,
if (!seg->part_index || !seg->output_offset)
return AVERROR(ENOMEM);
for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 1; ch++) {
union { double d; float f; } scale, iscale;
enum AVTXType tx_type;
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
cscale.f = 1.f;
scale.f = 1.f / sqrtf(2.f * part_size);
iscale.f = 1.f / sqrtf(2.f * part_size);
tx_type = AV_TX_FLOAT_RDFT;
break;
case AV_SAMPLE_FMT_DBLP:
cscale.d = 1.0;
scale.d = 1.0 / sqrt(2.0 * part_size);
iscale.d = 1.0 / sqrt(2.0 * part_size);
tx_type = AV_TX_DOUBLE_RDFT;
break;
}
switch (s->format) {
case AV_SAMPLE_FMT_FLTP:
scale.f = 1.f;
iscale.f = 1.f / part_size;
tx_type = AV_TX_FLOAT_RDFT;
break;
case AV_SAMPLE_FMT_DBLP:
scale.d = 1.0;
iscale.d = 1.0 / part_size;
tx_type = AV_TX_DOUBLE_RDFT;
break;
}
ret = av_tx_init(&seg->ctx, &seg->ctx_fn, tx_type,
0, 2 * part_size, &cscale, 0);
if (ret < 0)
return ret;
for (int ch = 0; ch < ctx->inputs[0]->ch_layout.nb_channels && part_size >= 1; ch++) {
ret = av_tx_init(&seg->tx[ch], &seg->tx_fn, tx_type,
0, 2 * part_size, &scale, 0);
if (ret < 0)
@ -224,6 +231,8 @@ static void uninit_segment(AVFilterContext *ctx, AudioFIRSegment *seg)
{
AudioFIRContext *s = ctx->priv;
av_tx_uninit(&seg->ctx);
if (seg->tx) {
for (int ch = 0; ch < s->nb_channels; ch++) {
av_tx_uninit(&seg->tx[ch]);

4
libavfilter/af_afir.h
View File

@ -49,8 +49,8 @@ typedef struct AudioFIRSegment {
AVFrame *input;
AVFrame *output;
AVTXContext **tx, **itx;
av_tx_fn tx_fn, itx_fn;
AVTXContext *ctx, **tx, **itx;
av_tx_fn ctx_fn, tx_fn, itx_fn;
} AudioFIRSegment;
typedef struct AudioFIRContext {

3
libavfilter/afir_template.c
View File

@ -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--;