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avfilter/af_earwax: fix filter behavior

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Previous filter output was incorrect. New one actually follows
graph in comments described on side of filter taps.
This commit is contained in:
Paul B Mahol
2020-12-07 21:09:08 +01:00
parent c369ceb244
commit f41de0436c
3 changed files with 118 additions and 50 deletions
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libavfilter/af_earwax.c
+97 -29
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@@ -34,9 +34,9 @@
#include "audio.h"
#include "formats.h"
#define NUMTAPS 64
#define NUMTAPS 32
static const int8_t filt[NUMTAPS] = {
static const int8_t filt[NUMTAPS * 2] = {
/* 30° 330° */
4, -6, /* 32 tap stereo FIR filter. */
4, -11, /* One side filters as if the */
@@ -72,7 +72,10 @@ static const int8_t filt[NUMTAPS] = {
4, 0};
typedef struct EarwaxContext {
int16_t taps[NUMTAPS * 2];
int16_t filter[2][NUMTAPS];
int16_t taps[4][NUMTAPS * 2];
AVFrame *frame[2];
} EarwaxContext;
static int query_formats(AVFilterContext *ctx)
@@ -83,7 +86,7 @@ static int query_formats(AVFilterContext *ctx)
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16 )) < 0 ||
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO )) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
@@ -94,7 +97,8 @@ static int query_formats(AVFilterContext *ctx)
}
//FIXME: replace with DSPContext.scalarproduct_int16
static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, int16_t *out)
static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
const int16_t *filt, int16_t *out)
{
int32_t sample;
int16_t j;
@@ -103,7 +107,7 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in
sample = 0;
for (j = 0; j < NUMTAPS; j++)
sample += in[j] * filt[j];
*out = av_clip_int16(sample >> 6);
*out = av_clip_int16(sample >> 7);
out++;
in++;
}
@@ -111,40 +115,102 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in
return out;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
static int config_input(AVFilterLink *inlink)
{
AVFilterLink *outlink = inlink->dst->outputs[0];
int16_t *taps, *endin, *in, *out;
AVFrame *outsamples = ff_get_audio_buffer(outlink, insamples->nb_samples);
EarwaxContext *s = inlink->dst->priv;
for (int i = 0; i < NUMTAPS; i++) {
s->filter[0][i] = filt[i * 2];
s->filter[1][i] = filt[i * 2 + 1];
}
return 0;
}
static void convolve(AVFilterContext *ctx, AVFrame *in,
int input_ch, int output_ch,
int filter_ch, int tap_ch)
{
EarwaxContext *s = ctx->priv;
int16_t *taps, *endin, *dst, *src;
int len;
if (!outsamples) {
av_frame_free(&insamples);
return AVERROR(ENOMEM);
}
av_frame_copy_props(outsamples, insamples);
taps = s->taps[tap_ch];
dst = (int16_t *)s->frame[input_ch]->data[output_ch];
src = (int16_t *)in->data[input_ch];
taps = ((EarwaxContext *)inlink->dst->priv)->taps;
out = (int16_t *)outsamples->data[0];
in = (int16_t *)insamples ->data[0];
len = FFMIN(NUMTAPS, 2*insamples->nb_samples);
len = FFMIN(NUMTAPS, in->nb_samples);
// copy part of new input and process with saved input
memcpy(taps+NUMTAPS, in, len * sizeof(*taps));
out = scalarproduct(taps, taps + len, out);
memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
// process current input
if (2*insamples->nb_samples >= NUMTAPS ){
endin = in + insamples->nb_samples * 2 - NUMTAPS;
scalarproduct(in, endin, out);
if (2*in->nb_samples >= NUMTAPS ){
endin = src + in->nb_samples - NUMTAPS;
scalarproduct(src, endin, s->filter[filter_ch], dst);
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
} else
memmove(taps, taps + 2*insamples->nb_samples, NUMTAPS * sizeof(*taps));
} else {
memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
}
}
av_frame_free(&insamples);
return ff_filter_frame(outlink, outsamples);
static void mix(AVFilterContext *ctx, AVFrame *out,
int output_ch, int f0, int f1, int i0, int i1)
{
EarwaxContext *s = ctx->priv;
const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
int16_t *dst = (int16_t *)out->data[output_ch];
for (int n = 0; n < out->nb_samples; n++)
dst[n] = av_clip_int16(srcl[n] + srcr[n]);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
EarwaxContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
for (int ch = 0; ch < 2; ch++) {
if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
av_frame_free(&s->frame[ch]);
s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
if (!s->frame[ch]) {
av_frame_free(&in);
av_frame_free(&out);
return AVERROR(ENOMEM);
}
}
}
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
convolve(ctx, in, 0, 0, 0, 0);
convolve(ctx, in, 0, 1, 1, 1);
convolve(ctx, in, 1, 0, 0, 2);
convolve(ctx, in, 1, 1, 1, 3);
mix(ctx, out, 0, 0, 1, 1, 0);
mix(ctx, out, 1, 0, 1, 0, 1);
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
EarwaxContext *s = ctx->priv;
av_frame_free(&s->frame[0]);
av_frame_free(&s->frame[1]);
}
static const AVFilterPad earwax_inputs[] = {
@@ -152,6 +218,7 @@ static const AVFilterPad earwax_inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
{ NULL }
};
@@ -169,6 +236,7 @@ AVFilter ff_af_earwax = {
.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
.query_formats = query_formats,
.priv_size = sizeof(EarwaxContext),
.uninit = uninit,
.inputs = earwax_inputs,
.outputs = earwax_outputs,
};