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ffmpeg.wasm-core
Commit bb54c0ae authored by Paul B Mahol's avatar Paul B Mahol
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avfilter/af_afftdn: switch to activate

parent bdfd2e3c
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Showing with 116 additions and 87 deletions
libavfilter/af_afftdn.c
View file @ bb54c0ae
...@@ -28,6 +28,7 @@ ...@@ -28,6 +28,7 @@
#include "avfilter.h" #include "avfilter.h"
#include "audio.h" #include "audio.h"
#include "formats.h" #include "formats.h"
#include "filters.h"
#define C (M_LN10 * 0.1) #define C (M_LN10 * 0.1)
#define RATIO 0.98 #define RATIO 0.98
...@@ -1153,7 +1154,7 @@ static void get_auto_noise_levels(AudioFFTDeNoiseContext *s, ...@@ -1153,7 +1154,7 @@ static void get_auto_noise_levels(AudioFFTDeNoiseContext *s,
} }
} }
static int filter_frame(AVFilterLink *inlink, AVFrame *frame) static int output_frame(AVFilterLink *inlink)
{ {
AVFilterContext *ctx = inlink->dst; AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0]; AVFilterLink *outlink = ctx->outputs[0];
...@@ -1162,117 +1163,145 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame) ...@@ -1162,117 +1163,145 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
ThreadData td; ThreadData td;
int ret = 0; int ret = 0;
if (s->pts == AV_NOPTS_VALUE) if (!in) {
s->pts = frame->pts; in = ff_get_audio_buffer(outlink, s->window_length);
if (!in)
return AVERROR(ENOMEM);
}
ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples); ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->window_length);
av_frame_free(&frame);
if (ret < 0) if (ret < 0)
return ret; return ret;
while (av_audio_fifo_size(s->fifo) >= s->window_length) { if (s->track_noise) {
if (!in) { for (int ch = 0; ch < inlink->channels; ch++) {
in = ff_get_audio_buffer(outlink, s->window_length); DeNoiseChannel *dnch = &s->dnch[ch];
if (!in) double levels[15];
return AVERROR(ENOMEM);
}
ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->window_length); get_auto_noise_levels(s, dnch, levels);
if (ret < 0) set_noise_profile(s, dnch, levels, 0);
break; }
if (s->track_noise) { if (s->noise_floor != s->last_noise_floor)
for (int ch = 0; ch < inlink->channels; ch++) { set_parameters(s);
DeNoiseChannel *dnch = &s->dnch[ch]; }
double levels[15];
get_auto_noise_levels(s, dnch, levels); if (s->sample_noise_start) {
set_noise_profile(s, dnch, levels, 0); for (int ch = 0; ch < inlink->channels; ch++) {
} DeNoiseChannel *dnch = &s->dnch[ch];
if (s->noise_floor != s->last_noise_floor) init_sample_noise(dnch);
set_parameters(s);
} }
s->sample_noise_start = 0;
s->sample_noise = 1;
}
if (s->sample_noise_start) { if (s->sample_noise) {
for (int ch = 0; ch < inlink->channels; ch++) { for (int ch = 0; ch < inlink->channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch]; DeNoiseChannel *dnch = &s->dnch[ch];
init_sample_noise(dnch); sample_noise_block(s, dnch, in, ch);
}
s->sample_noise_start = 0;
s->sample_noise = 1;
} }
}
if (s->sample_noise) { if (s->sample_noise_end) {
for (int ch = 0; ch < inlink->channels; ch++) { for (int ch = 0; ch < inlink->channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch]; DeNoiseChannel *dnch = &s->dnch[ch];
double sample_noise[15];
sample_noise_block(s, dnch, in, ch); finish_sample_noise(s, dnch, sample_noise);
} set_noise_profile(s, dnch, sample_noise, 1);
set_band_parameters(s, dnch);
} }
s->sample_noise = 0;
s->sample_noise_end = 0;
}
if (s->sample_noise_end) { s->block_count++;
for (int ch = 0; ch < inlink->channels; ch++) { td.in = in;
DeNoiseChannel *dnch = &s->dnch[ch]; ctx->internal->execute(ctx, filter_channel, &td, NULL,
double sample_noise[15]; FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
finish_sample_noise(s, dnch, sample_noise);
set_noise_profile(s, dnch, sample_noise, 1);
set_band_parameters(s, dnch);
}
s->sample_noise = 0;
s->sample_noise_end = 0;
}
s->block_count++; out = ff_get_audio_buffer(outlink, s->sample_advance);
td.in = in; if (!out) {
ctx->internal->execute(ctx, filter_channel, &td, NULL, ret = AVERROR(ENOMEM);
FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx))); goto end;
}
out = ff_get_audio_buffer(outlink, s->sample_advance); for (int ch = 0; ch < inlink->channels; ch++) {
if (!out) { DeNoiseChannel *dnch = &s->dnch[ch];
ret = AVERROR(ENOMEM); double *src = dnch->out_samples;
float *orig = (float *)in->extended_data[ch];
float *dst = (float *)out->extended_data[ch];
switch (s->output_mode) {
case IN_MODE:
for (int m = 0; m < s->sample_advance; m++)
dst[m] = orig[m];
break;
case OUT_MODE:
for (int m = 0; m < s->sample_advance; m++)
dst[m] = src[m];
break;
case NOISE_MODE:
for (int m = 0; m < s->sample_advance; m++)
dst[m] = orig[m] - src[m];
break; break;
default:
return AVERROR_BUG;
} }
memmove(src, src + s->sample_advance, (s->window_length - s->sample_advance) * sizeof(*src));
memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src));
}
for (int ch = 0; ch < inlink->channels; ch++) { av_audio_fifo_drain(s->fifo, s->sample_advance);
DeNoiseChannel *dnch = &s->dnch[ch];
double *src = dnch->out_samples; out->pts = s->pts;
float *orig = (float *)in->extended_data[ch]; ret = ff_filter_frame(outlink, out);
float *dst = (float *)out->extended_data[ch]; if (ret < 0)
goto end;
switch (s->output_mode) { s->pts += s->sample_advance;
case IN_MODE: end:
for (int m = 0; m < s->sample_advance; m++) av_frame_free(&in);
dst[m] = orig[m];
break;
case OUT_MODE:
for (int m = 0; m < s->sample_advance; m++)
dst[m] = src[m];
break;
case NOISE_MODE:
for (int m = 0; m < s->sample_advance; m++)
dst[m] = orig[m] - src[m];
break;
default:
return AVERROR_BUG;
}
memmove(src, src + s->sample_advance, (s->window_length - s->sample_advance) * sizeof(*src));
memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src));
}
av_audio_fifo_drain(s->fifo, s->sample_advance); return ret;
}
out->pts = s->pts; static int activate(AVFilterContext *ctx)
ret = ff_filter_frame(outlink, out); {
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioFFTDeNoiseContext *s = ctx->priv;
AVFrame *frame = NULL;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_frame(inlink, &frame);
if (ret < 0)
return ret;
if (ret > 0) {
if (s->pts == AV_NOPTS_VALUE)
s->pts = frame->pts;
ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
av_frame_free(&frame);
if (ret < 0) if (ret < 0)
break; return ret;
s->pts += s->sample_advance;
} }
av_frame_free(&in);
return ret; if (av_audio_fifo_size(s->fifo) >= s->window_length)
return output_frame(inlink);
FF_FILTER_FORWARD_STATUS(inlink, outlink);
if (ff_outlink_frame_wanted(outlink) &&
av_audio_fifo_size(s->fifo) < s->window_length) {
ff_inlink_request_frame(inlink);
return 0;
}
return FFERROR_NOT_READY;
} }
static av_cold void uninit(AVFilterContext *ctx) static av_cold void uninit(AVFilterContext *ctx)
...@@ -1393,7 +1422,6 @@ static const AVFilterPad inputs[] = { ...@@ -1393,7 +1422,6 @@ static const AVFilterPad inputs[] = {
{ {
.name = "default", .name = "default",
.type = AVMEDIA_TYPE_AUDIO, .type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input, .config_props = config_input,
}, },
{ NULL } { NULL }
...@@ -1413,6 +1441,7 @@ AVFilter ff_af_afftdn = { ...@@ -1413,6 +1441,7 @@ AVFilter ff_af_afftdn = {
.query_formats = query_formats, .query_formats = query_formats,
.priv_size = sizeof(AudioFFTDeNoiseContext), .priv_size = sizeof(AudioFFTDeNoiseContext),
.priv_class = &afftdn_class, .priv_class = &afftdn_class,
.activate = activate,
.uninit = uninit, .uninit = uninit,
.inputs = inputs, .inputs = inputs,
.outputs = outputs, .outputs = outputs,
......
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