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avfilter/af_afftdn: stop using fifo and rewritting pts

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release/5.1
Paul B Mahol 3 years ago
parent c1735bb139
commit 644b6ed3ff
1 changed files with 29 additions and 49 deletions
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  1. 78
      libavfilter/af_afftdn.c

78
libavfilter/af_afftdn.c
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@ -20,7 +20,6 @@
#include <float.h>
#include "libavutil/audio_fifo.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
@ -101,7 +100,6 @@ typedef struct AudioFFTDeNoiseContext {
float last_noise_balance;
int64_t block_count;
int64_t pts;
int channels;
int sample_noise;
int sample_noise_start;
@ -124,6 +122,8 @@ typedef struct AudioFFTDeNoiseContext {
DeNoiseChannel *dnch;
AVFrame *winframe;
double max_gain;
double max_var;
double gain_scale;
@ -138,8 +138,6 @@ typedef struct AudioFFTDeNoiseContext {
double vector_b[5];
double matrix_b[75];
double matrix_c[75];
AVAudioFifo *fifo;
} AudioFFTDeNoiseContext;
#define OFFSET(x) offsetof(AudioFFTDeNoiseContext, x)
@ -620,7 +618,6 @@ static int config_input(AVFilterLink *inlink)
if (!s->dnch)
return AVERROR(ENOMEM);
s->pts = AV_NOPTS_VALUE;
s->channels = inlink->channels;
s->sample_rate = inlink->sample_rate;
s->sample_advance = s->sample_rate / 80;
@ -831,6 +828,10 @@ static int config_input(AVFilterLink *inlink)
}
}
s->winframe = ff_get_audio_buffer(inlink, s->window_length);
if (!s->winframe)
return AVERROR(ENOMEM);
wscale = sqrt(16.0 / (9.0 * s->fft_length));
sum = 0.0;
for (int i = 0; i < s->window_length; i++) {
@ -857,10 +858,6 @@ static int config_input(AVFilterLink *inlink)
}
s->noise_band_count = s->noise_band_edge[16];
s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->fft_length);
if (!s->fifo)
return AVERROR(ENOMEM);
return 0;
}
@ -1160,23 +1157,24 @@ static void get_auto_noise_levels(AudioFFTDeNoiseContext *s,
}
}
static int output_frame(AVFilterLink *inlink)
static int output_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioFFTDeNoiseContext *s = ctx->priv;
const int output_mode = ctx->is_disabled ? IN_MODE : s->output_mode;
AVFrame *out = NULL, *in = NULL;
const int offset = s->window_length - s->sample_advance;
AVFrame *out = NULL;
ThreadData td;
int ret = 0;
in = ff_get_audio_buffer(outlink, s->window_length);
if (!in)
return AVERROR(ENOMEM);
for (int ch = 0; ch < s->channels; ch++) {
float *src = (float *)s->winframe->extended_data[ch];
ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, s->window_length);
if (ret < 0)
goto end;
memmove(src, &src[s->sample_advance], offset * sizeof(float));
memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
memset(&src[offset + in->nb_samples], 0, (s->sample_advance - in->nb_samples) * sizeof(float));
}
if (s->track_noise) {
for (int ch = 0; ch < inlink->channels; ch++) {
@ -1205,7 +1203,7 @@ static int output_frame(AVFilterLink *inlink)
for (int ch = 0; ch < inlink->channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
sample_noise_block(s, dnch, in, ch);
sample_noise_block(s, dnch, s->winframe, ch);
}
}
@ -1223,11 +1221,11 @@ static int output_frame(AVFilterLink *inlink)
}
s->block_count++;
td.in = in;
td.in = s->winframe;
ff_filter_execute(ctx, filter_channel, &td, NULL,
FFMIN(outlink->channels, ff_filter_get_nb_threads(ctx)));
out = ff_get_audio_buffer(outlink, s->sample_advance);
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
ret = AVERROR(ENOMEM);
goto end;
@ -1236,20 +1234,20 @@ static int output_frame(AVFilterLink *inlink)
for (int ch = 0; ch < inlink->channels; ch++) {
DeNoiseChannel *dnch = &s->dnch[ch];
double *src = dnch->out_samples;
float *orig = (float *)in->extended_data[ch];
const float *orig = (const float *)s->winframe->extended_data[ch];
float *dst = (float *)out->extended_data[ch];
switch (output_mode) {
case IN_MODE:
for (int m = 0; m < s->sample_advance; m++)
for (int m = 0; m < out->nb_samples; m++)
dst[m] = orig[m];
break;
case OUT_MODE:
for (int m = 0; m < s->sample_advance; m++)
for (int m = 0; m < out->nb_samples; m++)
dst[m] = src[m];
break;
case NOISE_MODE:
for (int m = 0; m < s->sample_advance; m++)
for (int m = 0; m < out->nb_samples; m++)
dst[m] = orig[m] - src[m];
break;
default:
@ -1261,13 +1259,11 @@ static int output_frame(AVFilterLink *inlink)
memset(src + (s->window_length - s->sample_advance), 0, s->sample_advance * sizeof(*src));
}
av_audio_fifo_drain(s->fifo, s->sample_advance);
out->pts = in->pts;
out->pts = s->pts;
ret = ff_filter_frame(outlink, out);
if (ret < 0)
goto end;
s->pts += av_rescale_q(s->sample_advance, (AVRational){1, outlink->sample_rate}, outlink->time_base);
end:
av_frame_free(&in);
@ -1279,34 +1275,19 @@ static int activate(AVFilterContext *ctx)
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioFFTDeNoiseContext *s = ctx->priv;
AVFrame *frame = NULL;
AVFrame *in = NULL;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_frame(inlink, &frame);
ret = ff_inlink_consume_samples(inlink, s->sample_advance, s->sample_advance, &in);
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)
return ret;
}
if (av_audio_fifo_size(s->fifo) >= s->window_length)
return output_frame(inlink);
if (ret > 0)
return output_frame(inlink, in);
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;
}
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
@ -1319,6 +1300,7 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->bin2band);
av_freep(&s->band_alpha);
av_freep(&s->band_beta);
av_frame_free(&s->winframe);
if (s->dnch) {
for (int ch = 0; ch < s->channels; ch++) {
@ -1343,8 +1325,6 @@ static av_cold void uninit(AVFilterContext *ctx)
}
av_freep(&s->dnch);
}
av_audio_fifo_free(s->fifo);
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,

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