[FFmpeg-cvslog] avfilter/af_afftfilt: switch to activate
Paul B Mahol
git at videolan.org
Wed May 8 16:29:38 EEST 2019
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed May 8 15:03:22 2019 +0200| [c539dd992cea8313ecb7565bb74645c81d7bf333] | committer: Paul B Mahol
avfilter/af_afftfilt: switch to activate
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c539dd992cea8313ecb7565bb74645c81d7bf333
---
libavfilter/af_afftfilt.c | 228 +++++++++++++++++++++++++---------------------
1 file changed, 126 insertions(+), 102 deletions(-)
diff --git a/libavfilter/af_afftfilt.c b/libavfilter/af_afftfilt.c
index 8518f08dc5..fcbebdde26 100644
--- a/libavfilter/af_afftfilt.c
+++ b/libavfilter/af_afftfilt.c
@@ -26,6 +26,7 @@
#include "libavcodec/avfft.h"
#include "libavutil/eval.h"
#include "audio.h"
+#include "filters.h"
#include "window_func.h"
typedef struct AFFTFiltContext {
@@ -46,7 +47,7 @@ typedef struct AFFTFiltContext {
int hop_size;
float overlap;
AVFrame *buffer;
- int start, end;
+ int eof;
int win_func;
float win_scale;
float *window_func_lut;
@@ -240,7 +241,7 @@ static int config_input(AVFilterLink *inlink)
return ret;
}
-static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
+static int filter_frame(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@@ -249,140 +250,163 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
const float f = 1. / s->win_scale;
double values[VAR_VARS_NB];
AVFrame *out, *in = NULL;
- int ch, n, ret, i, j, k;
- int start = s->start, end = s->end;
+ int ch, n, ret, i;
- if (s->pts == AV_NOPTS_VALUE)
- s->pts = frame->pts;
+ if (!in) {
+ in = ff_get_audio_buffer(outlink, window_size);
+ if (!in)
+ return AVERROR(ENOMEM);
+ }
- ret = av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
- av_frame_free(&frame);
+ ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, window_size);
if (ret < 0)
- return ret;
+ goto fail;
- while (av_audio_fifo_size(s->fifo) >= window_size) {
- if (!in) {
- in = ff_get_audio_buffer(outlink, window_size);
- if (!in)
- return AVERROR(ENOMEM);
+ for (ch = 0; ch < inlink->channels; ch++) {
+ const float *src = (float *)in->extended_data[ch];
+ FFTComplex *fft_data = s->fft_data[ch];
+
+ for (n = 0; n < in->nb_samples; n++) {
+ fft_data[n].re = src[n] * s->window_func_lut[n];
+ fft_data[n].im = 0;
}
- ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, window_size);
- if (ret < 0)
- break;
+ for (; n < window_size; n++) {
+ fft_data[n].re = 0;
+ fft_data[n].im = 0;
+ }
+ }
- for (ch = 0; ch < inlink->channels; ch++) {
- const float *src = (float *)in->extended_data[ch];
- FFTComplex *fft_data = s->fft_data[ch];
+ values[VAR_PTS] = s->pts;
+ values[VAR_SAMPLE_RATE] = inlink->sample_rate;
+ values[VAR_NBBINS] = window_size / 2;
+ values[VAR_CHANNELS] = inlink->channels;
- for (n = 0; n < in->nb_samples; n++) {
- fft_data[n].re = src[n] * s->window_func_lut[n];
- fft_data[n].im = 0;
- }
+ for (ch = 0; ch < inlink->channels; ch++) {
+ FFTComplex *fft_data = s->fft_data[ch];
- for (; n < window_size; n++) {
- fft_data[n].re = 0;
- fft_data[n].im = 0;
- }
- }
+ av_fft_permute(s->fft, fft_data);
+ av_fft_calc(s->fft, fft_data);
+ }
- values[VAR_PTS] = s->pts;
- values[VAR_SAMPLE_RATE] = inlink->sample_rate;
- values[VAR_NBBINS] = window_size / 2;
- values[VAR_CHANNELS] = inlink->channels;
+ for (ch = 0; ch < inlink->channels; ch++) {
+ FFTComplex *fft_data = s->fft_data[ch];
+ FFTComplex *fft_temp = s->fft_temp[ch];
+ float *buf = (float *)s->buffer->extended_data[ch];
+ int x;
+ values[VAR_CHANNEL] = ch;
- for (ch = 0; ch < inlink->channels; ch++) {
- FFTComplex *fft_data = s->fft_data[ch];
+ for (n = 0; n <= window_size / 2; n++) {
+ float fr, fi;
- av_fft_permute(s->fft, fft_data);
- av_fft_calc(s->fft, fft_data);
- }
+ values[VAR_BIN] = n;
+ values[VAR_REAL] = fft_data[n].re;
+ values[VAR_IMAG] = fft_data[n].im;
- for (ch = 0; ch < inlink->channels; ch++) {
- FFTComplex *fft_data = s->fft_data[ch];
- FFTComplex *fft_temp = s->fft_temp[ch];
- float *buf = (float *)s->buffer->extended_data[ch];
- int x;
- values[VAR_CHANNEL] = ch;
+ fr = av_expr_eval(s->real[ch], values, s);
+ fi = av_expr_eval(s->imag[ch], values, s);
- for (n = 0; n <= window_size / 2; n++) {
- float fr, fi;
+ fft_temp[n].re = fr;
+ fft_temp[n].im = fi;
+ }
- values[VAR_BIN] = n;
- values[VAR_REAL] = fft_data[n].re;
- values[VAR_IMAG] = fft_data[n].im;
+ for (n = window_size / 2 + 1, x = window_size / 2 - 1; n < window_size; n++, x--) {
+ fft_temp[n].re = fft_temp[x].re;
+ fft_temp[n].im = -fft_temp[x].im;
+ }
- fr = av_expr_eval(s->real[ch], values, s);
- fi = av_expr_eval(s->imag[ch], values, s);
+ av_fft_permute(s->ifft, fft_temp);
+ av_fft_calc(s->ifft, fft_temp);
- fft_temp[n].re = fr;
- fft_temp[n].im = fi;
- }
+ for (i = 0; i < window_size; i++) {
+ buf[i] += s->fft_temp[ch][i].re * f;
+ }
+ }
- for (n = window_size / 2 + 1, x = window_size / 2 - 1; n < window_size; n++, x--) {
- fft_temp[n].re = fft_temp[x].re;
- fft_temp[n].im = -fft_temp[x].im;
- }
+ out = ff_get_audio_buffer(outlink, s->hop_size);
+ if (!out) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
- av_fft_permute(s->ifft, fft_temp);
- av_fft_calc(s->ifft, fft_temp);
+ out->pts = s->pts;
+ s->pts += s->hop_size;
- start = s->start;
- end = s->end;
- k = end;
- for (i = 0, j = start; j < k && i < window_size; i++, j++) {
- buf[j] += s->fft_temp[ch][i].re * f;
- }
+ for (ch = 0; ch < inlink->channels; ch++) {
+ float *dst = (float *)out->extended_data[ch];
+ float *buf = (float *)s->buffer->extended_data[ch];
- for (; i < window_size; i++, j++) {
- buf[j] = s->fft_temp[ch][i].re * f;
- }
+ for (n = 0; n < s->hop_size; n++)
+ dst[n] = buf[n] * (1.f - s->overlap);
+ memmove(buf, buf + s->hop_size, window_size * 4);
+ }
- start += s->hop_size;
- end = j;
- }
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ goto fail;
- s->start = start;
- s->end = end;
+ av_audio_fifo_drain(s->fifo, s->hop_size);
- if (start >= window_size) {
- float *dst, *buf;
+fail:
+ av_frame_free(&in);
+ return ret < 0 ? ret : 0;
+}
- start -= window_size;
- end -= window_size;
+static int activate(AVFilterContext *ctx)
+{
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+ AFFTFiltContext *s = ctx->priv;
+ AVFrame *in = NULL;
+ int ret = 0, status;
+ int64_t pts;
- s->start = start;
- s->end = end;
+ FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
- out = ff_get_audio_buffer(outlink, window_size);
- if (!out) {
- ret = AVERROR(ENOMEM);
- break;
- }
+ if (!s->eof && av_audio_fifo_size(s->fifo) < s->window_size) {
+ ret = ff_inlink_consume_frame(inlink, &in);
+ if (ret < 0)
+ return ret;
- out->pts = s->pts;
- s->pts += window_size;
+ if (ret > 0) {
+ ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
+ in->nb_samples);
+ if (ret >= 0 && s->pts == AV_NOPTS_VALUE)
+ s->pts = in->pts;
- for (ch = 0; ch < inlink->channels; ch++) {
- dst = (float *)out->extended_data[ch];
- buf = (float *)s->buffer->extended_data[ch];
+ av_frame_free(&in);
+ if (ret < 0)
+ return ret;
+ }
+ }
- for (n = 0; n < window_size; n++) {
- dst[n] = buf[n] * (1 - s->overlap);
- }
- memmove(buf, buf + window_size, window_size * 4);
- }
+ if ((av_audio_fifo_size(s->fifo) >= s->window_size) ||
+ (av_audio_fifo_size(s->fifo) > 0 && s->eof)) {
+ ret = filter_frame(inlink);
+ if (av_audio_fifo_size(s->fifo) >= s->window_size)
+ ff_filter_set_ready(ctx, 100);
+ return ret;
+ }
- ret = ff_filter_frame(outlink, out);
- if (ret < 0)
- break;
+ if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
+ if (status == AVERROR_EOF) {
+ s->eof = 1;
+ if (av_audio_fifo_size(s->fifo) >= 0) {
+ ff_filter_set_ready(ctx, 100);
+ return 0;
+ }
}
+ }
- av_audio_fifo_drain(s->fifo, s->hop_size);
+ if (s->eof && av_audio_fifo_size(s->fifo) <= 0) {
+ ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
+ return 0;
}
- av_frame_free(&in);
- return ret < 0 ? ret : 0;
+ if (!s->eof)
+ FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+ return FFERROR_NOT_READY;
}
static int query_formats(AVFilterContext *ctx)
@@ -450,7 +474,6 @@ static const AVFilterPad inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
- .filter_frame = filter_frame,
},
{ NULL }
};
@@ -470,6 +493,7 @@ AVFilter ff_af_afftfilt = {
.priv_class = &afftfilt_class,
.inputs = inputs,
.outputs = outputs,
+ .activate = activate,
.query_formats = query_formats,
.uninit = uninit,
};
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