static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ShowFreqsContext *s = ctx->priv;
AVFrame *fin = NULL;
int ret = 0;
av_audio_fifo_write(s->fifo, (void **)in->extended_data, in->nb_samples);
while (av_audio_fifo_size(s->fifo) >= s->win_size) {
fin = ff_get_audio_buffer(inlink, s->win_size);
if (!fin) {
ret = AVERROR(ENOMEM);
goto fail;
}
fin->pts = s->pts;
s->pts += s->skip_samples;
ret = av_audio_fifo_peek(s->fifo, (void **)fin->extended_data, s->win_size);
if (ret < 0)
goto fail;
ret = plot_freqs(inlink, fin);
av_frame_free(&fin);
av_audio_fifo_drain(s->fifo, s->skip_samples);
if (ret < 0)
goto fail;
}
fail:
av_frame_free(&fin);
av_frame_free(&in);
return ret;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioSurroundContext *s = ctx->priv;
av_audio_fifo_write(s->fifo, (void **)in->extended_data,
in->nb_samples);
if (s->pts == AV_NOPTS_VALUE)
s->pts = in->pts;
av_frame_free(&in);
while (av_audio_fifo_size(s->fifo) >= s->buf_size) {
AVFrame *out;
int ret;
ret = av_audio_fifo_peek(s->fifo, (void **)s->input->extended_data, s->buf_size);
if (ret < 0)
return ret;
ctx->internal->execute(ctx, fft_channel, NULL, NULL, inlink->channels);
s->filter(ctx);
out = ff_get_audio_buffer(outlink, s->hop_size);
if (!out)
return AVERROR(ENOMEM);
ctx->internal->execute(ctx, ifft_channel, out, NULL, outlink->channels);
out->pts = s->pts;
if (s->pts != AV_NOPTS_VALUE)
s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
av_audio_fifo_drain(s->fifo, s->hop_size);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
return ret;
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AFFTFiltContext *s = ctx->priv;
const int window_size = s->window_size;
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;
av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
av_frame_free(&frame);
while (av_audio_fifo_size(s->fifo) >= window_size) {
if (!in) {
in = ff_get_audio_buffer(outlink, window_size);
if (!in)
return AVERROR(ENOMEM);
}
ret = av_audio_fifo_peek(s->fifo, (void **)in->extended_data, window_size);
if (ret < 0)
break;
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;
}
for (; n < window_size; n++) {
fft_data[n].re = 0;
fft_data[n].im = 0;
}
}
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];
float *buf = (float *)s->buffer->extended_data[ch];
int x;
values[VAR_CHANNEL] = ch;
av_fft_permute(s->fft, fft_data);
av_fft_calc(s->fft, fft_data);
for (n = 0; n < window_size / 2; n++) {
float fr, fi;
values[VAR_BIN] = n;
fr = av_expr_eval(s->real[ch], values, s);
fi = av_expr_eval(s->imag[ch], values, s);
fft_data[n].re *= fr;
fft_data[n].im *= fi;
}
for (n = window_size / 2 + 1, x = window_size / 2 - 1; n < window_size; n++, x--) {
fft_data[n].re = fft_data[x].re;
fft_data[n].im = -fft_data[x].im;
}
av_fft_permute(s->ifft, fft_data);
av_fft_calc(s->ifft, fft_data);
start = s->start;
end = s->end;
k = end;
for (i = 0, j = start; j < k && i < window_size; i++, j++) {
buf[j] += s->fft_data[ch][i].re * f;
}
for (; i < window_size; i++, j++) {
buf[j] = s->fft_data[ch][i].re * f;
}
start += s->hop_size;
end = j;
}
s->start = start;
s->end = end;
if (start >= window_size) {
float *dst, *buf;
start -= window_size;
end -= window_size;
s->start = start;
s->end = end;
out = ff_get_audio_buffer(outlink, window_size);
if (!out) {
ret = AVERROR(ENOMEM);
break;
}
out->pts = s->pts;
s->pts += window_size;
for (ch = 0; ch < inlink->channels; ch++) {
dst = (float *)out->extended_data[ch];
buf = (float *)s->buffer->extended_data[ch];
for (n = 0; n < window_size; n++) {
dst[n] = buf[n] * (1 - s->overlap);
}
memmove(buf, buf + window_size, window_size * 4);
}
ret = ff_filter_frame(outlink, out);
if (ret < 0)
break;
}
av_audio_fifo_drain(s->fifo, s->hop_size);
}
av_frame_free(&in);
return ret;
}
