static int config_output(AVFilterLink *outlink)
{
AudioPhaserContext *p = outlink->src->priv;
AVFilterLink *inlink = outlink->src->inputs[0];
p->delay_buffer_length = p->delay * 0.001 * inlink->sample_rate + 0.5;
p->delay_buffer = av_calloc(p->delay_buffer_length, sizeof(*p->delay_buffer) * inlink->channels);
p->modulation_buffer_length = inlink->sample_rate / p->speed + 0.5;
p->modulation_buffer = av_malloc_array(p->modulation_buffer_length, sizeof(*p->modulation_buffer));
if (!p->modulation_buffer || !p->delay_buffer)
return AVERROR(ENOMEM);
ff_generate_wave_table(p->type, AV_SAMPLE_FMT_S32,
p->modulation_buffer, p->modulation_buffer_length,
1., p->delay_buffer_length, M_PI / 2.0);
p->delay_pos = p->modulation_pos = 0;
switch (inlink->format) {
case AV_SAMPLE_FMT_DBL: p->phaser = phaser_dbl; break;
case AV_SAMPLE_FMT_DBLP: p->phaser = phaser_dblp; break;
case AV_SAMPLE_FMT_FLT: p->phaser = phaser_flt; break;
case AV_SAMPLE_FMT_FLTP: p->phaser = phaser_fltp; break;
case AV_SAMPLE_FMT_S16: p->phaser = phaser_s16; break;
case AV_SAMPLE_FMT_S16P: p->phaser = phaser_s16p; break;
case AV_SAMPLE_FMT_S32: p->phaser = phaser_s32; break;
case AV_SAMPLE_FMT_S32P: p->phaser = phaser_s32p; break;
default: av_assert0(0);
}
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
FlangerContext *s = ctx->priv;
s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
s->lfo_length = inlink->sample_rate / s->speed;
s->delay_last = av_calloc(inlink->channels, sizeof(*s->delay_last));
s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
if (!s->lfo || !s->delay_last)
return AVERROR(ENOMEM);
ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
floor(s->delay_min * inlink->sample_rate + 0.5),
s->max_samples - 2., 3 * M_PI_2);
return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
inlink->channels, s->max_samples,
inlink->format, 0);
}