{
AudioFIRContext *s = ctx->priv;
const float *src = (const float *)s->in[0]->extended_data[ch];
- int index1 = (s->index + 1) % 3;
- int index2 = (s->index + 2) % 3;
float *sum = s->sum[ch];
AVFrame *out = arg;
- float *block;
- float *dst;
+ float *block, *dst, *ptr;
int n, i, j;
memset(sum, 0, sizeof(*sum) * s->fft_length);
sum[1] = sum[2 * s->part_size];
av_rdft_calc(s->irdft[ch], sum);
- dst = (float *)s->buffer->extended_data[ch] + index1 * s->part_size;
+ dst = (float *)s->buffer->extended_data[ch];
for (n = 0; n < s->part_size; n++) {
dst[n] += sum[n];
}
- dst = (float *)s->buffer->extended_data[ch] + index2 * s->part_size;
+ ptr = (float *)out->extended_data[ch];
+ s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
+ emms_c();
+ dst = (float *)s->buffer->extended_data[ch];
memcpy(dst, sum + s->part_size, s->part_size * sizeof(*dst));
- dst = (float *)s->buffer->extended_data[ch] + s->index * s->part_size;
-
- if (out) {
- float *ptr = (float *)out->extended_data[ch];
- s->fdsp->vector_fmul_scalar(ptr, dst, s->wet_gain, FFALIGN(out->nb_samples, 4));
- emms_c();
- }
-
return 0;
}
if (s->pts != AV_NOPTS_VALUE)
s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
- s->index++;
- if (s->index == 3)
- s->index = 0;
-
av_frame_free(&in);
s->in[0] = NULL;
return AVERROR(ENOMEM);
}
- s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size * 3);
+ s->buffer = ff_get_audio_buffer(ctx->inputs[0], s->part_size);
if (!s->buffer)
return AVERROR(ENOMEM);