Commit dbf43ace authored by Paul B Mahol's avatar Paul B Mahol

afilter/af_afir: remove invalid delay

parent f266d2ac
......@@ -60,12 +60,9 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{
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);
......@@ -96,23 +93,18 @@ static int fir_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
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;
}
......@@ -138,10 +130,6 @@ static int fir_frame(AudioFIRContext *s, AVFrame *in, AVFilterLink *outlink)
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;
......@@ -329,7 +317,7 @@ static int convert_coeffs(AVFilterContext *ctx)
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);
......
......@@ -72,7 +72,6 @@ typedef struct AudioFIRContext {
AVFrame *buffer;
AVFrame *video;
int64_t pts;
int index;
AVFloatDSPContext *fdsp;
void (*fcmul_add)(float *sum, const float *t, const float *c,
......
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