avfilter/af_sofalizer: move modulo operation out of loop
Signed-off-by: Paul B Mahol <onemda@gmail.com>
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1acc90eaa5
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49d97d9bca
@ -654,15 +654,15 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
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const int n_samples = s->sofa.n_samples; /* length of one IR */
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const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */
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float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */
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int in_channels = in->channels; /* number of input channels */
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const int in_channels = in->channels; /* number of input channels */
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/* ring buffer length is: longest IR plus max. delay -> next power of 2 */
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int buffer_length = s->buffer_length;
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const int buffer_length = s->buffer_length;
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/* -1 for AND instead of MODULO (applied to powers of 2): */
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uint32_t modulo = (uint32_t)buffer_length - 1;
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const uint32_t modulo = (uint32_t)buffer_length - 1;
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float *buffer[10]; /* holds ringbuffer for each input channel */
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int wr = *write;
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int read;
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int i, j, l;
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int i, l;
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dst += offset;
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for (l = 0; l < in_channels; l++) {
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@ -688,8 +688,12 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
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* (mod buffer length) */
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read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo;
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for (j = 0; j < n_samples; j++)
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temp_src[j] = bptr[(read + j) & modulo];
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if (read + n_samples < buffer_length) {
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memcpy(temp_src, bptr + read, n_samples * sizeof(*temp_src));
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} else {
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memcpy(temp_src, bptr + read, (buffer_length - read) * sizeof(*temp_src));
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memcpy(temp_src + (buffer_length - read), bptr, (read - n_samples) * sizeof(*temp_src));
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}
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/* multiply signal and IR, and add up the results */
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dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples);
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