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avfilter/af_sofalizer: move modulo operation out of loop

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2015-12-13 11:52:23 +01:00
parent 1acc90eaa5
commit 49d97d9bca
1 changed files with 10 additions and 6 deletions
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16
libavfilter/af_sofalizer.c
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@ -654,15 +654,15 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
const int n_samples = s->sofa.n_samples; /* length of one IR */ const int n_samples = s->sofa.n_samples; /* length of one IR */
const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */ const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */
float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */ float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */
int in_channels = in->channels; /* number of input channels */ const int in_channels = in->channels; /* number of input channels */
/* ring buffer length is: longest IR plus max. delay -> next power of 2 */ /* ring buffer length is: longest IR plus max. delay -> next power of 2 */
int buffer_length = s->buffer_length; const int buffer_length = s->buffer_length;
/* -1 for AND instead of MODULO (applied to powers of 2): */ /* -1 for AND instead of MODULO (applied to powers of 2): */
uint32_t modulo = (uint32_t)buffer_length - 1; const uint32_t modulo = (uint32_t)buffer_length - 1;
float *buffer[10]; /* holds ringbuffer for each input channel */ float *buffer[10]; /* holds ringbuffer for each input channel */
int wr = *write; int wr = *write;
int read; int read;
int i, j, l; int i, l;
dst += offset; dst += offset;
for (l = 0; l < in_channels; l++) { for (l = 0; l < in_channels; l++) {
@ -688,8 +688,12 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
* (mod buffer length) */ * (mod buffer length) */
read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo; read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo;
for (j = 0; j < n_samples; j++) if (read + n_samples < buffer_length) {
temp_src[j] = bptr[(read + j) & modulo]; memcpy(temp_src, bptr + read, n_samples * sizeof(*temp_src));
} else {
memcpy(temp_src, bptr + read, (buffer_length - read) * sizeof(*temp_src));
memcpy(temp_src + (buffer_length - read), bptr, (read - n_samples) * sizeof(*temp_src));
}
/* multiply signal and IR, and add up the results */ /* multiply signal and IR, and add up the results */
dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples); dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples);