mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
avfilter/af_sofalizer: move modulo operation out of loop
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
parent
1acc90eaa5
commit
49d97d9bca
@ -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 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 */
|
||||
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 */
|
||||
int buffer_length = s->buffer_length;
|
||||
const int buffer_length = s->buffer_length;
|
||||
/* -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 */
|
||||
int wr = *write;
|
||||
int read;
|
||||
int i, j, l;
|
||||
int i, l;
|
||||
|
||||
dst += offset;
|
||||
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) */
|
||||
read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo;
|
||||
|
||||
for (j = 0; j < n_samples; j++)
|
||||
temp_src[j] = bptr[(read + j) & modulo];
|
||||
if (read + n_samples < buffer_length) {
|
||||
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 */
|
||||
dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples);
|
||||
|
Loading…
Reference in New Issue
Block a user