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authorPaul B Mahol <onemda@gmail.com>2018-12-25 19:17:25 +0100
committerPaul B Mahol <onemda@gmail.com>2018-12-25 19:17:25 +0100
commit6c7eb0708e659987b82e32f969c94f870ae2e3cd (patch)
tree8476cb5436106047d6b4682fce5a18e0a1e2fa49 /libavfilter
parent8d0b8c50bd234e84f373117a0fb4a564db051482 (diff)
avfilter/af_headphone: speed up fast convolution
Do IFFT only once per output channel.
Diffstat (limited to 'libavfilter')
-rw-r--r--libavfilter/af_headphone.c35
1 files changed, 23 insertions, 12 deletions
diff --git a/libavfilter/af_headphone.c b/libavfilter/af_headphone.c
index 51b5921c1c..dab897f00a 100644
--- a/libavfilter/af_headphone.c
+++ b/libavfilter/af_headphone.c
@@ -73,6 +73,7 @@ typedef struct HeadphoneContext {
float *data_ir[2];
float *temp_src[2];
FFTComplex *temp_fft[2];
+ FFTComplex *temp_afft[2];
FFTContext *fft[2], *ifft[2];
FFTComplex *data_hrtf[2];
@@ -158,6 +159,7 @@ typedef struct ThreadData {
float **ringbuffer;
float **temp_src;
FFTComplex **temp_fft;
+ FFTComplex **temp_afft;
} ThreadData;
static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
@@ -251,6 +253,7 @@ static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
const int buffer_length = s->buffer_length;
const uint32_t modulo = (uint32_t)buffer_length - 1;
FFTComplex *fft_in = s->temp_fft[jobnr];
+ FFTComplex *fft_acc = s->temp_afft[jobnr];
FFTContext *ifft = s->ifft[jobnr];
FFTContext *fft = s->fft[jobnr];
const int n_fft = s->n_fft;
@@ -273,6 +276,8 @@ static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
dst[2 * j] = 0;
}
+ memset(fft_acc, 0, sizeof(FFTComplex) * n_fft);
+
for (i = 0; i < in_channels; i++) {
if (i == s->lfe_channel) {
for (j = 0; j < in->nb_samples; j++) {
@@ -297,22 +302,22 @@ static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
const float re = fft_in[j].re;
const float im = fft_in[j].im;
- fft_in[j].re = re * hcomplex->re - im * hcomplex->im;
- fft_in[j].im = re * hcomplex->im + im * hcomplex->re;
+ fft_acc[j].re += re * hcomplex->re - im * hcomplex->im;
+ fft_acc[j].im += re * hcomplex->im + im * hcomplex->re;
}
+ }
- av_fft_permute(ifft, fft_in);
- av_fft_calc(ifft, fft_in);
+ av_fft_permute(ifft, fft_acc);
+ av_fft_calc(ifft, fft_acc);
- for (j = 0; j < in->nb_samples; j++) {
- dst[2 * j] += fft_in[j].re * fft_scale;
- }
+ for (j = 0; j < in->nb_samples; j++) {
+ dst[2 * j] += fft_acc[j].re * fft_scale;
+ }
- for (j = 0; j < ir_len - 1; j++) {
- int write_pos = (wr + j) & modulo;
+ for (j = 0; j < ir_len - 1; j++) {
+ int write_pos = (wr + j) & modulo;
- *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re * fft_scale;
- }
+ *(ringbuffer + write_pos) += fft_acc[in->nb_samples + j].re * fft_scale;
}
for (i = 0; i < out->nb_samples; i++) {
@@ -364,6 +369,7 @@ static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outli
td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
td.temp_fft = s->temp_fft;
+ td.temp_afft = s->temp_afft;
if (s->type == TIME_DOMAIN) {
ctx->internal->execute(ctx, headphone_convolute, &td, NULL, 2);
@@ -439,7 +445,10 @@ static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
s->temp_fft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
s->temp_fft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
- if (!s->temp_fft[0] || !s->temp_fft[1]) {
+ s->temp_afft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
+ s->temp_afft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
+ if (!s->temp_fft[0] || !s->temp_fft[1] ||
+ !s->temp_afft[0] || !s->temp_afft[1]) {
ret = AVERROR(ENOMEM);
goto fail;
}
@@ -819,6 +828,8 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->temp_src[1]);
av_freep(&s->temp_fft[0]);
av_freep(&s->temp_fft[1]);
+ av_freep(&s->temp_afft[0]);
+ av_freep(&s->temp_afft[1]);
av_freep(&s->data_hrtf[0]);
av_freep(&s->data_hrtf[1]);
av_freep(&s->fdsp);