@@ -367,9 +367,7 @@ static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
const int ir_len = s->ir_len;
int nb_input_channels = ctx->inputs[0]->channels;
float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10);
- FFTComplex *fft_in_l = NULL;
- FFTComplex *fft_in_r = NULL;
- int offset = 0, ret = 0;
+ int ret = 0;
int n_fft;
int i, j, k;
@@ -381,13 +379,6 @@ static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
s->n_fft = n_fft = 1 << (32 - ff_clz(ir_len + s->size));
if (s->type == FREQUENCY_DOMAIN) {
- fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
- fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));
- if (!fft_in_l || !fft_in_r) {
- ret = AVERROR(ENOMEM);
- goto fail;
- }
-
s->fft[0] = av_fft_init(av_log2(s->n_fft), 0);
s->fft[1] = av_fft_init(av_log2(s->n_fft), 0);
s->ifft[0] = av_fft_init(av_log2(s->n_fft), 1);
@@ -464,10 +455,9 @@ static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
data_ir_r[j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
}
} else {
- memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
- memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
+ FFTComplex *fft_in_l = s->data_hrtf[0] + idx * n_fft;
+ FFTComplex *fft_in_r = s->data_hrtf[1] + idx * n_fft;
- offset = idx * n_fft;
for (j = 0; j < len; j++) {
fft_in_l[j].re = ptr[j * 2 ] * gain_lin;
fft_in_r[j].re = ptr[j * 2 + 1] * gain_lin;
@@ -475,10 +465,8 @@ static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
av_fft_permute(s->fft[0], fft_in_l);
av_fft_calc(s->fft[0], fft_in_l);
- memcpy(s->data_hrtf[0] + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
av_fft_permute(s->fft[0], fft_in_r);
av_fft_calc(s->fft[0], fft_in_r);
- memcpy(s->data_hrtf[1] + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
}
} else {
int I, N = ctx->inputs[1]->channels;
@@ -499,10 +487,9 @@ static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
data_ir_r[j] = ptr[len * N - j * N - N + I + 1] * gain_lin;
}
} else {
- memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
- memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
+ FFTComplex *fft_in_l = s->data_hrtf[0] + idx * n_fft;
+ FFTComplex *fft_in_r = s->data_hrtf[1] + idx * n_fft;
- offset = idx * n_fft;
for (j = 0; j < len; j++) {
fft_in_l[j].re = ptr[j * N + I ] * gain_lin;
fft_in_r[j].re = ptr[j * N + I + 1] * gain_lin;
@@ -510,10 +497,8 @@ static int convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
av_fft_permute(s->fft[0], fft_in_l);
av_fft_calc(s->fft[0], fft_in_l);
- memcpy(s->data_hrtf[0] + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
av_fft_permute(s->fft[0], fft_in_r);
av_fft_calc(s->fft[0], fft_in_r);
- memcpy(s->data_hrtf[1] + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
}
}
}
@@ -528,9 +513,6 @@ fail:
for (i = 0; i < s->nb_inputs - 1; i++)
av_frame_free(&s->in[i + 1].frame);
- av_freep(&fft_in_l);
- av_freep(&fft_in_r);
-
return ret;
}
The headphone filter allocates a pair of buffers to be used as intermediate buffers lateron: Before every use they are zeroed, then some elements of the buffer are set and lateron the complete buffers are copied into another, bigger buffer. These intermediate buffers are unnecessary as the data can be directly written into the bigger buffer. Furthermore, the whole buffer has been zeroed initially and because no piece of this buffer is set twice (due to the fact that duplicate channel map entries are skipped), it is unnecessary to rezero the part of the big buffer that is about to be written to. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com> --- libavfilter/af_headphone.c | 28 +++++----------------------- 1 file changed, 5 insertions(+), 23 deletions(-)