avcodec: Remove deprecated AVPicture API

[ffmpeg] / libavfilter / af_headphone.c

diff --git a/libavfilter/af_headphone.c b/libavfilter/af_headphone.c
index bbbe32a475b6e2b377b93f8d36d21769ea692953..edf8e773d79d2679487be6b883f5bfd385b62f8c 100644 (file)
--- a/libavfilter/af_headphone.c
+++ b/libavfilter/af_headphone.c
@@ -75,11 +75,11 @@ typedef struct HeadphoneContext {
     FFTContext *fft[2], *ifft[2];
     FFTComplex *data_hrtf[2];
 
-    AVFloatDSPContext *fdsp;
+    float (*scalarproduct_float)(const float *v1, const float *v2, int len);
     struct hrir_inputs {
         int          ir_len;
         int          eof;
-    } *hrir_in;
+    } hrir_in[64];
     uint64_t mapping[64];
 } HeadphoneContext;
 
@@ -163,23 +163,22 @@ static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
     }
 
     for (i = 0; i < in->nb_samples; i++) {
-        const float *temp_ir = ir;
+        const float *cur_ir = ir;
 
         *dst = 0;
         for (l = 0; l < in_channels; l++) {
             *(buffer[l] + wr) = src[l];
         }
 
-        for (l = 0; l < in_channels; l++) {
+        for (l = 0; l < in_channels; cur_ir += air_len, l++) {
             const float *const bptr = buffer[l];
 
             if (l == s->lfe_channel) {
                 *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
-                temp_ir += air_len;
                 continue;
             }
 
-            read = (wr - (ir_len - 1) + buffer_length) & modulo;
+            read = (wr - (ir_len - 1)) & modulo;
 
             if (read + ir_len < buffer_length) {
                 memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src));
@@ -190,8 +189,7 @@ static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
                 memcpy(temp_src + len, bptr, (air_len - len) * sizeof(*temp_src));
             }
 
-            dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, FFALIGN(ir_len, 32));
-            temp_ir += air_len;
+            dst[0] += s->scalarproduct_float(cur_ir, temp_src, FFALIGN(ir_len, 32));
         }
 
         if (fabsf(dst[0]) > 1)
@@ -283,6 +281,8 @@ static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
 
     for (j = 0; j < in->nb_samples; j++) {
         dst[2 * j] += fft_acc[j].re * fft_scale;
+        if (fabsf(dst[2 * j]) > 1)
+            n_clippings[0]++;
     }
 
     for (j = 0; j < ir_len - 1; j++) {
@@ -291,14 +291,6 @@ static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
         *(ringbuffer + write_pos) += fft_acc[in->nb_samples + j].re * fft_scale;
     }
 
-    for (i = 0; i < out->nb_samples; i++) {
-        if (fabsf(dst[0]) > 1) {
-            n_clippings[0]++;
-        }
-
-        dst += 2;
-    }
-
     *write = wr;
 
     return 0;
@@ -654,10 +646,6 @@ static av_cold int init(AVFilterContext *ctx)
 
     parse_map(ctx);
 
-    s->hrir_in = av_calloc(s->nb_hrir_inputs, sizeof(*s->hrir_in));
-    if (!s->hrir_in)
-        return AVERROR(ENOMEM);
-
     for (i = 0; i < s->nb_hrir_inputs; i++) {
         char *name = av_asprintf("hrir%d", i);
         AVFilterPad pad = {
@@ -672,9 +660,13 @@ static av_cold int init(AVFilterContext *ctx)
         }
     }
 
-    s->fdsp = avpriv_float_dsp_alloc(0);
-    if (!s->fdsp)
-        return AVERROR(ENOMEM);
+    if (s->type == TIME_DOMAIN) {
+        AVFloatDSPContext *fdsp = avpriv_float_dsp_alloc(0);
+        if (!fdsp)
+            return AVERROR(ENOMEM);
+        s->scalarproduct_float = fdsp->scalarproduct_float;
+        av_free(fdsp);
+    }
 
     return 0;
 }
@@ -719,9 +711,7 @@ static av_cold void uninit(AVFilterContext *ctx)
     av_freep(&s->temp_afft[1]);
     av_freep(&s->data_hrtf[0]);
     av_freep(&s->data_hrtf[1]);
-    av_freep(&s->fdsp);
 
-    av_freep(&s->hrir_in);
     for (unsigned i = 1; i < ctx->nb_inputs; i++)
         av_freep(&ctx->input_pads[i].name);
 }