[FFmpeg-cvslog] avfilter/af_adynamicequalizer: add new structure to hold filtering state

[Paul B Mahol]

ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat Nov  4 14:34:18 2023 +0100| [43226efc2188ec2cd16d3b9c4e189bcde1a2a0e5] | committer: Paul B Mahol

avfilter/af_adynamicequalizer: add new structure to hold filtering state

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=43226efc2188ec2cd16d3b9c4e189bcde1a2a0e5
---

 libavfilter/adynamicequalizer_template.c | 35 ++++++++++++++++----------------
 libavfilter/af_adynamicequalizer.c       | 21 +++++++++++++++----
 2 files changed, 34 insertions(+), 22 deletions(-)

diff --git a/libavfilter/adynamicequalizer_template.c b/libavfilter/adynamicequalizer_template.c
index 26fcebf8bc..5667b50b1d 100644
--- a/libavfilter/adynamicequalizer_template.c
+++ b/libavfilter/adynamicequalizer_template.c
@@ -166,25 +166,25 @@ static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int n
     for (int ch = start; ch < end; ch++) {
         const ftype *src = (const ftype *)in->extended_data[ch];
         ftype *dst = (ftype *)out->extended_data[ch];
-        ftype *state = (ftype *)s->state->extended_data[ch];
-        const ftype threshold = detection == 0 ? state[5] : s->threshold;
-        ftype fa[3], fm[3];
+        ChannelContext *cc = &s->cc[ch];
+        const ftype threshold = detection == 0 ? fn(cc->threshold) : s->threshold;
+        ftype *fa = fn(cc->fa), *fm = fn(cc->fm);
+        ftype *fstate = fn(cc->fstate);
+        ftype *dstate = fn(cc->dstate);
+        ftype gain = fn(cc->gain);
 
         if (detection < 0)
-            state[5] = threshold;
-
-        memcpy(fa, state +  8, sizeof(fa));
-        memcpy(fm, state + 11, sizeof(fm));
+            fn(cc->threshold) = threshold;
 
         for (int n = 0; n < out->nb_samples; n++) {
-            ftype detect, gain, v, listen;
+            ftype detect, v, listen;
             ftype k, g;
 
-            detect = listen = fn(get_svf)(src[n], dm, da, state);
+            detect = listen = fn(get_svf)(src[n], dm, da, dstate);
             detect = FABS(detect);
 
             if (detection > 0)
-                state[5] = FMAX(state[5], detect);
+                fn(cc->threshold) = FMAX(fn(cc->threshold), detect);
 
             if (mode >= 0) {
                 if (direction == 0 && detect < threshold) {
@@ -200,17 +200,17 @@ static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int n
                 }
 
                 {
-                    ftype delta = detect - state[4];
+                    ftype delta = detect - gain;
 
                     if (delta > EPSILON)
-                        detect = state[4] + attack * delta;
+                        detect = gain + attack * delta;
                     else if (delta < -EPSILON)
-                        detect = state[4] + release * delta;
+                        detect = gain + release * delta;
                 }
             }
 
-            if (state[4] != detect) {
-                state[4] = gain = detect;
+            if (gain != detect) {
+                gain = detect;
 
                 switch (tftype) {
                 case 0:
@@ -251,13 +251,12 @@ static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int n
                 }
             }
 
-            v = fn(get_svf)(src[n], fm, fa, &state[2]);
+            v = fn(get_svf)(src[n], fm, fa, fstate);
             v = mode == -1 ? listen : v;
             dst[n] = ctx->is_disabled ? src[n] : v;
         }
 
-        memcpy(state +  8, fa, sizeof(fa));
-        memcpy(state + 11, fm, sizeof(fm));
+        fn(cc->gain) = gain;
     }
 
     return 0;
diff --git a/libavfilter/af_adynamicequalizer.c b/libavfilter/af_adynamicequalizer.c
index 2c674cbd7c..a84e8e5f57 100644
--- a/libavfilter/af_adynamicequalizer.c
+++ b/libavfilter/af_adynamicequalizer.c
@@ -23,6 +23,19 @@
 #include "audio.h"
 #include "formats.h"
 
+typedef struct ChannelContext {
+    double fa_double[3], fm_double[3];
+    double dstate_double[2];
+    double fstate_double[2];
+    double gain_double;
+    double threshold_double;
+    float fa_float[3], fm_float[3];
+    float dstate_float[2];
+    float fstate_float[2];
+    float gain_float;
+    float threshold_float;
+} ChannelContext;
+
 typedef struct AudioDynamicEqualizerContext {
     const AVClass *class;
 
@@ -52,7 +65,7 @@ typedef struct AudioDynamicEqualizerContext {
     double da_double[3], dm_double[3];
     float da_float[3], dm_float[3];
 
-    AVFrame *state;
+    ChannelContext *cc;
 } AudioDynamicEqualizerContext;
 
 static int query_formats(AVFilterContext *ctx)
@@ -96,8 +109,8 @@ static int config_input(AVFilterLink *inlink)
     AudioDynamicEqualizerContext *s = ctx->priv;
 
     s->format = inlink->format;
-    s->state = ff_get_audio_buffer(inlink, 16);
-    if (!s->state)
+    s->cc = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->cc));
+    if (!s->cc)
         return AVERROR(ENOMEM);
 
     switch (s->format) {
@@ -148,7 +161,7 @@ static av_cold void uninit(AVFilterContext *ctx)
 {
     AudioDynamicEqualizerContext *s = ctx->priv;
 
-    av_frame_free(&s->state);
+    av_freep(&s->cc);
 }
 
 #define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)