[FFmpeg-cvslog] avfilter/af_afir: make IR switching work also with minp != maxp

Paul B Mahol git at videolan.org
Tue Apr 25 22:25:01 EEST 2023


ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Tue Apr 25 15:21:51 2023 +0200| [1835f884b7ae475d2079d044003966981823502e] | committer: Paul B Mahol

avfilter/af_afir: make IR switching work also with minp != maxp

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=1835f884b7ae475d2079d044003966981823502e
---

 libavfilter/af_afir.c       | 76 ++++++++++++++++++++++++++++++++++++---------
 libavfilter/af_afir.h       |  2 ++
 libavfilter/afir_template.c | 30 ++----------------
 3 files changed, 67 insertions(+), 41 deletions(-)

diff --git a/libavfilter/af_afir.c b/libavfilter/af_afir.c
index 4acc24ccac..338e23063c 100644
--- a/libavfilter/af_afir.c
+++ b/libavfilter/af_afir.c
@@ -112,16 +112,59 @@ static int fir_channel(AVFilterContext *ctx, AVFrame *out, int ch)
     for (int offset = 0; offset < out->nb_samples; offset += min_part_size) {
         switch (s->format) {
         case AV_SAMPLE_FMT_FLTP:
-            if (prev_selir != selir)
-                fir_quantum_float(ctx, out, ch, offset, prev_selir);
-            fir_quantum_float(ctx, out, ch, offset, selir);
+            if (prev_selir != selir) {
+                const float *xfade0 = (const float *)s->xfade[0]->extended_data[ch];
+                const float *xfade1 = (const float *)s->xfade[1]->extended_data[ch];
+                float *src0 = (float *)s->fadein[0]->extended_data[ch];
+                float *src1 = (float *)s->fadein[1]->extended_data[ch];
+                float *dst = ((float *)out->extended_data[ch]) + offset;
+
+                memset(src0, 0, min_part_size * sizeof(float));
+                memset(src1, 0, min_part_size * sizeof(float));
+
+                fir_quantum_float(ctx, s->fadein[0], ch, offset, 0, prev_selir);
+                fir_quantum_float(ctx, s->fadein[1], ch, offset, 0, selir);
+
+                if (s->loading[ch] > 0 && s->loading[ch] <= min_part_size) {
+                    for (int n = 0; n < min_part_size; n++)
+                        dst[n] = xfade1[n] * src0[n] + xfade0[n] * src1[n];
+                    s->loading[ch] = 0;
+                } else {
+                    memcpy(dst, src0, min_part_size * sizeof(float));
+                }
+            } else {
+                fir_quantum_float(ctx, out, ch, offset, offset, selir);
+            }
             break;
         case AV_SAMPLE_FMT_DBLP:
-            if (prev_selir != selir)
-                fir_quantum_double(ctx, out, ch, offset, prev_selir);
-            fir_quantum_double(ctx, out, ch, offset, selir);
+            if (prev_selir != selir) {
+                const double *xfade0 = (const double *)s->xfade[0]->extended_data[ch];
+                const double *xfade1 = (const double *)s->xfade[1]->extended_data[ch];
+                double *src0 = (double *)s->fadein[0]->extended_data[ch];
+                double *src1 = (double *)s->fadein[1]->extended_data[ch];
+                double *dst = ((double *)out->extended_data[ch]) + offset;
+
+                memset(src0, 0, min_part_size * sizeof(double));
+                memset(src1, 0, min_part_size * sizeof(double));
+
+                fir_quantum_double(ctx, s->fadein[0], ch, offset, 0, prev_selir);
+                fir_quantum_double(ctx, s->fadein[1], ch, offset, 0, selir);
+
+                if (s->loading[ch] > 0 && s->loading[ch] <= min_part_size) {
+                    for (int n = 0; n < min_part_size; n++)
+                        dst[n] = xfade1[n] * src0[n] + xfade0[n] * src1[n];
+                    s->loading[ch] = 0;
+                } else {
+                    memcpy(dst, src0, min_part_size * sizeof(double));
+                }
+            } else {
+                fir_quantum_double(ctx, out, ch, offset, offset, selir);
+            }
             break;
         }
+
+        if (selir != prev_selir && s->loading[ch] > 0)
+            s->loading[ch] -= min_part_size;
     }
 
     return 0;
@@ -299,10 +342,11 @@ static int convert_coeffs(AVFilterContext *ctx, int selir)
         max_part_size = 1 << av_log2(s->maxp);
 
         for (int i = 0; left > 0; i++) {
-            int step = part_size == max_part_size ? INT_MAX : 1 + (i == 0);
+            int step = (part_size == max_part_size) ? INT_MAX : 1 + (i == 0);
             int nb_partitions = FFMIN(step, (left + part_size - 1) / part_size);
 
             s->nb_segments[selir] = i + 1;
+            s->max_offset[selir] = offset;
             ret = init_segment(ctx, &s->seg[selir][i], selir, offset, nb_partitions, part_size, i);
             if (ret < 0)
                 return ret;
@@ -311,8 +355,6 @@ static int convert_coeffs(AVFilterContext *ctx, int selir)
             part_size *= 2;
             part_size = FFMIN(part_size, max_part_size);
         }
-
-        s->max_offset[selir] = offset;
     }
 
 skip:
@@ -455,7 +497,7 @@ static int activate(AVFilterContext *ctx)
 
     if (s->selir != s->prev_selir && s->loading[0] <= 0) {
         for (int ch = 0; ch < s->nb_channels; ch++)
-            s->loading[ch] = s->max_offset[s->selir] + s->min_part_size;
+            s->loading[ch] = s->max_offset[s->selir] + s->max_part_size;
     }
 
     available = ff_inlink_queued_samples(ctx->inputs[0]);
@@ -464,11 +506,8 @@ static int activate(AVFilterContext *ctx)
     if (ret > 0)
         ret = fir_frame(s, in, outlink);
 
-    if (s->selir != s->prev_selir && s->loading[0] <= 0) {
+    if (s->selir != s->prev_selir && s->loading[0] <= 0)
         s->prev_selir = s->selir;
-        for (int ch = 0; ch < s->nb_channels; ch++)
-            s->loading[ch] = 0;
-    }
 
     if (ret < 0)
         return ret;
@@ -590,6 +629,11 @@ FF_ENABLE_DEPRECATION_WARNINGS
     if (!s->loading)
         return AVERROR(ENOMEM);
 
+    s->fadein[0] = ff_get_audio_buffer(outlink, s->min_part_size);
+    s->fadein[1] = ff_get_audio_buffer(outlink, s->min_part_size);
+    if (!s->fadein[0] || !s->fadein[1])
+        return AVERROR(ENOMEM);
+
     s->xfade[0] = ff_get_audio_buffer(outlink, s->min_part_size);
     s->xfade[1] = ff_get_audio_buffer(outlink, s->min_part_size);
     if (!s->xfade[0] || !s->xfade[1])
@@ -638,6 +682,9 @@ static av_cold void uninit(AVFilterContext *ctx)
         av_frame_free(&s->norm_ir[i]);
     }
 
+    av_frame_free(&s->fadein[0]);
+    av_frame_free(&s->fadein[1]);
+
     av_frame_free(&s->xfade[0]);
     av_frame_free(&s->xfade[1]);
 
@@ -723,6 +770,7 @@ static av_cold int init(AVFilterContext *ctx)
     ff_afir_init(&s->afirdsp);
 
     s->min_part_size = 1 << av_log2(s->minp);
+    s->max_part_size = 1 << av_log2(s->maxp);
 
     return 0;
 }
diff --git a/libavfilter/af_afir.h b/libavfilter/af_afir.h
index 5f2c68c5c6..099897d0bb 100644
--- a/libavfilter/af_afir.h
+++ b/libavfilter/af_afir.h
@@ -91,10 +91,12 @@ typedef struct AudioFIRContext {
 
     AVFrame *in;
     AVFrame *xfade[2];
+    AVFrame *fadein[2];
     AVFrame *ir[MAX_IR_STREAMS];
     AVFrame *norm_ir[MAX_IR_STREAMS];
     AVFrame *video;
     int min_part_size;
+    int max_part_size;
     int64_t pts;
 
     AudioFIRDSPContext afirdsp;
diff --git a/libavfilter/afir_template.c b/libavfilter/afir_template.c
index 5ddc79f6c1..0f0500d8b3 100644
--- a/libavfilter/afir_template.c
+++ b/libavfilter/afir_template.c
@@ -285,15 +285,13 @@ static void fn(fir_fadd)(AudioFIRContext *s, ftype *dst, const ftype *src, int n
     }
 }
 
-static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int offset, int selir)
+static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int ioffset, int offset, int selir)
 {
     AudioFIRContext *s = ctx->priv;
-    const ftype *in = (const ftype *)s->in->extended_data[ch] + offset;
+    const ftype *in = (const ftype *)s->in->extended_data[ch] + ioffset;
     ftype *blockout, *ptr = (ftype *)out->extended_data[ch] + offset;
     const int min_part_size = s->min_part_size;
     const int nb_samples = FFMIN(min_part_size, out->nb_samples - offset);
-    const ftype *xfade0 = (const ftype *)s->xfade[0]->extended_data[ch];
-    const ftype *xfade1 = (const ftype *)s->xfade[1]->extended_data[ch];
     const int nb_segments = s->nb_segments[selir];
     const float dry_gain = s->dry_gain;
 
@@ -368,18 +366,7 @@ static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int offse
         memcpy(dst, buf, part_size * sizeof(*dst));
         memcpy(buf, sumout + part_size, part_size * sizeof(*buf));
 
-        if (s->selir != s->prev_selir) {
-            if (selir == s->selir) {
-                if (s->loading[ch] <= min_part_size) {
-                    for (int n = 0; n < nb_samples; n++)
-                        ptr[n] += dst[n] * xfade0[n];
-                }
-            } else {
-                fn(fir_fadd)(s, ptr, dst, nb_samples);
-            }
-        } else {
-            fn(fir_fadd)(s, ptr, dst, nb_samples);
-        }
+        fn(fir_fadd)(s, ptr, dst, nb_samples);
 
         if (part_size != min_part_size)
             memmove(src, src + min_part_size, (seg->input_size - min_part_size) * sizeof(*src));
@@ -387,17 +374,6 @@ static int fn(fir_quantum)(AVFilterContext *ctx, AVFrame *out, int ch, int offse
         seg->part_index[ch] = (seg->part_index[ch] + 1) % nb_partitions;
     }
 
-    if (selir != s->selir) {
-        if (s->loading[ch] <= min_part_size) {
-            for (int n = 0; n < nb_samples; n++)
-                ptr[n] *= xfade1[n];
-        }
-        return 0;
-    }
-
-    if (s->selir != s->prev_selir)
-        s->loading[ch] -= min_part_size;
-
     if (s->wet_gain == 1.f)
         return 0;
 



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