Re: [FFmpeg-devel] [PATCH 2/3] avfilter/af_volumedetect.c: Add 32bit float audio support

["Rémi Denis-Courmont" <remi@remlab.net>]

Le 17 juin 2024 13:18:11 GMT+02:00, Yigithan Yigit <yigithanyigitdevel@gmail.com> a écrit :
>---
> libavfilter/af_volumedetect.c | 159 ++++++++++++++++++++++++++++------
> 1 file changed, 133 insertions(+), 26 deletions(-)
>
>diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c
>index 327801a7f9..dbbcd037a5 100644
>--- a/libavfilter/af_volumedetect.c
>+++ b/libavfilter/af_volumedetect.c
>@@ -20,27 +20,51 @@
> 
> #include "libavutil/channel_layout.h"
> #include "libavutil/avassert.h"
>+#include "libavutil/mem.h"
> #include "audio.h"
> #include "avfilter.h"
> #include "internal.h"
> 
>+#define MAX_DB_FLT 1024
> #define MAX_DB 91
>+#define HISTOGRAM_SIZE 0x10000
>+#define HISTOGRAM_SIZE_FLT (MAX_DB_FLT*2)
> 
> typedef struct VolDetectContext {
>-    /**
>-     * Number of samples at each PCM value.
>-     * histogram[0x8000 + i] is the number of samples at value i.
>-     * The extra element is there for symmetry.
>-     */
>-    uint64_t histogram[0x10001];
>+    uint64_t* histogram; ///< for integer number of samples at each PCM value, for float number of samples at each dB
>+    uint64_t nb_samples; ///< number of samples
>+    double sum2;         ///< sum of the squares of the samples
>+    double max;          ///< maximum sample value
>+    int is_float;        ///< true if the input is in floating point
> } VolDetectContext;
> 
>-static inline double logdb(uint64_t v)
>+static inline double logdb(double v, enum AVSampleFormat sample_fmt)
> {
>-    double d = v / (double)(0x8000 * 0x8000);
>-    if (!v)
>-        return MAX_DB;
>-    return -log10(d) * 10;
>+    if (sample_fmt == AV_SAMPLE_FMT_FLT) {
>+        if (!v)
>+            return MAX_DB_FLT;
>+        return -log10(v) * 10;
>+    } else {
>+        double d = v / (double)(0x8000 * 0x8000);
>+        if (!v)
>+            return MAX_DB;
>+        return -log10(d) * 10;
>+    }
>+}
>+
>+static void update_float_stats(VolDetectContext *vd, float *audio_data)
>+{
>+    double sample;
>+    int idx;
>+    if(!isnormal(*audio_data))
>+        return;

Do we really need to classify floats here? That's probably going to hurt perfs badly, and makes an otherwise very vectorisable function not so easily vectored.

>+    sample = fabsf(*audio_data);
>+    if (sample > vd->max)
>+        vd->max = sample;
>+    vd->sum2 += sample * sample;
>+    idx = lrintf(floorf(logdb(sample * sample, AV_SAMPLE_FMT_FLT))) + MAX_DB_FLT;

You're recomputing the same value again, and you seem to be rounding twice in a row?

>+    vd->histogram[idx]++;
>+    vd->nb_samples++;
> }
> 
> static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
>@@ -51,18 +75,41 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
>     int nb_channels = samples->ch_layout.nb_channels;
>     int nb_planes   = nb_channels;
>     int plane, i;
>-    int16_t *pcm;
>+    int planar = 0;
> 
>-    if (!av_sample_fmt_is_planar(samples->format)) {
>-        nb_samples *= nb_channels;
>+    planar = av_sample_fmt_is_planar(samples->format);
>+    if (!planar)
>         nb_planes = 1;
>+    if (vd->is_float) {
>+        float *audio_data;
>+        for (plane = 0; plane < nb_planes; plane++) {
>+            audio_data = (float *)samples->extended_data[plane];
>+            for (i = 0; i < nb_samples; i++) {
>+                if (planar) {
>+                    update_float_stats(vd, &audio_data[i]);
>+                } else {
>+                    for (int j = 0; j < nb_channels; j++)
>+                        update_float_stats(vd, &audio_data[i * nb_channels + j]);
>+                }
>+            }
>+        }
>+    } else {
>+        int16_t *pcm;
>+        for (plane = 0; plane < nb_planes; plane++) {
>+            pcm = (int16_t *)samples->extended_data[plane];
>+            for (i = 0; i < nb_samples; i++) {
>+                if (planar) {
>+                    vd->histogram[pcm[i] + 0x8000]++;
>+                    vd->nb_samples++;
>+                } else {
>+                    for (int j = 0; j < nb_channels; j++) {
>+                        vd->histogram[pcm[i * nb_channels + j] + 0x8000]++;
>+                        vd->nb_samples++;
>+                    }
>+                }
>+            }
>+        }

Can't we pick the correct implementation (planar/packed and float/int) once and for all whilst configuring the filter?

>     }
>-    for (plane = 0; plane < nb_planes; plane++) {
>-        pcm = (int16_t *)samples->extended_data[plane];
>-        for (i = 0; i < nb_samples; i++)
>-            vd->histogram[pcm[i] + 0x8000]++;
>-    }
>-
>     return ff_filter_frame(inlink->dst->outputs[0], samples);
> }
> 
>@@ -73,6 +120,20 @@ static void print_stats(AVFilterContext *ctx)
>     uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
>     uint64_t histdb[MAX_DB + 1] = { 0 };
> 
>+    if (!vd->nb_samples)
>+        return;
>+    if (vd->is_float) {
>+        av_log(ctx, AV_LOG_INFO, "n_samples: %" PRId64 "\n", vd->nb_samples);
>+        av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(vd->sum2 / vd->nb_samples, AV_SAMPLE_FMT_FLT));
>+        av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -2.0*logdb(vd->max, AV_SAMPLE_FMT_FLT));
>+        for (i = 0; i < HISTOGRAM_SIZE_FLT && !vd->histogram[i]; i++);
>+        for (; i >= 0 && sum < vd->nb_samples / 1000; i++) {
>+            if (!vd->histogram[i])
>+                continue;
>+            av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %" PRId64 "\n", MAX_DB_FLT - i, vd->histogram[i]);
>+            sum += vd->histogram[i];
>+        }
>+    } else {
>     for (i = 0; i < 0x10000; i++)
>         nb_samples += vd->histogram[i];
>     av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);
>@@ -92,26 +153,61 @@ static void print_stats(AVFilterContext *ctx)
>         return;
>     power = (power + nb_samples_shift / 2) / nb_samples_shift;
>     av_assert0(power <= 0x8000 * 0x8000);
>-    av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power));
>+    av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb((double)power, AV_SAMPLE_FMT_S16));
> 
>     max_volume = 0x8000;
>     while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] &&
>                              !vd->histogram[0x8000 - max_volume])
>         max_volume--;
>-    av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume));
>+    av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb((double)(max_volume * max_volume), AV_SAMPLE_FMT_S16));
> 
>     for (i = 0; i < 0x10000; i++)
>-        histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i];
>+        histdb[(int)logdb((double)(i - 0x8000) * (i - 0x8000), AV_SAMPLE_FMT_S16)] += vd->histogram[i];
>     for (i = 0; i <= MAX_DB && !histdb[i]; i++);
>     for (; i <= MAX_DB && sum < nb_samples / 1000; i++) {
>-        av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]);
>+        av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", -i, histdb[i]);
>         sum += histdb[i];
>     }
>+    }
>+}
>+
>+static int config_output(AVFilterLink *outlink)
>+{
>+    AVFilterContext *ctx = outlink->src;
>+    VolDetectContext *vd = ctx->priv;
>+    size_t histogram_size;
>+
>+    vd->is_float = outlink->format == AV_SAMPLE_FMT_FLT ||
>+                   outlink->format == AV_SAMPLE_FMT_FLTP;
>+
>+    if (!vd->is_float) {
>+        /*
>+        * Number of samples at each PCM value.
>+        * Only used for integer formats.
>+        * For 16 bit signed PCM there are 65536.
>+        * histogram[0x8000 + i] is the number of samples at value i.
>+        * The extra element is there for symmetry.
>+        */
>+        histogram_size = HISTOGRAM_SIZE + 1;
>+    } else {
>+        /*
>+        * The histogram is used to store the number of samples at each dB
>+        * instead of the number of samples at each PCM value.
>+        */
>+        histogram_size = HISTOGRAM_SIZE_FLT + 1;
>+    }
>+    vd->histogram = av_calloc(histogram_size, sizeof(uint64_t));
>+    if (!vd->histogram)
>+        return AVERROR(ENOMEM);
>+    return 0;
> }
> 
> static av_cold void uninit(AVFilterContext *ctx)
> {
>+    VolDetectContext *vd = ctx->priv;
>     print_stats(ctx);
>+    if (vd->histogram)
>+        av_freep(&vd->histogram);
> }
> 
> static const AVFilterPad volumedetect_inputs[] = {
>@@ -122,6 +218,14 @@ static const AVFilterPad volumedetect_inputs[] = {
>     },
> };
> 
>+static const AVFilterPad volumedetect_outputs[] = {
>+    {
>+        .name         = "default",
>+        .type         = AVMEDIA_TYPE_AUDIO,
>+        .config_props = config_output,
>+    },
>+};
>+
> const AVFilter ff_af_volumedetect = {
>     .name          = "volumedetect",
>     .description   = NULL_IF_CONFIG_SMALL("Detect audio volume."),
>@@ -129,6 +233,9 @@ const AVFilter ff_af_volumedetect = {
>     .uninit        = uninit,
>     .flags         = AVFILTER_FLAG_METADATA_ONLY,
>     FILTER_INPUTS(volumedetect_inputs),
>-    FILTER_OUTPUTS(ff_audio_default_filterpad),
>-    FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P),
>+    FILTER_OUTPUTS(volumedetect_outputs),
>+    FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_S16,
>+                      AV_SAMPLE_FMT_S16P,
>+                      AV_SAMPLE_FMT_FLT,
>+                      AV_SAMPLE_FMT_FLTP),
> };
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