[FFmpeg-devel] [PATCH 2/3] avfilter/af_volumedetect.c: Add 32bit float audio support
Rémi Denis-Courmont
remi at remlab.net
Mon Jun 17 17:52:39 EEST 2024
Le 17 juin 2024 13:18:11 GMT+02:00, Yigithan Yigit <yigithanyigitdevel at 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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