[FFmpeg-devel] [PATCH v2 2/3] avfilter/af_volumedetect.c: Add 32bit float audio support
Yigithan Yigit
yigithanyigitdevel at gmail.com
Tue May 14 03:55:15 EEST 2024
---
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;
+ 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;
+ 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++;
+ }
+ }
+ }
+ }
}
- 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),
};
--
2.44.0
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