Fixes #9613
---
libavfilter/af_volumedetect.c | 234 +++++++++++++++++++++++++---------
1 file changed, 172 insertions(+), 62 deletions(-)
diff --git a/libavfilter/af_volumedetect.c b/libavfilter/af_volumedetect.c
index 8b001d1cf2..d13d043f88 100644
--- a/libavfilter/af_volumedetect.c
+++ b/libavfilter/af_volumedetect.c
@@ -24,94 +24,193 @@
#include "avfilter.h"
#include "internal.h"
+#define NOISE_FLOOR_DB_FLT -758
+#define MAX_DB_FLT 770
+#define MAX_DB 91
+#define HISTOGRAM_SIZE 0x10000
+
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(double v, enum AVSampleFormat sample_fmt)
+{
+ /*
+ * Since it is a not a power value, able to use 20.0 * log10(v)
+ */
+ if (sample_fmt == AV_SAMPLE_FMT_FLT) {
+ if (!v)
+ return MAX_DB_FLT;
+ return 20.0 * log10(v);
+ } 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 max_sample;
+ max_sample = fabsf(*audio_data);
+ if (max_sample > vd->max)
+ vd->max = max_sample;
+ vd->sum2 += *audio_data * *audio_data;
+ vd->histogram[(int)logdb(max_sample, AV_SAMPLE_FMT_FLT) + MAX_DB_FLT]++;
+ vd->nb_samples++;
+}
+
static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
{
AVFilterContext *ctx = inlink->dst;
VolDetectContext *vd = ctx->priv;
- int nb_samples = samples->nb_samples;
int nb_channels = samples->ch_layout.nb_channels;
int nb_planes = nb_channels;
+ int planar = 0;
int plane, i;
- int16_t *pcm;
- 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 < samples->nb_samples; i++) {
+ /*
+ * If the input is planar, the samples are in the seperated
planes.
+ * if the input is not planar, the samples are interleaved.
+ * if the input is not planar, split the samples into the
planes.
+ */
+ 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 < samples->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);
}
-#define MAX_DB 91
-
-static inline double logdb(uint64_t v)
+static void print_stats(AVFilterContext *ctx)
{
- double d = v / (double)(0x8000 * 0x8000);
- if (!v)
- return MAX_DB;
- return -log10(d) * 10;
+ VolDetectContext *vd = ctx->priv;
+
+ if (!vd->nb_samples)
+ return;
+ if (vd->is_float) {
+ double rms;
+ int i, sum = 0;
+ av_log(ctx, AV_LOG_INFO, "n_samples: %" PRId64 "\n", vd->nb_samples);
+ rms = sqrt(vd->sum2 / vd->nb_samples);
+ av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", logdb(rms,
AV_SAMPLE_FMT_FLT));
+ av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", logdb(vd->max,
AV_SAMPLE_FMT_FLT));
+ for (i = MAX_DB_FLT - NOISE_FLOOR_DB_FLT; i >= 0 && !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 {
+ int i, max_volume, shift;
+ uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
+ uint64_t histdb[MAX_DB + 1] = {0};
+ for (i = 0; i < 0x10000; i++)
+ nb_samples += vd->histogram[i];
+ av_log(ctx, AV_LOG_INFO, "n_samples: %" PRId64 "\n", nb_samples);
+ /*
+ * If nb_samples > 1<<34, there is a risk of overflow in the
+ * multiplication or the sum: shift all histogram values to avoid
that.
+ * The total number of samples must be recomputed to avoid rounding
+ * errors.
+ */
+ shift = av_log2(nb_samples >> 33);
+ for (i = 0; i < 0x10000; i++) {
+ nb_samples_shift += vd->histogram[i] >> shift;
+ power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift);
+ }
+ if (!nb_samples_shift)
+ 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((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((double)(max_volume * max_volume), AV_SAMPLE_FMT_S16));
+ for (i = 0; i < 0x10000; 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]);
+ sum += histdb[i];
+ }
+ }
}
-static void print_stats(AVFilterContext *ctx)
+static int config_output(AVFilterLink *outlink)
{
+ AVFilterContext *ctx = outlink->src;
VolDetectContext *vd = ctx->priv;
- int i, max_volume, shift;
- uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
- uint64_t histdb[MAX_DB + 1] = { 0 };
-
- for (i = 0; i < 0x10000; i++)
- nb_samples += vd->histogram[i];
- av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);
- if (!nb_samples)
- return;
- /* If nb_samples > 1<<34, there is a risk of overflow in the
- multiplication or the sum: shift all histogram values to avoid that.
- The total number of samples must be recomputed to avoid rounding
- errors. */
- shift = av_log2(nb_samples >> 33);
- for (i = 0; i < 0x10000; i++) {
- nb_samples_shift += vd->histogram[i] >> shift;
- power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift);
- }
- if (!nb_samples_shift)
- 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));
-
- 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));
-
- for (i = 0; i < 0x10000; i++)
- histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += 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]);
- sum += histdb[i];
+ 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.
+ */
+ vd->histogram = av_calloc(HISTOGRAM_SIZE + 1, sizeof(uint64_t));
+ if (!vd->histogram)
+ return AVERROR(ENOMEM);
+ } else {
+ /*
+ * The histogram is used to store the number of samples at each dB
+ * instead of the number of samples at each PCM value.
+ * The range of dB is from -758 to 770.
+ */
+ vd->histogram = av_calloc(MAX_DB_FLT - NOISE_FLOOR_DB_FLT + 1,
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 +221,14 @@ 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 +236,9 @@ 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
> On Mar 23, 2024, at 6:21 PM, Paul B Mahol <one...@gmail.com> wrote:
>
> On Sat, Mar 23, 2024 at 3:28 PM Yiğithan Yiğit <yigithanyigi...@gmail.com
> <mailto:yigithanyigi...@gmail.com>>
> wrote:
>
>> Hi,
>>
>> According to your advices, I made some changes of mine last patch. I feel
>> like this one way more better. I removed trivial calculations but I want to
>> say I am not proud of how I handled histogram in float despite 16 bit
>> integer histogram. I am storing dB values instead of storing samples. I
>> feel this one is more convenient. Still I am open to advices.
>>
>>
> I see no patch at all.
>
> volumedetect displays histogram typically with 1dB steps, so build
> histogram with 1dB range between each bin.
> for float, only use normal values, no +inf/subnormals/nans etc.
> I bet there is less than current 2^16 entries in histogram table of filter
> context to fill.
> There is no need to convert each input sample to dB scale. Just to
> calculate ranges for each 1dB entry calculate range in linear space and
> every such sample that is in such range get added to such histogram bin
> entry.
> Or if you calculate in dB scale anyway than just round(ceilf/floorf/lrintf)
> dB value (removing fractional parts) and add it into histogram table, do
> not forget to count >+/-1.0 values too. (ones with >0dB values), you can
> use normal mean/max/peak calculations (do not use histogram to calculate
> them for float/double).
>
>
>
>> Thank you.
>> Yigithan
>>
>>
>> 
>>
>>> On Mar 21, 2024, at 11:30 PM, Paul B Mahol <one...@gmail.com
>>> <mailto:one...@gmail.com>> wrote:
>>>
>>> On Wed, Mar 20, 2024 at 11:55 PM Yiğithan Yiğit <
>> yigithanyigi...@gmail.com <mailto:yigithanyigi...@gmail.com>
>> <mailto:yigithanyigi...@gmail.com>>
>>> wrote:
>>>
>>>>
>>>>> On Mar 21, 2024, at 12:10 AM, Paul B Mahol <one...@gmail.com
>>>>> <mailto:one...@gmail.com>> wrote:
>>>>>
>>>>> Why? This is pointless.
>>>>>
>>>>> volumedetect have histogram output, float patch does not have it at
>> all.
>>>>> Use astats filter.
>>>>>
>>>>> On Wed, Mar 20, 2024 at 9:47 PM Yiğithan Yiğit <
>>>> yigithanyigi...@gmail.com>
>>>>> wrote:
>>>>>
>>>>>> _______________________________________________
>>>>>> ffmpeg-devel mailing list
>>>>>> ffmpeg-devel@ffmpeg.org
>>>>>> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>>>>>>
>>>>>> To unsubscribe, visit link above, or email
>>>>>> ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
>>>>>>
>>>>> _______________________________________________
>>>>> ffmpeg-devel mailing list
>>>>> ffmpeg-devel@ffmpeg.org
>>>>> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>>>>>
>>>>> To unsubscribe, visit link above, or email
>>>>> ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe”.
>>>>
>>>> I am a beginner/student also new at open source but I love FFmpeg and
>>>> using in my daily life. From my perspective volumedetect way more user
>>>> friendly. I believe adding this patch would be useful to people such as
>>>> #9613. The reason lack of histogram output for float mostly for my
>>>> indecision about range of the histogram. I am open the suggestions and
>>>> after that I can make a new patch.
>>>>
>>>
>>> It is trivial (to some people) to add histogram per dB for float/double
>>> inputs.
>>> But this patch just does some extremely trivial math calculations so that
>>> float input have completely different output from integer ones.
>>> That is very odd and unfriendly from my perspective.
>>>
>>> Besides if you only interested in discrete sample audio peak finder in
>>> audio input use astats and measure_overall=Peak_level options.
>>> Yes they are not default on. Because more statistics are more important
>>> than single number.
>>>
>>> I'm not against adding proper and useful and correct float/double support
>>> to volumedetect, but it needs to have same/similar structure of output as
>>> integer sample format input audio, otherwise it just looks lazy and prone
>>> for users wondering what is going on when they use different sample
>> formats
>>> in theirs graphs.
>>>
>>>
>>>>
>>>> Best Regards
>>>> Yigithan
>>>>
>>>>
>>>> _______________________________________________
>>>> ffmpeg-devel mailing list
>>>> ffmpeg-devel@ffmpeg.org <mailto:ffmpeg-devel@ffmpeg.org>
>>>> <mailto:ffmpeg-devel@ffmpeg.org>
>>>> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>>>>
>>>> To unsubscribe, visit link above, or email
>>>> ffmpeg-devel-requ...@ffmpeg.org <mailto:ffmpeg-devel-requ...@ffmpeg.org>
>>>> <mailto:ffmpeg-devel-requ...@ffmpeg.org>
>> with subject "unsubscribe".
>>>>
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