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ffmpeg.wasm-core
Commit 4f226714 authored by Paul B Mahol's avatar Paul B Mahol
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avfilter/af_astats: add support for other sample formats

parent b746ed70
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Showing with 89 additions and 25 deletions
libavfilter/af_astats.c
View file @ 4f226714
......@@ -31,6 +31,7 @@ typedef struct ChannelStats {
double sigma_x, sigma_x2;
double avg_sigma_x2, min_sigma_x2, max_sigma_x2;
double min, max;
double nmin, nmax;
double min_run, max_run;
double min_runs, max_runs;
double min_diff, max_diff;
......@@ -50,6 +51,7 @@ typedef struct {
int metadata;
int reset_count;
int nb_frames;
int maxbitdepth;
} AudioStatsContext;
#define OFFSET(x) offsetof(AudioStatsContext, x)
......@@ -69,6 +71,9 @@ static int query_formats(AVFilterContext *ctx)
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
......@@ -101,8 +106,8 @@ static void reset_stats(AudioStatsContext *s)
for (c = 0; c < s->nb_channels; c++) {
ChannelStats *p = &s->chstats[c];
p->min = p->min_sigma_x2 = DBL_MAX;
p->max = p->max_sigma_x2 = DBL_MIN;
p->min = p->nmin = p->min_sigma_x2 = DBL_MAX;
p->max = p->nmax = p->max_sigma_x2 = DBL_MIN;
p->min_diff = DBL_MAX;
p->max_diff = DBL_MIN;
p->sigma_x = 0;
......@@ -134,15 +139,16 @@ static int config_output(AVFilterLink *outlink)
s->mult = exp((-1 / s->time_constant / outlink->sample_rate));
s->tc_samples = 5 * s->time_constant * outlink->sample_rate + .5;
s->nb_frames = 0;
s->maxbitdepth = av_get_bytes_per_sample(outlink->format) * 8;
reset_stats(s);
return 0;
}
static void bit_depth(uint64_t mask, uint64_t imask, AVRational *depth)
static void bit_depth(AudioStatsContext *s, uint64_t mask, uint64_t imask, AVRational *depth)
{
unsigned result = 64;
unsigned result = s->maxbitdepth;
mask = mask & (~imask);
......@@ -156,10 +162,11 @@ static void bit_depth(uint64_t mask, uint64_t imask, AVRational *depth)
depth->num++;
}
static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d, double nd, int64_t i)
{
if (d < p->min) {
p->min = d;
p->nmin = nd;
p->min_run = 1;
p->min_runs = 0;
p->min_count = 1;
......@@ -172,6 +179,7 @@ static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
if (d > p->max) {
p->max = d;
p->nmax = nd;
p->max_run = 1;
p->max_runs = 0;
p->max_count = 1;
......@@ -182,15 +190,15 @@ static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d)
p->max_runs += p->max_run * p->max_run;
}
p->sigma_x += d;
p->sigma_x2 += d * d;
p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * d * d;
p->sigma_x += nd;
p->sigma_x2 += nd * nd;
p->avg_sigma_x2 = p->avg_sigma_x2 * s->mult + (1.0 - s->mult) * nd * nd;
p->min_diff = FFMIN(p->min_diff, fabs(d - p->last));
p->max_diff = FFMAX(p->max_diff, fabs(d - p->last));
p->diff1_sum += fabs(d - p->last);
p->last = d;
p->mask |= llrint(d * (UINT64_C(1) << 63));
p->imask &= llrint(d * (UINT64_C(1) << 63));
p->mask |= i;
p->imask &= i;
if (p->nb_samples >= s->tc_samples) {
p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2);
......@@ -220,6 +228,7 @@ static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
uint64_t mask = 0, imask = 0xFFFFFFFFFFFFFFFF, min_count = 0, max_count = 0, nb_samples = 0;
double min_runs = 0, max_runs = 0,
min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,
nmin = DBL_MAX, nmax = DBL_MIN,
max_sigma_x = 0,
diff1_sum = 0,
sigma_x = 0,
......@@ -237,6 +246,8 @@ static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
min = FFMIN(min, p->min);
max = FFMAX(max, p->max);
nmin = FFMIN(nmin, p->nmin);
nmax = FFMAX(nmax, p->nmax);
min_diff = FFMIN(min_diff, p->min_diff);
max_diff = FFMAX(max_diff, p->max_diff);
diff1_sum += p->diff1_sum,
......@@ -260,14 +271,14 @@ static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
set_meta(metadata, c + 1, "Min_difference", "%f", p->min_diff);
set_meta(metadata, c + 1, "Max_difference", "%f", p->max_diff);
set_meta(metadata, c + 1, "Mean_difference", "%f", p->diff1_sum / (p->nb_samples - 1));
set_meta(metadata, c + 1, "Peak_level", "%f", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
set_meta(metadata, c + 1, "Peak_level", "%f", LINEAR_TO_DB(FFMAX(-p->nmin, p->nmax)));
set_meta(metadata, c + 1, "RMS_level", "%f", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
set_meta(metadata, c + 1, "RMS_peak", "%f", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
set_meta(metadata, c + 1, "RMS_trough", "%f", LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
set_meta(metadata, c + 1, "Crest_factor", "%f", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);
set_meta(metadata, c + 1, "Flat_factor", "%f", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));
set_meta(metadata, c + 1, "Peak_count", "%f", (float)(p->min_count + p->max_count));
bit_depth(p->mask, p->imask, &depth);
bit_depth(s, p->mask, p->imask, &depth);
set_meta(metadata, c + 1, "Bit_depth", "%f", depth.num);
set_meta(metadata, c + 1, "Bit_depth2", "%f", depth.den);
}
......@@ -278,13 +289,13 @@ static void set_metadata(AudioStatsContext *s, AVDictionary **metadata)
set_meta(metadata, 0, "Overall.Min_difference", "%f", min_diff);
set_meta(metadata, 0, "Overall.Max_difference", "%f", max_diff);
set_meta(metadata, 0, "Overall.Mean_difference", "%f", diff1_sum / (nb_samples - s->nb_channels));
set_meta(metadata, 0, "Overall.Peak_level", "%f", LINEAR_TO_DB(FFMAX(-min, max)));
set_meta(metadata, 0, "Overall.Peak_level", "%f", LINEAR_TO_DB(FFMAX(-nmin, nmax)));
set_meta(metadata, 0, "Overall.RMS_level", "%f", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
set_meta(metadata, 0, "Overall.RMS_peak", "%f", LINEAR_TO_DB(sqrt(max_sigma_x2)));
set_meta(metadata, 0, "Overall.RMS_trough", "%f", LINEAR_TO_DB(sqrt(min_sigma_x2)));
set_meta(metadata, 0, "Overall.Flat_factor", "%f", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
set_meta(metadata, 0, "Overall.Peak_count", "%f", (float)(min_count + max_count) / (double)s->nb_channels);
bit_depth(mask, imask, &depth);
bit_depth(s, mask, imask, &depth);
set_meta(metadata, 0, "Overall.Bit_depth", "%f", depth.num);
set_meta(metadata, 0, "Overall.Bit_depth2", "%f", depth.den);
set_meta(metadata, 0, "Overall.Number_of_samples", "%f", nb_samples / s->nb_channels);
......@@ -295,7 +306,6 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
AudioStatsContext *s = inlink->dst->priv;
AVDictionary **metadata = avpriv_frame_get_metadatap(buf);
const int channels = s->nb_channels;
const double *src;
int i, c;
if (s->reset_count > 0) {
......@@ -310,20 +320,71 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
case AV_SAMPLE_FMT_DBLP:
for (c = 0; c < channels; c++) {
ChannelStats *p = &s->chstats[c];
src = (const double *)buf->extended_data[c];
const double *src = (const double *)buf->extended_data[c];
for (i = 0; i < buf->nb_samples; i++, src++)
update_stat(s, p, *src);
update_stat(s, p, *src, *src, llrint(*src * (UINT64_C(1) << 63)));
}
break;
case AV_SAMPLE_FMT_DBL:
src = (const double *)buf->extended_data[0];
case AV_SAMPLE_FMT_DBL: {
const double *src = (const double *)buf->extended_data[0];
for (i = 0; i < buf->nb_samples; i++) {
for (c = 0; c < channels; c++, src++)
update_stat(s, &s->chstats[c], *src);
update_stat(s, &s->chstats[c], *src, *src, llrint(*src * (UINT64_C(1) << 63)));
}}
break;
case AV_SAMPLE_FMT_FLTP:
for (c = 0; c < channels; c++) {
ChannelStats *p = &s->chstats[c];
const float *src = (const float *)buf->extended_data[c];
for (i = 0; i < buf->nb_samples; i++, src++)
update_stat(s, p, *src, *src, llrint(*src * (UINT64_C(1) << 63)));
}
break;
case AV_SAMPLE_FMT_FLT: {
const float *src = (const float *)buf->extended_data[0];
for (i = 0; i < buf->nb_samples; i++) {
for (c = 0; c < channels; c++, src++)
update_stat(s, &s->chstats[c], *src, *src / (double)INT16_MAX, llrint(*src * (UINT64_C(1) << 63)));
}}
break;
case AV_SAMPLE_FMT_S32P:
for (c = 0; c < channels; c++) {
ChannelStats *p = &s->chstats[c];
const int32_t *src = (const int32_t *)buf->extended_data[c];
for (i = 0; i < buf->nb_samples; i++, src++)
update_stat(s, p, *src, *src / (double)INT32_MAX, *src);
}
break;
case AV_SAMPLE_FMT_S32: {
const int32_t *src = (const int32_t *)buf->extended_data[0];
for (i = 0; i < buf->nb_samples; i++) {
for (c = 0; c < channels; c++, src++)
update_stat(s, &s->chstats[c], *src, *src / (double)INT32_MAX, *src);
}}
break;
case AV_SAMPLE_FMT_S16P:
for (c = 0; c < channels; c++) {
ChannelStats *p = &s->chstats[c];
const int16_t *src = (const int16_t *)buf->extended_data[c];
for (i = 0; i < buf->nb_samples; i++, src++)
update_stat(s, p, *src, *src / (double)INT16_MAX, *src);
}
break;
case AV_SAMPLE_FMT_S16: {
const int16_t *src = (const int16_t *)buf->extended_data[0];
for (i = 0; i < buf->nb_samples; i++) {
for (c = 0; c < channels; c++, src++)
update_stat(s, &s->chstats[c], *src, *src / (double)INT16_MAX, *src);
}}
break;
}
if (s->metadata)
......@@ -338,6 +399,7 @@ static void print_stats(AVFilterContext *ctx)
uint64_t mask = 0, imask = 0xFFFFFFFFFFFFFFFF, min_count = 0, max_count = 0, nb_samples = 0;
double min_runs = 0, max_runs = 0,
min = DBL_MAX, max = DBL_MIN, min_diff = DBL_MAX, max_diff = 0,
nmin = DBL_MAX, nmax = DBL_MIN,
max_sigma_x = 0,
diff1_sum = 0,
sigma_x = 0,
......@@ -355,6 +417,8 @@ static void print_stats(AVFilterContext *ctx)
min = FFMIN(min, p->min);
max = FFMAX(max, p->max);
nmin = FFMIN(nmin, p->nmin);
nmax = FFMAX(nmax, p->nmax);
min_diff = FFMIN(min_diff, p->min_diff);
max_diff = FFMAX(max_diff, p->max_diff);
diff1_sum += p->diff1_sum,
......@@ -379,15 +443,15 @@ static void print_stats(AVFilterContext *ctx)
av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", p->min_diff);
av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", p->max_diff);
av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", p->diff1_sum / (p->nb_samples - 1));
av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->min, p->max)));
av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-p->nmin, p->nmax)));
av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(p->sigma_x2 / p->nb_samples)));
av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(p->max_sigma_x2)));
if (p->min_sigma_x2 != 1)
av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n",LINEAR_TO_DB(sqrt(p->min_sigma_x2)));
av_log(ctx, AV_LOG_INFO, "Crest factor: %f\n", p->sigma_x2 ? FFMAX(-p->min, p->max) / sqrt(p->sigma_x2 / p->nb_samples) : 1);
av_log(ctx, AV_LOG_INFO, "Crest factor: %f\n", p->sigma_x2 ? FFMAX(-p->nmin, p->nmax) / sqrt(p->sigma_x2 / p->nb_samples) : 1);
av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((p->min_runs + p->max_runs) / (p->min_count + p->max_count)));
av_log(ctx, AV_LOG_INFO, "Peak count: %"PRId64"\n", p->min_count + p->max_count);
bit_depth(p->mask, p->imask, &depth);
bit_depth(s, p->mask, p->imask, &depth);
av_log(ctx, AV_LOG_INFO, "Bit depth: %u/%u\n", depth.num, depth.den);
}
......@@ -398,14 +462,14 @@ static void print_stats(AVFilterContext *ctx)
av_log(ctx, AV_LOG_INFO, "Min difference: %f\n", min_diff);
av_log(ctx, AV_LOG_INFO, "Max difference: %f\n", max_diff);
av_log(ctx, AV_LOG_INFO, "Mean difference: %f\n", diff1_sum / (nb_samples - s->nb_channels));
av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-min, max)));
av_log(ctx, AV_LOG_INFO, "Peak level dB: %f\n", LINEAR_TO_DB(FFMAX(-nmin, nmax)));
av_log(ctx, AV_LOG_INFO, "RMS level dB: %f\n", LINEAR_TO_DB(sqrt(sigma_x2 / nb_samples)));
av_log(ctx, AV_LOG_INFO, "RMS peak dB: %f\n", LINEAR_TO_DB(sqrt(max_sigma_x2)));
if (min_sigma_x2 != 1)
av_log(ctx, AV_LOG_INFO, "RMS trough dB: %f\n", LINEAR_TO_DB(sqrt(min_sigma_x2)));
av_log(ctx, AV_LOG_INFO, "Flat factor: %f\n", LINEAR_TO_DB((min_runs + max_runs) / (min_count + max_count)));
av_log(ctx, AV_LOG_INFO, "Peak count: %f\n", (min_count + max_count) / (double)s->nb_channels);
bit_depth(mask, imask, &depth);
bit_depth(s, mask, imask, &depth);
av_log(ctx, AV_LOG_INFO, "Bit depth: %u/%u\n", depth.num, depth.den);
av_log(ctx, AV_LOG_INFO, "Number of samples: %"PRId64"\n", nb_samples / s->nb_channels);
}
......
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