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avfilter/af_astats: drop usage of histogram for noise floor calculation

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This commit is contained in:
Paul B Mahol 2023-08-13 12:54:16 +02:00
parent 80fdf51b8f
commit 90afcc2bfd
1 changed files with 80 additions and 26 deletions
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104
libavfilter/af_astats.c
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@ -86,8 +86,10 @@ typedef struct ChannelStats {
uint64_t nb_infs;
uint64_t nb_denormals;
double *win_samples;
uint64_t histogram[HISTOGRAM_SIZE];
double *sorted_samples;
uint64_t ehistogram[HISTOGRAM_SIZE];
int sorted_front;
int sorted_back;
int win_pos;
int max_index;
double noise_floor;
@ -191,9 +193,12 @@ static void reset_stats(AudioStatsContext *s)
p->noise_floor_count = 0;
p->entropy = 0;
p->win_pos = 0;
p->sorted_front = 0;
p->sorted_back = 0;
memset(p->win_samples, 0, s->tc_samples * sizeof(*p->win_samples));
memset(p->histogram, 0, sizeof(p->histogram));
memset(p->ehistogram, 0, sizeof(p->ehistogram));
for (int n = 0; n < s->tc_samples; n++)
p->sorted_samples[n] = -1.0;
}
}
@ -214,6 +219,10 @@ static int config_output(AVFilterLink *outlink)
p->win_samples = av_calloc(s->tc_samples, sizeof(*p->win_samples));
if (!p->win_samples)
return AVERROR(ENOMEM);
p->sorted_samples = av_calloc(s->tc_samples, sizeof(*p->sorted_samples));
if (!p->sorted_samples)
return AVERROR(ENOMEM);
}
s->mult = exp((-1 / s->time_constant / outlink->sample_rate));
@ -260,6 +269,63 @@ static double calc_entropy(AudioStatsContext *s, ChannelStats *p)
return -entropy / log2(HISTOGRAM_SIZE);
}
static double calc_noise_floor(double *ss, double x, double px,
int n, int *ffront, int *bback)
{
double r, ax = fabs(x);
int front = *ffront;
int back = *bback;
int empty = front == back && ss[front] == -1.0;
if (!empty && fabs(px) == ss[front]) {
ss[front] = -1.0;
if (back != front) {
front--;
if (front < 0)
front = n - 1;
}
empty = front == back;
}
if (!empty && ax >= ss[front]) {
while (1) {
ss[front] = -1.0;
if (back == front) {
empty = 1;
break;
}
front--;
if (front < 0)
front = n - 1;
}
}
while (!empty && ax >= ss[back]) {
ss[back] = -1.0;
if (back == front) {
empty = 1;
break;
}
back++;
if (back >= n)
back = 0;
}
if (!empty) {
back--;
if (back < 0)
back = n - 1;
}
ss[back] = ax;
r = ss[front];
*ffront = front;
*bback = back;
return r;
}
static inline void update_minmax(AudioStatsContext *s, ChannelStats *p, double d)
{
if (d < p->min)
@ -271,7 +337,7 @@ static inline void update_minmax(AudioStatsContext *s, ChannelStats *p, double d
static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d, double nd, int64_t i)
{
double abs_d = FFABS(d);
double drop;
double drop, noise_floor;
int index;
if (p->abs_peak < abs_d) {
@ -331,24 +397,21 @@ static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d,
p->win_samples[p->win_pos] = nd;
index = av_clip(lrint(av_clipd(FFABS(nd), 0.0, 1.0) * HISTOGRAM_MAX), 0, HISTOGRAM_MAX);
p->max_index = FFMAX(p->max_index, index);
p->histogram[index]++;
p->ehistogram[index]++;
if (!isnan(p->noise_floor))
p->histogram[av_clip(lrint(av_clipd(FFABS(drop), 0.0, 1.0) * HISTOGRAM_MAX), 0, HISTOGRAM_MAX)]--;
p->win_pos++;
while (p->histogram[p->max_index] == 0)
p->max_index--;
if (p->win_pos >= s->tc_samples || !isnan(p->noise_floor)) {
double noise_floor = 1.;
if (p->win_pos >= s->tc_samples)
p->win_pos = 0;
for (int i = p->max_index; i >= 0; i--) {
if (p->histogram[i]) {
noise_floor = i / (double)HISTOGRAM_MAX;
break;
}
if (p->nb_samples >= s->tc_samples) {
p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2);
p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2);
}
p->nb_samples++;
if (p->nb_samples >= s->tc_samples) {
noise_floor = calc_noise_floor(p->sorted_samples, nd, drop,
s->tc_samples, &p->sorted_front, &p->sorted_back);
if (isnan(p->noise_floor)) {
p->noise_floor = noise_floor;
p->noise_floor_count = 1;
@ -361,16 +424,6 @@ static inline void update_stat(AudioStatsContext *s, ChannelStats *p, double d,
}
}
}
if (p->win_pos >= s->tc_samples) {
p->win_pos = 0;
}
if (p->nb_samples >= s->tc_samples) {
p->max_sigma_x2 = FFMAX(p->max_sigma_x2, p->avg_sigma_x2);
p->min_sigma_x2 = FFMIN(p->min_sigma_x2, p->avg_sigma_x2);
}
p->nb_samples++;
}
static inline void update_float_stat(AudioStatsContext *s, ChannelStats *p, float d)
@ -847,6 +900,7 @@ static av_cold void uninit(AVFilterContext *ctx)
ChannelStats *p = &s->chstats[i];
av_freep(&p->win_samples);
av_freep(&p->sorted_samples);
}
}
av_freep(&s->chstats);