[FFmpeg-cvslog] avfilter/af_silenceremove: add real peak detector
Paul B Mahol
git at videolan.org
Fri May 26 11:17:30 EEST 2023
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Thu May 25 23:06:50 2023 +0200| [2b5166addf9956f0617e6007bc02387cde9927dd] | committer: Paul B Mahol
avfilter/af_silenceremove: add real peak detector
Rename old peak detector to more correct name one.
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=2b5166addf9956f0617e6007bc02387cde9927dd
---
doc/filters.texi | 3 +-
libavfilter/af_silenceremove.c | 42 ++++++++++++++++-----
libavfilter/silenceremove_template.c | 71 ++++++++++++++++++++++++++++++++----
tests/fate/filter-audio.mak | 2 +-
4 files changed, 98 insertions(+), 20 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index 47b26fe92f..6f15b54d10 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -6461,8 +6461,7 @@ With @var{all}, only if all channels are detected as non-silence will cause
stopped trimming of silence.
@item detection
-Set how is silence detected. Can be @code{rms} or @code{peak}. Second is faster
-and works better with digital silence which is exactly 0.
+Set how is silence detected. Can be @code{avg}, @code{rms} or @code{peak}.
Default value is @code{rms}.
@item window
diff --git a/libavfilter/af_silenceremove.c b/libavfilter/af_silenceremove.c
index e0592c2368..28c972f86f 100644
--- a/libavfilter/af_silenceremove.c
+++ b/libavfilter/af_silenceremove.c
@@ -33,8 +33,10 @@
#include "internal.h"
enum SilenceDetect {
- D_PEAK,
+ D_AVG,
D_RMS,
+ D_PEAK,
+ D_NB
};
enum ThresholdMode {
@@ -75,6 +77,12 @@ typedef struct SilenceRemoveContext {
AVFrame *start_window;
AVFrame *stop_window;
+ int *start_front;
+ int *start_back;
+
+ int *stop_front;
+ int *stop_back;
+
int64_t window_duration;
int start_window_pos;
@@ -100,8 +108,8 @@ typedef struct SilenceRemoveContext {
int detection;
- float (*compute_flt)(float *c, float s, float ws, int size);
- double (*compute_dbl)(double *c, double s, double ws, int size);
+ float (*compute_flt)(float *c, float s, float ws, int size, int *front, int *back);
+ double (*compute_dbl)(double *c, double s, double ws, int size, int *front, int *back);
} SilenceRemoveContext;
#define OFFSET(x) offsetof(SilenceRemoveContext, x)
@@ -120,9 +128,10 @@ static const AVOption silenceremove_options[] = {
{ "stop_threshold", "set threshold for stop silence detection", OFFSET(stop_threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, DBL_MAX, AF },
{ "stop_silence", "set stop duration of silence part to keep", OFFSET(stop_silence_opt), AV_OPT_TYPE_DURATION, {.i64=0}, 0, INT32_MAX, AF },
{ "stop_mode", "set which channel will trigger trimming from end", OFFSET(stop_mode), AV_OPT_TYPE_INT, {.i64=T_ANY}, T_ANY, T_ALL, AF, "mode" },
- { "detection", "set how silence is detected", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=D_RMS}, D_PEAK,D_RMS, AF, "detection" },
- { "peak", "use absolute values of samples", 0, AV_OPT_TYPE_CONST, {.i64=D_PEAK},0, 0, AF, "detection" },
- { "rms", "use squared values of samples", 0, AV_OPT_TYPE_CONST, {.i64=D_RMS}, 0, 0, AF, "detection" },
+ { "detection", "set how silence is detected", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=D_RMS}, 0, D_NB-1, AF, "detection" },
+ { "avg", "use mean absolute values of samples", 0, AV_OPT_TYPE_CONST, {.i64=D_AVG}, 0, 0, AF, "detection" },
+ { "rms", "use root mean squared values of samples", 0, AV_OPT_TYPE_CONST, {.i64=D_RMS}, 0, 0, AF, "detection" },
+ { "peak", "use max absolute values of samples", 0, AV_OPT_TYPE_CONST, {.i64=D_PEAK},0, 0, AF, "detection" },
{ "window", "set duration of window for silence detection", OFFSET(window_duration_opt), AV_OPT_TYPE_DURATION, {.i64=20000}, 0, 100000000, AF },
{ NULL }
};
@@ -201,7 +210,9 @@ static int config_output(AVFilterLink *outlink)
s->start_window = ff_get_audio_buffer(outlink, s->window_duration);
s->stop_window = ff_get_audio_buffer(outlink, s->window_duration);
- if (!s->start_window || !s->stop_window)
+ s->start_cache = av_calloc(outlink->ch_layout.nb_channels, s->window_duration * sizeof(*s->start_cache));
+ s->stop_cache = av_calloc(outlink->ch_layout.nb_channels, s->window_duration * sizeof(*s->stop_cache));
+ if (!s->start_window || !s->stop_window || !s->start_cache || !s->stop_cache)
return AVERROR(ENOMEM);
s->start_queuef = ff_get_audio_buffer(outlink, s->start_silence + 1);
@@ -209,14 +220,20 @@ static int config_output(AVFilterLink *outlink)
if (!s->start_queuef || !s->stop_queuef)
return AVERROR(ENOMEM);
- s->start_cache = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->start_cache));
- s->stop_cache = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->stop_cache));
- if (!s->start_cache || !s->stop_cache)
+ s->start_front = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->start_front));
+ s->start_back = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->start_back));
+ s->stop_front = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->stop_front));
+ s->stop_back = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->stop_back));
+ if (!s->start_front || !s->start_back || !s->stop_front || !s->stop_back)
return AVERROR(ENOMEM);
clear_windows(s);
switch (s->detection) {
+ case D_AVG:
+ s->compute_flt = compute_avg_flt;
+ s->compute_dbl = compute_avg_dbl;
+ break;
case D_PEAK:
s->compute_flt = compute_peak_flt;
s->compute_dbl = compute_peak_dbl;
@@ -374,8 +391,13 @@ static av_cold void uninit(AVFilterContext *ctx)
av_frame_free(&s->stop_window);
av_frame_free(&s->start_queuef);
av_frame_free(&s->stop_queuef);
+
av_freep(&s->start_cache);
av_freep(&s->stop_cache);
+ av_freep(&s->start_front);
+ av_freep(&s->start_back);
+ av_freep(&s->stop_front);
+ av_freep(&s->stop_back);
}
static const AVFilterPad silenceremove_inputs[] = {
diff --git a/libavfilter/silenceremove_template.c b/libavfilter/silenceremove_template.c
index 1a12435ee6..ef63ea1e7e 100644
--- a/libavfilter/silenceremove_template.c
+++ b/libavfilter/silenceremove_template.c
@@ -99,8 +99,8 @@ static void fn(queue_sample)(AVFilterContext *ctx,
*window_pos = 0;
}
-static ftype fn(compute_peak)(ftype *cache, ftype sample, ftype wsample,
- int window_size)
+static ftype fn(compute_avg)(ftype *cache, ftype sample, ftype wsample,
+ int window_size, int *unused, int *unused2)
{
ftype r;
@@ -111,8 +111,49 @@ static ftype fn(compute_peak)(ftype *cache, ftype sample, ftype wsample,
return r / window_size;
}
+static ftype fn(compute_peak)(ftype *peak, ftype sample, ftype wsample,
+ int size, int *ffront, int *bback)
+{
+ ftype r, abs_sample = FABS(sample);
+ int front = *ffront;
+ int back = *bback;
+
+ if (front != back && abs_sample > peak[front]) {
+ while (front != back) {
+ front--;
+ if (front < 0)
+ front = size - 1;
+ }
+ }
+
+ while (front != back && abs_sample > peak[back]) {
+ back++;
+ if (back >= size)
+ back = 0;
+ }
+
+ if (front != back && FABS(wsample) == peak[front]) {
+ front--;
+ if (front < 0)
+ front = size - 1;
+ }
+
+ back--;
+ if (back < 0)
+ back = size - 1;
+ av_assert2(back != front);
+ peak[back] = abs_sample;
+
+ r = peak[front];
+
+ *ffront = front;
+ *bback = back;
+
+ return r;
+}
+
static ftype fn(compute_rms)(ftype *cache, ftype sample, ftype wsample,
- int window_size)
+ int window_size, int *unused, int *unused2)
{
ftype r;
@@ -143,6 +184,9 @@ static void fn(filter_start)(AVFilterContext *ctx,
const int start_duration = s->start_duration;
ftype *start_cache = (ftype *)s->start_cache;
const int start_silence = s->start_silence;
+ int window_size = start_window_nb_samples;
+ int *front = s->start_front;
+ int *back = s->start_back;
fn(queue_sample)(ctx, src, start,
&s->start_queue_pos,
@@ -153,15 +197,20 @@ static void fn(filter_start)(AVFilterContext *ctx,
start_nb_samples,
start_window_nb_samples);
+ if (s->detection != D_PEAK)
+ window_size = s->start_window_size;
+
for (int ch = 0; ch < nb_channels; ch++) {
ftype start_sample = start[start_pos + ch];
ftype start_ow = startw[start_wpos + ch];
ftype tstart;
- tstart = fn(s->compute)(start_cache + ch,
+ tstart = fn(s->compute)(start_cache + ch * start_window_nb_samples,
start_sample,
start_ow,
- s->start_window_size);
+ window_size,
+ front + ch,
+ back + ch);
startw[start_wpos + ch] = start_sample;
@@ -226,6 +275,9 @@ static void fn(filter_stop)(AVFilterContext *ctx,
ftype *stop_cache = (ftype *)s->stop_cache;
const int stop_silence = s->stop_silence;
const int restart = s->restart;
+ int window_size = stop_window_nb_samples;
+ int *front = s->stop_front;
+ int *back = s->stop_back;
fn(queue_sample)(ctx, src, stop,
&s->stop_queue_pos,
@@ -236,15 +288,20 @@ static void fn(filter_stop)(AVFilterContext *ctx,
stop_nb_samples,
stop_window_nb_samples);
+ if (s->detection != D_PEAK)
+ window_size = s->stop_window_size;
+
for (int ch = 0; ch < nb_channels; ch++) {
ftype stop_sample = stop[stop_pos + ch];
ftype stop_ow = stopw[stop_wpos + ch];
ftype tstop;
- tstop = fn(s->compute)(stop_cache + ch,
+ tstop = fn(s->compute)(stop_cache + ch * stop_window_nb_samples,
stop_sample,
stop_ow,
- s->stop_window_size);
+ window_size,
+ front + ch,
+ back + ch);
stopw[stop_wpos + ch] = stop_sample;
diff --git a/tests/fate/filter-audio.mak b/tests/fate/filter-audio.mak
index eff32b9f81..445c0f9217 100644
--- a/tests/fate/filter-audio.mak
+++ b/tests/fate/filter-audio.mak
@@ -184,7 +184,7 @@ fate-filter-pan-downmix2: SRC = $(TARGET_PATH)/tests/data/asynth-44100-11.wav
fate-filter-pan-downmix2: CMD = framecrc -ss 3.14 -i $(SRC) -frames:a 20 -filter:a "pan=5C|c0=0.7*c0+0.7*c10|c1=c9|c2=c8|c3=c7|c4=c6"
FATE_AFILTER-$(call ALLYES, LAVFI_INDEV, AEVALSRC_FILTER SILENCEREMOVE_FILTER) += fate-filter-silenceremove
-fate-filter-silenceremove: CMD = framecrc -auto_conversion_filters -f lavfi -i "aevalsrc=between(t\,1\,2)+between(t\,4\,5)+between(t\,7\,9):d=10:n=8192,silenceremove=start_periods=0:start_duration=0:start_threshold=0:stop_periods=-1:stop_duration=0:stop_threshold=-90dB:window=0:detection=peak"
+fate-filter-silenceremove: CMD = framecrc -auto_conversion_filters -f lavfi -i "aevalsrc=between(t\,1\,2)+between(t\,4\,5)+between(t\,7\,9):d=10:n=8192,silenceremove=start_periods=0:start_duration=0:start_threshold=0:stop_periods=-1:stop_duration=0:stop_threshold=-90dB:window=0:detection=avg"
FATE_AFILTER_SAMPLES-$(call FILTERDEMDECENCMUX, STEREOTOOLS, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-stereotools
fate-filter-stereotools: SRC = $(TARGET_SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
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