[FFmpeg-cvslog] avfilter/af_alimiter: add 3 more options

[Paul B Mahol]

ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sun Dec  6 14:57:05 2015 +0100| [45b451c89239b85aec02f946aca4b67894a027b2] | committer: Paul B Mahol

avfilter/af_alimiter: add 3 more options

Signed-off-by: Paul B Mahol <onemda at gmail.com>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=45b451c89239b85aec02f946aca4b67894a027b2
---

 doc/filters.texi          |   13 +++++++++++--
 libavfilter/af_alimiter.c |   23 ++++++++++++++++-------
 2 files changed, 27 insertions(+), 9 deletions(-)

diff --git a/doc/filters.texi b/doc/filters.texi
index 1aa6841..4b8e8de 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -833,9 +833,14 @@ that the delay it produces is the attack time you set.
 The filter accepts the following options:
 
 @table @option
+ at item level_in
+Set input gain. Default is 1.
+
+ at item level_out
+Set output gain. Default is 1.
+
 @item limit
-Don't let signals above this level pass the limiter. The removed amplitude is
-added automatically. Default is 1.
+Don't let signals above this level pass the limiter. Default is 1.
 
 @item attack
 The limiter will reach its attenuation level in this amount of time in
@@ -853,6 +858,10 @@ time.
 @item asc_level
 Select how much the release time is affected by ASC, 0 means nearly no changes
 in release time while 1 produces higher release times.
+
+ at item level
+Auto level output signal. Default is enabled.
+This normalizes audio back to 0dB if enabled.
 @end table
 
 Depending on picked setting it is recommended to upsample input 2x or 4x times
diff --git a/libavfilter/af_alimiter.c b/libavfilter/af_alimiter.c
index c3e3ff2..46211a7 100644
--- a/libavfilter/af_alimiter.c
+++ b/libavfilter/af_alimiter.c
@@ -41,7 +41,10 @@ typedef struct AudioLimiterContext {
     double attack;
     double release;
     double att;
+    double level_in;
+    double level_out;
     int auto_release;
+    int auto_level;
     double asc;
     int asc_c;
     int asc_pos;
@@ -64,11 +67,14 @@ typedef struct AudioLimiterContext {
 #define F AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption alimiter_options[] = {
-    { "limit",     "set limit",     OFFSET(limit),        AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625,    1, A|F },
-    { "attack",    "set attack",    OFFSET(attack),       AV_OPT_TYPE_DOUBLE, {.dbl=5},    0.1,   80, A|F },
-    { "release",   "set release",   OFFSET(release),      AV_OPT_TYPE_DOUBLE, {.dbl=50},     1, 8000, A|F },
-    { "asc",       "enable asc",    OFFSET(auto_release), AV_OPT_TYPE_BOOL,   {.i64=0},      0,    1, A|F },
-    { "asc_level", "set asc level", OFFSET(asc_coeff),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5},    0,    1, A|F },
+    { "level_in",  "set input level",  OFFSET(level_in),     AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625,   64, A|F },
+    { "level_out", "set output level", OFFSET(level_out),    AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625,   64, A|F },
+    { "limit",     "set limit",        OFFSET(limit),        AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625,    1, A|F },
+    { "attack",    "set attack",       OFFSET(attack),       AV_OPT_TYPE_DOUBLE, {.dbl=5},    0.1,   80, A|F },
+    { "release",   "set release",      OFFSET(release),      AV_OPT_TYPE_DOUBLE, {.dbl=50},     1, 8000, A|F },
+    { "asc",       "enable asc",       OFFSET(auto_release), AV_OPT_TYPE_BOOL,   {.i64=0},      0,    1, A|F },
+    { "asc_level", "set asc level",    OFFSET(asc_coeff),    AV_OPT_TYPE_DOUBLE, {.dbl=0.5},    0,    1, A|F },
+    { "level",     "auto level",       OFFSET(auto_level),   AV_OPT_TYPE_BOOL,   {.i64=1},      0,    1, A|F },
     { NULL }
 };
 
@@ -118,6 +124,9 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
     const double release = s->release;
     const double limit = s->limit;
     double *nextdelta = s->nextdelta;
+    double level = s->auto_level ? 1 / limit : 1;
+    const double level_out = s->level_out;
+    const double level_in = s->level_in;
     int *nextpos = s->nextpos;
     AVFrame *out;
     double *buf;
@@ -139,7 +148,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
         double peak = 0;
 
         for (c = 0; c < channels; c++) {
-            double sample = src[c];
+            double sample = src[c] * level_in;
 
             buffer[s->pos + c] = sample;
             peak = FFMAX(peak, fabs(sample));
@@ -255,7 +264,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
             s->delta = 0.;
 
         for (c = 0; c < channels; c++)
-            dst[c] = av_clipd(dst[c], -limit, limit);
+            dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
 
         s->pos = (s->pos + channels) % buffer_size;
         src += channels;