[SCM] calf/master: Merge branch 'master' of ssh://repo.or.cz/srv/git/calf

[js at users.alioth.debian.org]

+ Flanger, Multi Chorus, Phaser, Vintage Delay: add dry amount knobs for Aux Bus type uses
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The following commit has been merged in the master branch:
commit 0c595c3dbb0365b7afaf6f4219121f565938f578
Merge: fd05dc021c79dbc0c3f1133003770dcc98f6ae70 1609fdc19bc20731c5bf9460e192caf5f91dc1da
Author: Thor Harald Johansen <thj at thj.no>
Date:   Thu Nov 27 21:19:22 2008 +0100

    Merge branch 'master' of ssh://repo.or.cz/srv/git/calf

diff --combined src/calf/modules.h
index 94a5934,4ad0936..4c46d70
--- a/src/calf/modules.h
+++ b/src/calf/modules.h
@@@ -78,7 -78,7 +78,7 @@@ public
      }
      void set_sample_rate(uint32_t sr);
      void params_changed() {
-         float dry = 1.0;
+         float dry = *params[par_dryamount];
          float wet = *params[par_amount];
          float rate = *params[par_rate]; // 0.01*pow(1000.0f,*params[par_rate]);
          float min_delay = *params[par_delay] / 1000.0;
@@@ -139,7 -139,7 +139,7 @@@ public
          is_active = false;
      }
      void params_changed() {
-         float dry = 1.0;
+         float dry = *params[par_dryamount];
          float wet = *params[par_amount];
          float rate = *params[par_rate]; // 0.01*pow(1000.0f,*params[par_rate]);
          float base_frq = *params[par_freq];
@@@ -372,6 -372,7 +372,7 @@@ public
      float buffers[2][MAX_DELAY];
      int bufptr, deltime_l, deltime_r, mixmode, medium, old_medium;
      gain_smoothing amt_left, amt_right, fb_left, fb_right;
+     float dry;
      
      dsp::biquad_d2<float> biquad_left[2], biquad_right[2];
      
@@@ -388,7 -389,8 +389,8 @@@
          deltime_l = dsp::fastf2i_drm(unit * *params[par_time_l]);
          deltime_r = dsp::fastf2i_drm(unit * *params[par_time_r]);
          amt_left.set_inertia(*params[par_amount]); amt_right.set_inertia(*params[par_amount]);
-         float fb = *params[par_feedback];;
+         float fb = *params[par_feedback];
+         dry = *params[par_dryamount];
          mixmode = dsp::fastf2i_drm(*params[par_mixmode]);
          medium = dsp::fastf2i_drm(*params[par_medium]);
          if (mixmode == 0)
@@@ -433,8 -435,8 +435,8 @@@
              float in_left = buffers[v][(bufptr - deltime_l) & ADDR_MASK], in_right = buffers[1 - v][(bufptr - deltime_r) & ADDR_MASK], out_left, out_right, del_left, del_right;
              dsp::sanitize(in_left), dsp::sanitize(in_right);
  
-             out_left = ins[0][i] + in_left * amt_left.get();
-             out_right = ins[1][i] + in_right * amt_right.get();
+             out_left = dry * ins[0][i] + in_left * amt_left.get();
+             out_right = dry * ins[1][i] + in_right * amt_right.get();
              del_left = ins[0][i] + in_left * fb_left.get();
              del_right = ins[1][i] + in_right * fb_right.get();
              
@@@ -645,7 -647,7 +647,7 @@@ public
      void params_changed()
      {
          // delicious copy-pasta from flanger module - it'd be better to keep it common or something
-         float dry = 1.0;
+         float dry = *params[par_dryamount];
          float wet = *params[par_amount];
          float rate = *params[par_rate];
          float min_delay = *params[par_delay] / 1000.0;
@@@ -669,10 -671,6 +671,6 @@@
      uint32_t process(uint32_t offset, uint32_t numsamples, uint32_t inputs_mask, uint32_t outputs_mask) {
          left.process(outs[0] + offset, ins[0] + offset, numsamples);
          right.process(outs[1] + offset, ins[1] + offset, numsamples);
-         if (params[par_lfophase_l])
-             *params[par_lfophase_l] = (double)left.lfo.phase * 360.0 / 4096.0;
-         if (params[par_lfophase_r])
-             *params[par_lfophase_r] = (double)right.lfo.phase * 360.0 / 4096.0;
          return outputs_mask; // XXXKF allow some delay after input going blank
      }
      void activate();
@@@ -723,7 -721,7 +721,7 @@@ public
  
          bool rms = *params[param_detection] == 0;
          bool average = *params[param_stereo_link] == 0;
-         float aweighting = *params[param_aweighting] > 0.5f;
+         bool aweighting = *params[param_aweighting] > 0.5f;
          float linThreshold = *params[param_threshold];
          ratio = *params[param_ratio];
          float attack = *params[param_attack];
@@@ -812,10 -810,10 +810,10 @@@
          return slope * output_gain(slope, false) * makeup;
      }
      
 -    inline float output_gain(float linSlope, bool doRms) {
 -         if(doRms ? linSlope > adjKneeStart : linSlope > linKneeStart) {
 +    inline float output_gain(float linSlope, bool rms) {
 +         if(linSlope > rms ? adjKneeStart : linKneeStart) {
              float slope = log(linSlope);
 -            if(doRms) slope *= 0.5f;
 +            if(rms) slope *= 0.5f;
  
              float gain = 0.f;
              float delta = 0.f;

-- 
calf audio plugins packaging