Tweak DSP optimization.

mixer/src/AUX_Module.C

@@ -108,7 +108,15 @@ AUX_Module::handle_sample_rate_change ( nframes_t n )
 void
 AUX_Module::process ( nframes_t nframes )
 {
-    if ( !bypass() )
+    if ( unlikely( bypass() ) )
+    {
+        for ( unsigned int i = 0; i < audio_input.size(); ++i )
+        {
+            if ( audio_input[i].connected() )
+                buffer_fill_with_silence( (sample_t*)aux_audio_output[i].jack_port()->buffer(nframes), nframes );
+        }
+    }
+    else
     {
         float gt = DB_CO( control_input[0].control_value() );
  
@@ -116,9 +124,8 @@ AUX_Module::process ( nframes_t nframes )
     
         bool use_gainbuf = smoothing.apply( gainbuf, nframes, gt );
 
-        if ( use_gainbuf )
+        if ( unlikely( use_gainbuf ) )
         {
-        
             for ( unsigned int i = 0; i < audio_input.size(); ++i )
             {
                 if ( audio_input[i].connected() )
@@ -135,14 +142,6 @@ AUX_Module::process ( nframes_t nframes )
             }
         }
     }
-    else
-    {
-        for ( unsigned int i = 0; i < audio_input.size(); ++i )
-        {
-            if ( audio_input[i].connected() )
-                buffer_fill_with_silence( (sample_t*)aux_audio_output[i].jack_port()->buffer(nframes), nframes );
-        }
-    }
 }
 
 void

mixer/src/Gain_Module.C

@@ -105,13 +105,20 @@ Gain_Module::handle_sample_rate_change ( nframes_t n )
 void
 Gain_Module::process ( nframes_t nframes )
 {
+
+    if ( unlikely( bypass() ) )
+    {
+        /* nothing to do */
+    }
+    else
+    {
         const float gt = DB_CO( control_input[1].control_value() ? -90.f : control_input[0].control_value() );
 
         sample_t gainbuf[nframes];
     
         bool use_gainbuf = smoothing.apply( gainbuf, nframes, gt );
         
-    if ( use_gainbuf )
+        if ( unlikely( use_gainbuf ) )
         {
             for ( int i = audio_input.size(); i--; )
             {
@@ -131,4 +138,5 @@ Gain_Module::process ( nframes_t nframes )
                     buffer_apply_gain( (sample_t*)audio_input[i].buffer(), nframes, gt );
                 }
             }
+    }
 }

mixer/src/Meter_Module.C

@@ -174,8 +174,6 @@ Meter_Module::process ( nframes_t nframes )
 {
     for ( unsigned int i = 0; i < audio_input.size(); ++i )
     {
-        if ( audio_input[i].connected() )
-        {
 //            float dB = 20 * log10( get_peak_sample( (float*)audio_input[i].buffer(), nframes ) / 2.0f );
         float dB = 20 * log10( buffer_get_peak( (sample_t*) audio_input[i].buffer(), nframes ) );
 
@@ -183,5 +181,4 @@ Meter_Module::process ( nframes_t nframes )
         if (dB > control_value[i])
             control_value[i] = dB;
     }
-    }
 }

mixer/src/Mono_Pan_Module.C

@@ -80,12 +80,7 @@ Mono_Pan_Module::configure_inputs ( int )
 void
 Mono_Pan_Module::process ( nframes_t nframes )
 {
-
+    if ( unlikely( bypass() ) )
-    if ( audio_input[0].connected() &&
-         audio_output[0].connected() &&
-         audio_output[1].connected() )
-    {
-        if ( bypass() )
     {
         buffer_copy( (sample_t*)audio_output[1].buffer(), (sample_t*)audio_input[0].buffer(), nframes );
     }
@@ -93,12 +88,11 @@ Mono_Pan_Module::process ( nframes_t nframes )
     {
         const float gt = (control_input[0].control_value() + 1.0f) * 0.5f;
 
-            if ( ! smoothing.target_reached( gt ) )
-            {            
         sample_t gainbuf[nframes];            
+        bool use_gainbuf = smoothing.apply( gainbuf, nframes, gt );
         
-                smoothing.apply( gainbuf, nframes, gt );
+        if ( unlikely( use_gainbuf ) )
-                
+        {            
             /* right channel */
                 
             buffer_copy_and_apply_gain_buffer( (sample_t*)audio_output[1].buffer(),
@@ -124,5 +118,4 @@ Mono_Pan_Module::process ( nframes_t nframes )
             buffer_apply_gain( (sample_t*)audio_output[0].buffer(), nframes, 1.0f - gt);
         }
     }
-    }
 }

mixer/src/Plugin_Module.C

@@ -795,12 +795,7 @@ Plugin_Module::process ( nframes_t nframes )
 {
     handle_port_connection_change();
 
-    if ( !bypass() )
+    if ( unlikely( bypass() ) )
-    {
-        for ( unsigned int i = 0; i < _idata->handle.size(); ++i )
-            _idata->descriptor->run( _idata->handle[i], nframes );
-    }
-    else
     {
         /* If this is a mono to stereo plugin, then duplicate the input channel... */
         /* There's not much we can do to automatically support other configurations. */
@@ -808,9 +803,16 @@ Plugin_Module::process ( nframes_t nframes )
         {
             buffer_copy( (sample_t*)audio_output[1].buffer(), (sample_t*)audio_input[0].buffer(), nframes );
         }
+
+        _latency = 0;
     }
+    else
+    {
+        for ( unsigned int i = 0; i < _idata->handle.size(); ++i )
+            _idata->descriptor->run( _idata->handle[i], nframes );
 
         _latency = get_plugin_latency();
+    }
 }
 
 

mixer/src/Spatializer_Module.C

@@ -595,8 +595,6 @@ Spatializer_Module::draw ( void )
 void
 Spatializer_Module::process ( nframes_t nframes )
 {
-    if ( !bypass() )
-    {
     float azimuth = control_input[0].control_value();
     float elevation = control_input[1].control_value();
     float radius = control_input[2].control_value();
@@ -651,7 +649,7 @@ Spatializer_Module::process ( nframes_t nframes )
         use_gainbuf = late_gain_smoothing.apply( gainbuf, nframes, late_gain );
             
         /* gain effects */
-            if ( use_gainbuf )
+        if ( unlikely( use_gainbuf ) )
             buffer_apply_gain_buffer( (sample_t*)aux_audio_output[0].jack_port()->buffer(nframes), gainbuf, nframes );
         else
             buffer_apply_gain( (sample_t*)aux_audio_output[0].jack_port()->buffer(nframes), nframes, late_gain );
@@ -695,7 +693,7 @@ Spatializer_Module::process ( nframes_t nframes )
         for ( int i = 1; i < 5; i++ )
         {
             /* gain effects */
-                if ( use_gainbuf )
+            if ( unlikely( use_gainbuf ) )
                 buffer_apply_gain_buffer( (sample_t*)aux_audio_output[i].jack_port()->buffer(nframes), gainbuf, nframes );
             else
                 buffer_apply_gain( (sample_t*)aux_audio_output[i].jack_port()->buffer(nframes), nframes, early_gain );
@@ -714,7 +712,7 @@ Spatializer_Module::process ( nframes_t nframes )
     for ( unsigned int i = 0; i < audio_input.size(); i++ )
     {
         /* gain effects */
-            if ( use_gainbuf )
+        if ( unlikely( use_gainbuf ) )
             buffer_apply_gain_buffer( (sample_t*)audio_input[i].buffer(), gainbuf, nframes );
         else
             buffer_apply_gain( (sample_t*)audio_input[i].buffer(), nframes, gain );
@@ -723,9 +721,9 @@ Spatializer_Module::process ( nframes_t nframes )
         _lowpass[i]->run_lowpass( (sample_t*)audio_input[i].buffer(), cutoff_frequency, nframes );
 
         /* delay effects */
-            if ( speed_of_sound )
+        if ( likely( speed_of_sound ) )
         {
-                if ( use_delaybuf )
+            if ( unlikely( use_delaybuf ) )
                 _delay[i]->run( (sample_t*)audio_input[i].buffer(), delaybuf, 0, nframes );
             else
                 _delay[i]->run( (sample_t*)audio_input[i].buffer(), 0, delay_seconds, nframes );
@@ -757,7 +755,6 @@ Spatializer_Module::process ( nframes_t nframes )
                              width,
                              nframes );
     }
-    }
 }
 
 void

nonlib/dsp.C

@@ -46,17 +46,21 @@ buffer_apply_gain ( sample_t * __restrict__ buf, nframes_t nframes, float g )
 {
     sample_t * buf_ = (sample_t*) assume_aligned(buf);
 	
-    if ( g != 1.0f )
+    if ( g == 1.0f )
-        while ( nframes-- )
+        return;
-	  *(buf_++) *= g;
+    
+    for ( nframes_t i = 0; i < nframes; i++ )
+        buf_[i] *= g;
 }
 
 void
 buffer_apply_gain_unaligned ( sample_t * __restrict__ buf, nframes_t nframes, float g )
 {
-    if ( g != 1.0f )
+    if ( g == 1.0f )
-        while ( nframes-- )
+        return;
-	  *(buf++) *= g;
+    
+    for ( nframes_t i = 0; i < nframes; i++ )
+        buf[i] *= g;
 }
 
 void
@@ -65,8 +69,8 @@ buffer_apply_gain_buffer ( sample_t * __restrict__ buf, const sample_t * __restr
     sample_t * buf_ = (sample_t*) assume_aligned(buf);
     const sample_t * gainbuf_ = (const sample_t*) assume_aligned(gainbuf);
 
-    while ( nframes-- )
+    for ( nframes_t i = 0; i < nframes; i++ )
-        *(buf_++) *= *(gainbuf_++);
+        buf_[i] *= gainbuf_[i];
 }
 
 void
@@ -76,8 +80,8 @@ buffer_copy_and_apply_gain_buffer ( sample_t * __restrict__ dst, const sample_t
     const sample_t * src_ = (const sample_t*) assume_aligned(src);
     const sample_t * gainbuf_ = (const sample_t*) assume_aligned(gainbuf);
     
-	while ( nframes-- )
+    for ( nframes_t i = 0; i < nframes; i++ )
-        *(dst_++) = *(src_++) * *(gainbuf_++);
+        dst_[i] = src_[i] * gainbuf_[i];
 }
 
 void
@@ -86,8 +90,8 @@ buffer_mix ( sample_t * __restrict__ dst, const sample_t * __restrict__ src, nfr
     sample_t * dst_ = (sample_t*) assume_aligned(dst);
     const sample_t * src_ = (const sample_t*) assume_aligned(src);
 
-    while ( nframes-- )
+    for ( nframes_t i = 0; i < nframes; i++ )
-        *(dst_++) += *(src_++);
+        dst_[i] += src_[i];
 }
 
 void
@@ -96,8 +100,8 @@ buffer_mix_with_gain ( sample_t * __restrict__ dst, const sample_t * __restrict_
     sample_t * dst_ = (sample_t*) assume_aligned(dst);
     const sample_t * src_ = (const sample_t*) assume_aligned(src);
 
-    while ( nframes-- )
+    for ( nframes_t i = 0; i < nframes; i++ )
-        *(dst_++) += *(src_++) * g;
+        dst_[i] += src_[i] * g;
 }
 
 void
@@ -181,7 +185,9 @@ buffer_is_digital_black ( sample_t *buf, nframes_t nframes )
 {
     while ( nframes-- )
     {
-        if ( 0 != buf[nframes] )
+        if (! *(buf++) )
+            continue;
+
         return false;
     }
 
@@ -193,15 +199,19 @@ buffer_get_peak ( const sample_t * __restrict__ buf, nframes_t nframes )
 {
     const sample_t * buf_ = (const sample_t*) assume_aligned(buf);
 
-    float p = 0.0f;
+    float pmax = 0.0f;
+    float pmin = 0.0f;
 
-    while ( nframes-- )
+    for ( nframes_t i = 0; i < nframes; i++ )
     {
-	const float s = fabs(*(buf_++));
+        pmax = buf_[i] > pmax ? buf_[i] : pmax;
-        p = s > p ? s : p;
+        pmin = buf_[i] < pmin ? buf_[i] : pmin;
     }
 
-    return p;
+    pmax = fabsf(pmax);
+    pmin = fabsf(pmin);
+    
+    return pmax > pmin ? pmax : pmin;
 }
 
 void

nonlib/dsp.h

@@ -79,3 +79,10 @@ static inline float interpolate_cubic ( const float fr, const float inm1, const
 
 #define DEG2RAD       0.01745329251f
 #define ONEOVERSQRT2  0.70710678118f
+
+#ifndef likely
+#define likely(x) __builtin_expect(x,1)
+#endif
+#ifndef unlikely
+#define unlikely(x) __builtin_expect(x,0)
+#endif