Add a simple leveller/compressor.

diff --git a/Makefile b/Makefile
index ae22aa885a5ef4b75b065ec542dcc4f1d34309e9..a5c7b66a79a7be95bd37f8378226f1af6f5fd8cc 100644 (file)
--- a/Makefile
+++ b/Makefile
@@ -1,11 +1,11 @@
 CXXFLAGS=-O2 -ffast-math -g -Wall
 
 CXXFLAGS=-O2 -ffast-math -g -Wall
 

-all: synth decode sync
+all: synth decode sync level

 
 %.o: %.cpp
        $(CXX) -MMD -MP $(CPPFLAGS) $(CXXFLAGS) -o $@ -c $<
 
 
 %.o: %.cpp
        $(CXX) -MMD -MP $(CPPFLAGS) $(CXXFLAGS) -o $@ -c $<
 

-OBJS=decode.o synth.o synth_main.o interpolate.o sync.o
+OBJS=decode.o synth.o synth_main.o interpolate.o sync.o level.o

 
 DEPS=$(OBJS:.o=.d)
 -include $(DEPS)
 
 DEPS=$(OBJS:.o=.d)
 -include $(DEPS)


@@ -19,5 +19,8 @@ synth: synth.o synth_main.o
 sync: interpolate.o sync.o
        $(CXX) -o $@ $^ $(LDFLAGS)
 
 sync: interpolate.o sync.o
        $(CXX) -o $@ $^ $(LDFLAGS)
 

+level: level.o
+       $(CXX) -o $@ $^ $(LDFLAGS)
+

 clean:
 clean:

-       $(RM) synth decode sync $(OBJS) $(DEPS)
+       $(RM) synth decode sync level $(OBJS) $(DEPS)

diff --git a/level.cpp b/level.cpp
new file mode 100644 (file)
index 0000000..e14704a
--- /dev/null
+++ b/level.cpp
@@ -0,0 +1,125 @@
+// A small program to try to even out the audio levels within a file
+// (essentially a compressor with infinite lookahead).
+
+#include <stdio.h>
+#include <math.h>
+#include <vector>
+#include <algorithm>
+
+#define BUFSIZE 4096
+#define WAVE_FREQ 44100.0
+
+// The frequency to filter on, in Hertz. Larger values makes the
+// compressor react faster, but if it is too large, you'll
+// ruin the waveforms themselves.
+#define LPFILTER_FREQ 50.0
+
+// The minimum estimated sound level at any given point.
+// If you decrease this, you'll be able to amplify really silent signals
+// by more, but you'll also increase the level of silent (ie. noise-only) segments,
+// possibly caused misdetected pulses in these segments.
+#define MIN_LEVEL 0.05
+
+// A final scalar to get the audio within approximately the right range.
+// Increase to _lower_ overall volume.
+#define DAMPENING_FACTOR 5.0
+
+// 6dB/oct per round.
+#define FILTER_DEPTH 4
+
+struct stereo_sample {
+       short left, right;
+};
+
+inline short clip(int x)
+{
+       if (x < -32768) {
+               return x;
+       } else if (x > 32767) {
+               return 32767;
+       } else {
+               return short(x);
+       }
+}
+
+static float a1, a2, b0, b1, b2;
+static float d0, d1;
+
+static void filter_init(float cutoff_radians)
+{
+       float resonance = 1.0f / sqrt(2.0f);
+       float sn, cs;
+       sincosf(cutoff_radians, &sn, &cs);
+
+       float alpha = float(sn / (2 * resonance));
+
+       // coefficients for lowpass filter
+        float a0 = 1 + alpha;
+       b0 = (1 - cs) * 0.5f;
+       b1 = 1 - cs;
+       b2 = b0;
+        a1 = -2 * cs;
+        a2 = 1 - alpha;
+
+       float invA0 = 1.0f / a0;
+       b0 *= invA0;
+       b1 *= invA0;
+       b2 *= invA0;
+       a1 *= invA0;
+       a2 *= invA0;
+
+       // reset filter delays
+       d0 = d1 = 0.0f;
+}
+
+static float filter_update(float in)
+{
+       float out = b0*in + d0;
+       d0 = b1 * in - a1 * out + d1;
+       d1 = b2 * in - a2 * out;
+       return out;
+}
+
+int main(int argc, char **argv)
+{
+       std::vector<short> pcm;
+
+       while (!feof(stdin)) {
+               short buf[BUFSIZE];
+               ssize_t ret = fread(buf, sizeof(short), BUFSIZE, stdin);
+               if (ret >= 0) {
+                       pcm.insert(pcm.end(), buf, buf + ret);
+               }
+       }
+       
+       // filter forwards, then backwards (perfect phase filtering)
+       std::vector<float> filtered_samples, refiltered_samples;
+       filtered_samples.resize(pcm.size());
+       refiltered_samples.resize(pcm.size());
+
+       filter_init(M_PI * LPFILTER_FREQ / WAVE_FREQ);
+       for (unsigned i = 0; i < pcm.size(); ++i) {
+               filtered_samples[i] = filter_update(fabs(pcm[i]));
+       }
+       filter_init(M_PI * LPFILTER_FREQ / WAVE_FREQ);
+       for (unsigned i = pcm.size(); i --> 0; ) {
+               refiltered_samples[i] = filter_update(filtered_samples[i]);
+       }
+
+       for (int i = 1; i < FILTER_DEPTH; ++i) {
+               filter_init(M_PI * LPFILTER_FREQ / WAVE_FREQ);
+               for (unsigned i = 0; i < pcm.size(); ++i) {
+                       filtered_samples[i] = filter_update(refiltered_samples[i]);
+               }
+               filter_init(M_PI * LPFILTER_FREQ / WAVE_FREQ);
+               for (unsigned i = pcm.size(); i --> 0; ) {
+                       refiltered_samples[i] = filter_update(filtered_samples[i]);
+               }
+       }
+
+       for (unsigned i = 0; i < pcm.size(); ++i) {
+               float f = DAMPENING_FACTOR * std::max(refiltered_samples[i] * (1.0 / 32768.0), MIN_LEVEL);
+               short s = clip(lrintf(pcm[i] / f));
+               fwrite(&s, sizeof(s), 1, stdout);
+       }
+}