Optimise audio filters; tweak order of the LPFs in the upmixers.

[dcpomatic.git] / src / lib / audio_filter.cc

diff --git a/src/lib/audio_filter.cc b/src/lib/audio_filter.cc
index 10f3849dffa3aa271b9521b14c4af24d9f1533e0..2c92236b6f55fe1b2a0c4ffd3bab3afaad30f5bb 100644 (file)
--- a/src/lib/audio_filter.cc
+++ b/src/lib/audio_filter.cc
@@ -21,14 +21,14 @@
 #include "audio_buffers.h"
 #include <cmath>
 
-using std::vector;
 using std::min;
 using boost::shared_ptr;
 
-vector<float>
+/** @return array of floats which the caller must destroy with delete[] */
+float *
 AudioFilter::sinc_blackman (float cutoff, bool invert) const
 {
-       vector<float> ir (_M + 1);
+       float* ir = new float[_M + 1];
 
        /* Impulse response */
 
@@ -66,6 +66,11 @@ AudioFilter::sinc_blackman (float cutoff, bool invert) const
        return ir;
 }
 
+AudioFilter::~AudioFilter ()
+{
+       delete[] _ir;
+}
+
 shared_ptr<AudioBuffers>
 AudioFilter::run (shared_ptr<const AudioBuffers> in)
 {
@@ -76,18 +81,24 @@ AudioFilter::run (shared_ptr<const AudioBuffers> in)
                _tail->make_silent ();
        }
 
-       for (int i = 0; i < in->channels(); ++i) {
-               for (int j = 0; j < in->frames(); ++j) {
+       int const channels = in->channels ();
+       int const frames = in->frames ();
+
+       for (int i = 0; i < channels; ++i) {
+               float* tail_p = _tail->data (i);
+               float* in_p = in->data (i);
+               float* out_p = out->data (i);
+               for (int j = 0; j < frames; ++j) {
                        float s = 0;
                        for (int k = 0; k <= _M; ++k) {
                                if ((j - k) < 0) {
-                                       s += _tail->data(i)[j - k + _M + 1] * _ir[k];
+                                       s += tail_p[j - k + _M + 1] * _ir[k];
                                } else {
-                                       s += in->data(i)[j - k] * _ir[k];
+                                       s += in_p[j - k] * _ir[k];
                                }
                        }
 
-                       out->data(i)[j] = s;
+                       out_p[j] = s;
                }
        }
 
@@ -122,14 +133,18 @@ HighPassAudioFilter::HighPassAudioFilter (float transition_bandwidth, float cuto
 BandPassAudioFilter::BandPassAudioFilter (float transition_bandwidth, float lower, float higher)
        : AudioFilter (transition_bandwidth)
 {
-       vector<float> lpf = sinc_blackman (lower, false);
-       vector<float> hpf = sinc_blackman (higher, true);
+       float* lpf = sinc_blackman (lower, false);
+       float* hpf = sinc_blackman (higher, true);
 
-       _ir.resize (_M + 1);
+       delete[] _ir;
+       _ir = new float[_M + 1];
        for (int i = 0; i <= _M; ++i) {
                _ir[i] = lpf[i] + hpf[i];
        }
 
+       delete[] lpf;
+       delete[] hpf;
+
        /* We now have a band-stop, so invert for band-pass */
        for (int i = 0; i <= _M; ++i) {
                _ir[i] = -_ir[i];