If Resampler doesn't support desired ratio, fallback to VResampler.

Signed-off-by: Stefan Westerfeld <stefan@space.twc.de>

src/audiowmark.cc

@@ -12,6 +12,7 @@
 #include "random.hh"
 
 #include <zita-resampler/resampler.h>
+#include <zita-resampler/vresampler.h>
 
 #include <assert.h>
 
@@ -501,23 +502,11 @@ mark_pad (const WavData& wav_data, size_t frame, const vector<vector<complex<flo
     }
 }
 
-WavData
-resample (const WavData& wav_data, int rate)
+template<class R>
+static void
+process_resampler (R& resampler, const vector<float>& in, vector<float>& out)
 {
-  if (rate == wav_data.sample_rate())
-    return wav_data;
-
-  Resampler resampler;
-
-  const int n_channels = wav_data.n_channels();
-  const int hlen = 16;
-  const double ratio = double (rate) / wav_data.sample_rate();
-
-  const vector<float>& in = wav_data.samples();
-  vector<float> out (lrint (in.size() / n_channels * ratio) * n_channels);
-
-  resampler.setup (wav_data.sample_rate(), rate, n_channels, hlen);
-  resampler.out_count = out.size() / n_channels;
+  resampler.out_count = out.size() / resampler.nchan();
   resampler.out_data = &out[0];
 
   /* avoid timeshift: zita needs k/2 - 1 samples before the actual input */
@@ -525,7 +514,7 @@ resample (const WavData& wav_data, int rate)
   resampler.inp_data  = nullptr;
   resampler.process();
 
-  resampler.inp_count = in.size() / n_channels;
+  resampler.inp_count = in.size() / resampler.nchan();
   resampler.inp_data = (float *) &in[0];
   resampler.process();
 
@@ -533,8 +522,36 @@ resample (const WavData& wav_data, int rate)
   resampler.inp_count = resampler.inpsize() / 2;
   resampler.inp_data  = nullptr;
   resampler.process();
+}
 
-  return WavData (out, wav_data.n_channels(), rate, wav_data.bit_depth());
+WavData
+resample (const WavData& wav_data, int rate)
+{
+  if (rate == wav_data.sample_rate())
+    return wav_data;
+
+  const int n_channels = wav_data.n_channels();
+  const int hlen = 16;
+  const double ratio = double (rate) / wav_data.sample_rate();
+
+  const vector<float>& in = wav_data.samples();
+  vector<float> out (lrint (in.size() / n_channels * ratio) * n_channels);
+
+  Resampler resampler;
+  if (resampler.setup (wav_data.sample_rate(), rate, n_channels, hlen) == 0)
+    {
+      process_resampler (resampler, in, out);
+      return WavData (out, wav_data.n_channels(), rate, wav_data.bit_depth());
+    }
+
+  VResampler vresampler;
+  if (vresampler.setup (ratio, n_channels, hlen) == 0)
+    {
+      process_resampler (vresampler, in, out);
+      return WavData (out, wav_data.n_channels(), rate, wav_data.bit_depth());
+    }
+  fprintf (stderr, "audiowmark: resampling from rate %d to rate %d not supported.\n", wav_data.sample_rate(), rate);
+  exit (1);
 }
 
 int