Move resample function to its own cc/hh file.

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

src/Makefile.am

@@ -6,7 +6,8 @@ COMMON_SRC = utils.hh utils.cc convcode.hh convcode.cc random.hh random.cc wavda
 	     sfoutputstream.cc sfoutputstream.hh rawinputstream.cc rawinputstream.hh rawoutputstream.cc rawoutputstream.hh \
 	     rawconverter.cc rawconverter.hh mp3inputstream.cc mp3inputstream.hh wmcommon.cc wmcommon.hh fft.cc fft.hh \
 	     limiter.cc limiter.hh shortcode.cc shortcode.hh mpegts.cc mpegts.hh hls.cc hls.hh audiobuffer.hh \
-	     wmget.cc wmadd.cc syncfinder.cc syncfinder.hh wmspeed.cc wmspeed.hh threadpool.cc threadpool.hh
+	     wmget.cc wmadd.cc syncfinder.cc syncfinder.hh wmspeed.cc wmspeed.hh threadpool.cc threadpool.hh \
+	     resample.cc resample.hh
 COMMON_LIBS = $(SNDFILE_LIBS) $(FFTW_LIBS) $(LIBGCRYPT_LIBS) $(LIBMPG123_LIBS) $(FFMPEG_LIBS)
 
 AM_CXXFLAGS = $(SNDFILE_CFLAGS) $(FFTW_CFLAGS) $(LIBGCRYPT_CFLAGS) $(LIBMPG123_CFLAGS) $(FFMPEG_CFLAGS)

src/resample.cc

+/*
+ * Copyright (C) 2018-2020 Stefan Westerfeld
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include "resample.hh"
+
+#include <assert.h>
+#include <math.h>
+
+#include <zita-resampler/resampler.h>
+#include <zita-resampler/vresampler.h>
+
+using std::vector;
+
+template<class R>
+static void
+process_resampler (R& resampler, const vector<float>& in, vector<float>& out)
+{
+  resampler.out_count = out.size() / resampler.nchan();
+  resampler.out_data = &out[0];
+
+  /* avoid timeshift: zita needs k/2 - 1 samples before the actual input */
+  resampler.inp_count = resampler.inpsize () / 2 - 1;
+  resampler.inp_data  = nullptr;
+  resampler.process();
+
+  resampler.inp_count = in.size() / resampler.nchan();
+  resampler.inp_data = (float *) &in[0];
+  resampler.process();
+
+  /* zita needs k/2 samples after the actual input */
+  resampler.inp_count = resampler.inpsize() / 2;
+  resampler.inp_data  = nullptr;
+  resampler.process();
+}
+
+WavData
+resample (const WavData& wav_data, int rate)
+{
+  /* in our application, resampling should only be called if it is necessary
+   * since using the resampler with input rate == output rate would be slow
+   */
+  assert (rate != wav_data.sample_rate());
+
+  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() / wav_data.n_channels() * ratio) * wav_data.n_channels());
+
+  /* zita-resampler provides two resampling algorithms
+   *
+   * a fast optimized version: Resampler
+   *   this is an optimized version, which works for many common cases,
+   *   like resampling between 22050, 32000, 44100, 48000, 96000 Hz
+   *
+   * a slower version: VResampler
+   *   this works for arbitary rates (like 33333 -> 44100 resampling)
+   *
+   * so we try using Resampler, and if that fails fall back to VResampler
+   */
+  Resampler resampler;
+  if (resampler.setup (wav_data.sample_rate(), rate, wav_data.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, wav_data.n_channels(), hlen) == 0)
+    {
+      process_resampler (vresampler, in, out);
+      return WavData (out, wav_data.n_channels(), rate, wav_data.bit_depth());
+    }
+  error ("audiowmark: resampling from rate %d to rate %d not supported.\n", wav_data.sample_rate(), rate);
+  exit (1);
+}
+
+

src/resample.hh

+/*
+ * Copyright (C) 2018-2020 Stefan Westerfeld
+ *
+ * This program is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#ifndef AUDIOWMARK_RESAMPLE_HH
+#define AUDIOWMARK_RESAMPLE_HH
+
+#include "wavdata.hh"
+
+WavData resample (const WavData& wav_data, int rate);
+
+#endif /* AUDIOWMARK_RESAMPLE_HH */

src/wmget.cc

@@ -18,15 +18,13 @@
 #include <string>
 #include <algorithm>
 
-#include <zita-resampler/resampler.h>
-#include <zita-resampler/vresampler.h>
-
 #include "wavdata.hh"
 #include "wmcommon.hh"
 #include "wmspeed.hh"
 #include "convcode.hh"
 #include "shortcode.hh"
 #include "syncfinder.hh"
+#include "resample.hh"
 #include "fft.hh"
 
 using std::string;
@@ -35,70 +33,6 @@ using std::min;
 using std::max;
 using std::complex;
 
-template<class R>
-static void
-process_resampler (R& resampler, const vector<float>& in, vector<float>& out)
-{
-  resampler.out_count = out.size() / resampler.nchan();
-  resampler.out_data = &out[0];
-
-  /* avoid timeshift: zita needs k/2 - 1 samples before the actual input */
-  resampler.inp_count = resampler.inpsize () / 2 - 1;
-  resampler.inp_data  = nullptr;
-  resampler.process();
-
-  resampler.inp_count = in.size() / resampler.nchan();
-  resampler.inp_data = (float *) &in[0];
-  resampler.process();
-
-  /* zita needs k/2 samples after the actual input */
-  resampler.inp_count = resampler.inpsize() / 2;
-  resampler.inp_data  = nullptr;
-  resampler.process();
-}
-
-static WavData
-resample (const WavData& wav_data, int rate)
-{
-  /* in our application, resampling should only be called if it is necessary
-   * since using the resampler with input rate == output rate would be slow
-   */
-  assert (rate != wav_data.sample_rate());
-
-  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() / wav_data.n_channels() * ratio) * wav_data.n_channels());
-
-  /* zita-resampler provides two resampling algorithms
-   *
-   * a fast optimized version: Resampler
-   *   this is an optimized version, which works for many common cases,
-   *   like resampling between 22050, 32000, 44100, 48000, 96000 Hz
-   *
-   * a slower version: VResampler
-   *   this works for arbitary rates (like 33333 -> 44100 resampling)
-   *
-   * so we try using Resampler, and if that fails fall back to VResampler
-   */
-  Resampler resampler;
-  if (resampler.setup (wav_data.sample_rate(), rate, wav_data.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, wav_data.n_channels(), hlen) == 0)
-    {
-      process_resampler (vresampler, in, out);
-      return WavData (out, wav_data.n_channels(), rate, wav_data.bit_depth());
-    }
-  error ("audiowmark: resampling from rate %d to rate %d not supported.\n", wav_data.sample_rate(), rate);
-  exit (1);
-}
-
 static vector<float>
 normalize_soft_bits (const vector<float>& soft_bits)
 {