Move shared class AudioBuffer to its own header.

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

src/Makefile.am

@@ -5,7 +5,8 @@ COMMON_SRC = utils.hh utils.cc convcode.hh convcode.cc random.hh random.cc wavda
 	     audiostream.cc audiostream.hh sfinputstream.cc sfinputstream.hh stdoutwavoutputstream.cc stdoutwavoutputstream.hh \
 	     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 hlsoutputstream.cc hlsoutputstream.hh
+	     limiter.cc limiter.hh shortcode.cc shortcode.hh mpegts.cc mpegts.hh hlsoutputstream.cc hlsoutputstream.hh \
+	     audiobuffer.hh
 COMMON_LIBS = $(SNDFILE_LIBS) $(FFTW_LIBS) $(LIBGCRYPT_LIBS) $(LIBMPG123_LIBS) $(FFMPEG_LIBS)
 
 audiowmark_SOURCES = audiowmark.cc wmget.cc wmadd.cc $(COMMON_SRC)

src/audiobuffer.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_AUDIO_BUFFER_HH
+#define AUDIOWMARK_AUDIO_BUFFER_HH
+
+#include <assert.h>
+
+class AudioBuffer
+{
+  const int           n_channels = 0;
+  std::vector<float>  buffer;
+
+public:
+  AudioBuffer (int n_channels) :
+    n_channels (n_channels)
+  {
+  }
+  void
+  write_frames (const std::vector<float>& samples)
+  {
+    buffer.insert (buffer.end(), samples.begin(), samples.end());
+  }
+  std::vector<float>
+  read_frames (size_t frames)
+  {
+    assert (frames * n_channels <= buffer.size());
+    const auto begin = buffer.begin();
+    const auto end   = begin + frames * n_channels;
+    std::vector<float> result (begin, end);
+    buffer.erase (begin, end);
+    return result;
+  }
+  size_t
+  can_read_frames() const
+  {
+    return buffer.size() / n_channels;
+  }
+};
+
+#endif /* AUDIOWMARK_AUDIO_BUFFER_HH */

src/hlsoutputstream.hh

@@ -19,6 +19,7 @@
 #define AUDIOWMARK_HLS_OUTPUT_STREAM_HH
 
 #include "audiostream.hh"
+#include "audiobuffer.hh"
 
 #include <assert.h>
 
@@ -32,39 +33,6 @@ extern "C" {
 #define av_err2str(errnum) av_make_error_string((char*)__builtin_alloca(AV_ERROR_MAX_STRING_SIZE), AV_ERROR_MAX_STRING_SIZE, errnum)
 }
 
-/* FIXME: fix duplication with wmadd.cc */
-class AudioBuffer
-{
-  const int           n_channels = 0;
-  std::vector<float>  buffer;
-
-public:
-  AudioBuffer (int n_channels) :
-    n_channels (n_channels)
-  {
-  }
-  void
-  write_frames (const std::vector<float>& samples)
-  {
-    buffer.insert (buffer.end(), samples.begin(), samples.end());
-  }
-  std::vector<float>
-  read_frames (size_t frames)
-  {
-    assert (frames * n_channels <= buffer.size());
-    const auto begin = buffer.begin();
-    const auto end   = begin + frames * n_channels;
-    std::vector<float> result (begin, end);
-    buffer.erase (begin, end);
-    return result;
-  }
-  size_t
-  can_read_frames() const
-  {
-    return buffer.size() / n_channels;
-  }
-};
-
 class HLSOutputStream : public AudioOutputStream {
   AVStream         *m_st = nullptr;
   AVCodecContext   *m_enc = nullptr;

src/wmadd.cc

@@ -31,6 +31,7 @@
 #include "rawoutputstream.hh"
 #include "stdoutwavoutputstream.hh"
 #include "shortcode.hh"
+#include "audiobuffer.hh"
 
 using std::string;
 using std::vector;
@@ -344,38 +345,6 @@ public:
   }
 };
 
-class AudioBuffer
-{
-  const int     n_channels = 0;
-  vector<float> buffer;
-
-public:
-  AudioBuffer (int n_channels) :
-    n_channels (n_channels)
-  {
-  }
-  void
-  write_frames (const vector<float>& samples)
-  {
-    buffer.insert (buffer.end(), samples.begin(), samples.end());
-  }
-  vector<float>
-  read_frames (size_t frames)
-  {
-    assert (frames * n_channels <= buffer.size());
-    const auto begin = buffer.begin();
-    const auto end   = begin + frames * n_channels;
-    vector<float> result (begin, end);
-    buffer.erase (begin, end);
-    return result;
-  }
-  size_t
-  can_read_frames() const
-  {
-    return buffer.size() / n_channels;
-  }
-};
-
 class ResamplerImpl
 {
 public: