Pull out the PostFilter to its own NonlinearBeamformer API

webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 40 matching lines @@
   RTC_CHECK_EQ(array_geometry.size(), num_mics);
 
   NonlinearBeamformer bf(array_geometry);
   bf.Initialize(kChunkSizeMs, in_file.sample_rate());
 
   printf("Input file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
          FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
   printf("Output file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
          FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
 
-  ChannelBuffer<float> in_buf(
+  ChannelBuffer<float> buf(
       rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
       in_file.num_channels());
-  ChannelBuffer<float> out_buf(
-      rtc::CheckedDivExact(out_file.sample_rate(), kChunksPerSecond),
-      out_file.num_channels());
 
-  std::vector<float> interleaved(in_buf.size());
+  std::vector<float> interleaved(buf.size());
   while (in_file.ReadSamples(interleaved.size(),
                              &interleaved[0]) == interleaved.size()) {
+    buf.set_num_channels(in_file.num_channels());
     FloatS16ToFloat(&interleaved[0], interleaved.size(), &interleaved[0]);
-    Deinterleave(&interleaved[0], in_buf.num_frames(),
-                 in_buf.num_channels(), in_buf.channels());
+    Deinterleave(&interleaved[0], buf.num_frames(),
+                 buf.num_channels(), buf.channels());
 
-    bf.ProcessChunk(in_buf, &out_buf);
+    bf.AnalyzeChunk(buf);
+    bf.PostFilter(&buf);

[peah-webrtc, 2016/06/08 12:04:55]

Does this really work? The AnalyzeChunk does not downmix the signal, right? But
afaics the PostFilter assumes a mono signal and CHECKS otherwise. Or am I
missing something?

[aluebs-webrtc, 2016/06/09 02:11:46]

It did, because it doesn't use the number of channels of the ChannelBuffer at
all. Instead it uses the number of channels set in the constructor. But added a
DCHECK to make it more restrictive.

 
-    Interleave(out_buf.channels(), out_buf.num_frames(),
-               out_buf.num_channels(), &interleaved[0]);
-    FloatToFloatS16(&interleaved[0], interleaved.size(), &interleaved[0]);
-    out_file.WriteSamples(&interleaved[0], interleaved.size());
+    Interleave(buf.channels(), buf.num_frames(),
+               buf.num_channels(), &interleaved[0]);
+    FloatToFloatS16(buf.channels()[0], buf.num_frames(), &interleaved[0]);
+    out_file.WriteSamples(&interleaved[0], buf.num_frames());
   }
 
   return 0;
 }
 
 }  // namespace webrtc
 
 int main(int argc, char* argv[]) {
   return webrtc::main(argc, argv);
 }