Add aecdump support to audioproc_f.

webrtc/modules/audio_processing/test/test_utils.cc

 /*
  *  Copyright (c) 2015 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 13 matching lines @@
 #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
 #error "Need to convert samples to little-endian when writing to PCM file"
 #endif
   fwrite(samples, sizeof(*samples), num_samples, file_handle_);
 }
 
 void RawFile::WriteSamples(const float* samples, size_t num_samples) {
   fwrite(samples, sizeof(*samples), num_samples, file_handle_);
 }
 
+ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
+    : file_(file.Pass()) {}
+
+void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {

[aluebs-webrtc, 2015/10/24 00:53:35]

Check the sample rate and number of channels corresponds to the WavWriter?

[Andrew MacDonald, 2015/10/29 00:44:50]

ChannelBuffer doesn't know about sample rate, but I can check the number of
channels.

[aluebs-webrtc, 2015/10/29 01:03:20]

Great point about sample rate.

+  interleaved_.resize(buffer.size());
+  Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
+             &interleaved_[0]);
+  FloatToFloatS16(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  file_->WriteSamples(&interleaved_[0], interleaved_.size());
+}
+
 void WriteIntData(const int16_t* data,
                   size_t length,
                   WavWriter* wav_file,
                   RawFile* raw_file) {
   if (wav_file) {
     wav_file->WriteSamples(data, length);
   }
   if (raw_file) {
     raw_file->WriteSamples(data, length);
   }
@@ skipping 46 matching lines @@
     case 1:
       return AudioProcessing::kMono;
     case 2:
       return AudioProcessing::kStereo;
     default:
       assert(false);
       return AudioProcessing::kMono;
   }
 }
 
-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
-                                      size_t num_mics) {
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions) {
   const std::vector<float> values = ParseList<float>(mic_positions);
-  RTC_CHECK_EQ(values.size(), 3 * num_mics)
-      << "Could not parse mic_positions or incorrect number of points.";
+  const size_t num_mics =
+      rtc::CheckedDivExact(values.size(), static_cast<size_t>(3));
+  RTC_CHECK_GT(num_mics, 0u);

[aluebs-webrtc, 2015/10/24 00:53:34]

Maybe add a fancy error message here too?

[Andrew MacDonald, 2015/10/29 00:44:50]

Done.

 
   std::vector<Point> result;
   result.reserve(num_mics);
   for (size_t i = 0; i < values.size(); i += 3) {
-    double x = values[i + 0];
-    double y = values[i + 1];
-    double z = values[i + 2];
-    result.push_back(Point(x, y, z));
+    result.push_back(Point(values[i + 0], values[i + 1], values[i + 2]));
   }
 
   return result;
 }
 
+std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
+                                      size_t num_mics) {
+  std::vector<Point> result = ParseArrayGeometry(mic_positions);
+  RTC_CHECK_EQ(result.size(), num_mics)
+      << "Could not parse mic_positions or incorrect number of points.";
+  return result;
+}
 
 }  // namespace webrtc