Add aecdump support to audioproc_f.

webrtc/modules/audio_processing/test/test_utils.cc

Index: webrtc/modules/audio_processing/test/test_utils.cc
diff --git a/webrtc/modules/audio_processing/test/test_utils.cc b/webrtc/modules/audio_processing/test/test_utils.cc
index 1b9ac3ce4cb8b40c3bf852c9579386618c058005..47bd3144ccbbe87289982254ccc45069d7efea9e 100644
--- a/webrtc/modules/audio_processing/test/test_utils.cc
+++ b/webrtc/modules/audio_processing/test/test_utils.cc
@@ -31,6 +31,35 @@ void RawFile::WriteSamples(const float* samples, size_t num_samples) {
   fwrite(samples, sizeof(*samples), num_samples, file_handle_);
 }
 
+ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file)
+    : file_(file.Pass()) {}
+
+bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
+  interleaved_.resize(buffer->size());
+  if (file_->ReadSamples(interleaved_.size(), &interleaved_[0]) !=
+      interleaved_.size()) {
+    return false;
+  }
+
+  FloatS16ToFloat(&interleaved_[0], interleaved_.size(), &interleaved_[0]);
+  Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
+               buffer->channels());
+  return true;
+}
+
+ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
+    : file_(file.Pass()) {}
+
+void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
+  RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
+  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,
@@ -92,28 +121,32 @@ AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels) {
     case 2:
       return AudioProcessing::kStereo;
     default:
-      assert(false);
+      RTC_CHECK(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) << "mic_positions is not large enough.";
 
   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