Bitexactness test for the highpass filter

webrtc/modules/audio_processing/test/audio_buffer_tools.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.
+ */
+
+#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
+
+namespace webrtc {
+
+void SetupFrame(StreamConfig stream_config,
+                std::vector<float*>* frame,
+                std::vector<float>* frame_samples) {
+  frame_samples->resize(stream_config.num_channels() *
+                        stream_config.num_frames());
+  frame->resize(stream_config.num_frames());

[the sun, 2016/02/22 19:50:58]

should be sized to stream_config.num_channels()

[peah-webrtc, 2016/02/22 23:11:36]

Great find!

Done.

+  for (int ch = 0; ch < stream_config.num_channels(); ++ch) {
+    (*frame)[ch] = &(*frame_samples)[ch * stream_config.num_frames()];
+  }
+}
+
+void CopyVectorToAudioBuffer(StreamConfig stream_config,
+                             const std::vector<float>& source,
+                             AudioBuffer* destination) {
+  std::vector<float*> input;
+  std::vector<float> input_samples;
+
+  SetupFrame(stream_config, &input, &input_samples);
+
+  RTC_DCHECK_EQ(input_samples.size(), source.size());
+  input_samples = source;
+
+  destination->CopyFrom(&input[0], stream_config);
+}
+
+std::vector<float> ExtractVectorFromAudioBuffer(StreamConfig stream_config,
+                                                AudioBuffer* source) {
+  std::vector<float*> output;
+  std::vector<float> output_samples;
+
+  SetupFrame(stream_config, &output, &output_samples);
+
+  source->CopyTo(stream_config, &output[0]);
+
+  return output_samples;
+}
+
+}  // namespace webrtc