Cleaning up audio_decoder_test.cc and adding ResampleInputAudioFile

This CL contains some cleaning up and refactoring of
audio_decoder_test.cc. A new class ResampleInputAudioFile is created
and used in the tests.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7528 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc

@@ -0,0 +1,42 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+namespace test {
+
+bool ResampleInputAudioFile::Read(size_t samples,
+                                  int output_rate_hz,
+                                  int16_t* destination) {
+  const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
+  CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
+      << "Frame size and sample rates don't add up to an integer.";
+  scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
+  if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
+    return false;
+  resampler_.ResetIfNeeded(
+      file_rate_hz_, output_rate_hz, kResamplerSynchronous);
+  int output_length = 0;
+  CHECK_EQ(resampler_.Push(temp_destination.get(),
+                           static_cast<int>(samples_to_read),
+                           destination,
+                           static_cast<int>(samples),
+                           output_length),
+           0);
+  CHECK_EQ(static_cast<int>(samples), output_length);
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h

@@ -0,0 +1,40 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/common_audio/resampler/include/resampler.h"
+#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for handling a looping input audio file with resampling.
+class ResampleInputAudioFile : public InputAudioFile {
+ public:
+  ResampleInputAudioFile(const std::string file_name, int file_rate_hz)
+      : InputAudioFile(file_name), file_rate_hz_(file_rate_hz) {}
+
+  bool Read(size_t samples, int output_rate_hz, int16_t* destination);
+
+ private:
+  const int file_rate_hz_;
+  Resampler resampler_;
+  DISALLOW_COPY_AND_ASSIGN(ResampleInputAudioFile);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_