Conversational Speech tool, MultiEndCall class and unit tests via mocking

webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h

Index: webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h b/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
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index 0000000000000000000000000000000000000000..519c718bb94d2147873d7275e40096ee16a6f89d
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+/*
+ *  Copyright (c) 2017 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_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_
+
+#include <cstddef>
+#include <string>
+
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class MockWavReader : public WavReaderInterface {
+ public:
+  MockWavReader(
+      int sample_rate, size_t num_channels, size_t num_samples)
+          : sample_rate_(sample_rate), num_channels_(num_channels),
+            num_samples_(num_samples) {}
+  ~MockWavReader() = default;
+
+  // TOOD(alessiob): use ON_CALL to return random samples.
+  MOCK_METHOD2(ReadFloatSamples, size_t(size_t, float*));
+  MOCK_METHOD2(ReadInt16Samples, size_t(size_t, int16_t*));
+
+  // TOOD(alessiob): use ON_CALL to return properties.
+  MOCK_CONST_METHOD0(sample_rate, int());
+  MOCK_CONST_METHOD0(num_channels, size_t());
+  MOCK_CONST_METHOD0(num_samples, size_t());
+
+ private:
+  int sample_rate_;

[hlundin-webrtc, 2017/03/27 09:46:35]

Make these consts if you can.

[AleBzk, 2017/03/27 11:17:06]

Done.

+  size_t num_channels_;
+  size_t num_samples_;
+};
+
+}  // namespace conversational_speech
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_