Created a mock AudioTransport.

webrtc/modules/audio_device/include/mock_audio_transport.h

Index: webrtc/modules/audio_device/include/mock_audio_transport.h
diff --git a/webrtc/modules/audio_device/include/mock_audio_transport.h b/webrtc/modules/audio_device/include/mock_audio_transport.h
new file mode 100644
index 0000000000000000000000000000000000000000..a05e7ea839298ea6ab1acd3e14c8b173ffada042
--- /dev/null
+++ b/webrtc/modules/audio_device/include/mock_audio_transport.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 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_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_
+
+#include "webrtc/modules/audio_device/include/audio_device_defines.h"
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockAudioTransport : public AudioTransport {
+ public:
+  MockAudioTransport() {}
+  ~MockAudioTransport() {}
+
+  MOCK_METHOD10(RecordedDataIsAvailable,
+                int32_t(const void* audioSamples,
+                        const size_t nSamples,
+                        const size_t nBytesPerSample,
+                        const size_t nChannels,
+                        const uint32_t samplesPerSec,
+                        const uint32_t totalDelayMS,
+                        const int32_t clockDrift,
+                        const uint32_t currentMicLevel,
+                        const bool keyPressed,
+                        uint32_t& newMicLevel));
+
+  MOCK_METHOD8(NeedMorePlayData,
+               int32_t(const size_t nSamples,
+                       const size_t nBytesPerSample,
+                       const size_t nChannels,
+                       const uint32_t samplesPerSec,
+                       void* audioSamples,
+                       size_t& nSamplesOut,
+                       int64_t* elapsed_time_ms,
+                       int64_t* ntp_time_ms));
+
+  MOCK_METHOD6(PushCaptureData,
+               void(int voe_channel,
+                    const void* audio_data,
+                    int bits_per_sample,
+                    int sample_rate,
+                    size_t number_of_channels,
+                    size_t number_of_frames));
+
+  MOCK_METHOD7(PullRenderData,
+               void(int bits_per_sample,
+                    int sample_rate,
+                    size_t number_of_channels,
+                    size_t number_of_frames,
+                    void* audio_data,
+                    int64_t* elapsed_time_ms,
+                    int64_t* ntp_time_ms));
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_MOCK_AUDIO_TRANSPORT_H_