Add some VoE and AudioProcessing mocks.

Includes a bit of shared helpers in fake_common.h.

Review URL: https://webrtc-codereview.appspot.com/1221004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3722 4adac7df-926f-26a2-2b94-8c16560cd09d

2 files changed, 124 insertions, 0 deletions

diff --git a/voice_engine/include/mock/fake_voe_external_media.h b/voice_engine/include/mock/fake_voe_external_media.h
new file mode 100644
index 00000000..8b608ec2
--- /dev/null
+++ b/voice_engine/include/mock/fake_voe_external_media.h
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2013 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_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
+#define WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
+
+#include <map>
+
+#include "webrtc/test/fake_common.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/voice_engine/include/voe_external_media.h"
+
+namespace webrtc {
+
+class FakeVoEExternalMedia : public VoEExternalMedia {
+ public:
+  FakeVoEExternalMedia() {}
+  virtual ~FakeVoEExternalMedia() {}
+
+  WEBRTC_STUB(Release, ());
+  WEBRTC_FUNC(RegisterExternalMediaProcessing,
+      (int channel, ProcessingTypes type, VoEMediaProcess& processObject)) {
+    callback_map_[type] = &processObject;
+    return 0;
+  }
+  WEBRTC_FUNC(DeRegisterExternalMediaProcessing,
+      (int channel, ProcessingTypes type)) {
+    callback_map_.erase(type);
+    return 0;
+  }
+  WEBRTC_STUB(SetExternalRecordingStatus, (bool enable));
+  WEBRTC_STUB(SetExternalPlayoutStatus, (bool enable));
+  WEBRTC_STUB(ExternalRecordingInsertData,
+      (const WebRtc_Word16 speechData10ms[], int lengthSamples,
+       int samplingFreqHz, int current_delay_ms));
+  WEBRTC_STUB(ExternalPlayoutGetData,
+      (WebRtc_Word16 speechData10ms[], int samplingFreqHz,
+       int current_delay_ms, int& lengthSamples));
+  WEBRTC_STUB(GetAudioFrame, (int channel, int desired_sample_rate_hz,
+                              AudioFrame* frame));
+  WEBRTC_STUB(SetExternalMixing, (int channel, bool enable));
+
+  // Use this to trigger the Process() callback to a registered media processor.
+  // If |audio| is NULL, a zero array of the correct length will be forwarded.
+  void CallProcess(ProcessingTypes type, int16_t* audio,
+                   int samples_per_channel, int sample_rate_hz,
+                   int num_channels) {
+    const int length = samples_per_channel * num_channels;
+    scoped_array<int16_t> data;
+    if (!audio) {
+      data.reset(new int16_t[length]);
+      memset(data.get(), 0, length * sizeof(data[0]));
+      audio = data.get();
+    }
+
+    std::map<ProcessingTypes, VoEMediaProcess*>::const_iterator it =
+        callback_map_.find(type);
+    if (it != callback_map_.end()) {
+      it->second->Process(0, type, audio, samples_per_channel, sample_rate_hz,
+                          num_channels == 2 ? true : false);
+    }
+  }
+
+ private:
+  std::map<ProcessingTypes, VoEMediaProcess*> callback_map_;
+};
+
+}  // namespace webrtc
+
+#endif  //  WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_FAKE_VOE_EXTERNAL_MEDIA_H_
diff --git a/voice_engine/include/mock/mock_voe_volume_control.h b/voice_engine/include/mock/mock_voe_volume_control.h
new file mode 100644
index 00000000..20b09696
--- /dev/null
+++ b/voice_engine/include/mock/mock_voe_volume_control.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2013 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_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_
+#define WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "webrtc/voice_engine/include/voe_volume_control.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+class MockVoEVolumeControl : public VoEVolumeControl {
+ public:
+  MOCK_METHOD0(Release, int());
+  MOCK_METHOD1(SetSpeakerVolume, int(unsigned int volume));
+  MOCK_METHOD1(GetSpeakerVolume, int(unsigned int& volume));
+  MOCK_METHOD1(SetSystemOutputMute, int(bool enable));
+  MOCK_METHOD1(GetSystemOutputMute, int(bool &enabled));
+  MOCK_METHOD1(SetMicVolume, int(unsigned int volume));
+  MOCK_METHOD1(GetMicVolume, int(unsigned int& volume));
+  MOCK_METHOD2(SetInputMute, int(int channel, bool enable));
+  MOCK_METHOD2(GetInputMute, int(int channel, bool& enabled));
+  MOCK_METHOD1(SetSystemInputMute, int(bool enable));
+  MOCK_METHOD1(GetSystemInputMute, int(bool& enabled));
+  MOCK_METHOD1(GetSpeechInputLevel, int(unsigned int& level));
+  MOCK_METHOD2(GetSpeechOutputLevel, int(int channel, unsigned int& level));
+  MOCK_METHOD1(GetSpeechInputLevelFullRange, int(unsigned int& level));
+  MOCK_METHOD2(GetSpeechOutputLevelFullRange,
+               int(int channel, unsigned int& level));
+  MOCK_METHOD2(SetChannelOutputVolumeScaling, int(int channel, float scaling));
+  MOCK_METHOD2(GetChannelOutputVolumeScaling, int(int channel, float& scaling));
+  MOCK_METHOD3(SetOutputVolumePan, int(int channel, float left, float right));
+  MOCK_METHOD3(GetOutputVolumePan, int(int channel, float& left, float& right));
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_VOICE_ENGINE_INCLUDE_MOCK_MOCK_VOE_VOLUME_CONTROL_H_