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@3722 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2013-03-25 21:20:38 +00:00
parent b87cc85beb
commit c83a00ad49
5 changed files with 437 additions and 0 deletions

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/*
* 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_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
namespace webrtc {
class MockEchoCancellation : public EchoCancellation {
public:
MOCK_METHOD1(Enable,
int(bool enable));
MOCK_CONST_METHOD0(is_enabled,
bool());
MOCK_METHOD1(enable_drift_compensation,
int(bool enable));
MOCK_CONST_METHOD0(is_drift_compensation_enabled,
bool());
MOCK_METHOD1(set_device_sample_rate_hz,
int(int rate));
MOCK_CONST_METHOD0(device_sample_rate_hz,
int());
MOCK_METHOD1(set_stream_drift_samples,
void(int drift));
MOCK_CONST_METHOD0(stream_drift_samples,
int());
MOCK_METHOD1(set_suppression_level,
int(SuppressionLevel level));
MOCK_CONST_METHOD0(suppression_level,
SuppressionLevel());
MOCK_CONST_METHOD0(stream_has_echo,
bool());
MOCK_METHOD1(enable_metrics,
int(bool enable));
MOCK_CONST_METHOD0(are_metrics_enabled,
bool());
MOCK_METHOD1(GetMetrics,
int(Metrics* metrics));
MOCK_METHOD1(enable_delay_logging,
int(bool enable));
MOCK_CONST_METHOD0(is_delay_logging_enabled,
bool());
MOCK_METHOD2(GetDelayMetrics,
int(int* median, int* std));
MOCK_CONST_METHOD0(aec_core,
struct AecCore*());
};
class MockEchoControlMobile : public EchoControlMobile {
public:
MOCK_METHOD1(Enable,
int(bool enable));
MOCK_CONST_METHOD0(is_enabled,
bool());
MOCK_METHOD1(set_routing_mode,
int(RoutingMode mode));
MOCK_CONST_METHOD0(routing_mode,
RoutingMode());
MOCK_METHOD1(enable_comfort_noise,
int(bool enable));
MOCK_CONST_METHOD0(is_comfort_noise_enabled,
bool());
MOCK_METHOD2(SetEchoPath,
int(const void* echo_path, size_t size_bytes));
MOCK_CONST_METHOD2(GetEchoPath,
int(void* echo_path, size_t size_bytes));
};
class MockGainControl : public GainControl {
public:
MOCK_METHOD1(Enable,
int(bool enable));
MOCK_CONST_METHOD0(is_enabled,
bool());
MOCK_METHOD1(set_stream_analog_level,
int(int level));
MOCK_METHOD0(stream_analog_level,
int());
MOCK_METHOD1(set_mode,
int(Mode mode));
MOCK_CONST_METHOD0(mode,
Mode());
MOCK_METHOD1(set_target_level_dbfs,
int(int level));
MOCK_CONST_METHOD0(target_level_dbfs,
int());
MOCK_METHOD1(set_compression_gain_db,
int(int gain));
MOCK_CONST_METHOD0(compression_gain_db,
int());
MOCK_METHOD1(enable_limiter,
int(bool enable));
MOCK_CONST_METHOD0(is_limiter_enabled,
bool());
MOCK_METHOD2(set_analog_level_limits,
int(int minimum, int maximum));
MOCK_CONST_METHOD0(analog_level_minimum,
int());
MOCK_CONST_METHOD0(analog_level_maximum,
int());
MOCK_CONST_METHOD0(stream_is_saturated,
bool());
};
class MockHighPassFilter : public HighPassFilter {
public:
MOCK_METHOD1(Enable,
int(bool enable));
MOCK_CONST_METHOD0(is_enabled,
bool());
};
class MockLevelEstimator : public LevelEstimator {
public:
MOCK_METHOD1(Enable,
int(bool enable));
MOCK_CONST_METHOD0(is_enabled,
bool());
MOCK_METHOD0(RMS,
int());
};
class MockNoiseSuppression : public NoiseSuppression {
public:
MOCK_METHOD1(Enable,
int(bool enable));
MOCK_CONST_METHOD0(is_enabled,
bool());
MOCK_METHOD1(set_level,
int(Level level));
MOCK_CONST_METHOD0(level,
Level());
MOCK_CONST_METHOD0(speech_probability,
float());
};
class MockVoiceDetection : public VoiceDetection {
public:
MOCK_METHOD1(Enable,
int(bool enable));
MOCK_CONST_METHOD0(is_enabled,
bool());
MOCK_CONST_METHOD0(stream_has_voice,
bool());
MOCK_METHOD1(set_stream_has_voice,
int(bool has_voice));
MOCK_METHOD1(set_likelihood,
int(Likelihood likelihood));
MOCK_CONST_METHOD0(likelihood,
Likelihood());
MOCK_METHOD1(set_frame_size_ms,
int(int size));
MOCK_CONST_METHOD0(frame_size_ms,
int());
};
class MockAudioProcessing : public AudioProcessing {
public:
MockAudioProcessing()
: echo_cancellation_(new MockEchoCancellation),
echo_control_mobile_(new MockEchoControlMobile),
gain_control_(new MockGainControl),
high_pass_filter_(new MockHighPassFilter),
level_estimator_(new MockLevelEstimator),
noise_suppression_(new MockNoiseSuppression),
voice_detection_(new MockVoiceDetection) {
}
virtual ~MockAudioProcessing() {
}
MOCK_METHOD0(Initialize,
int());
MOCK_METHOD1(set_sample_rate_hz,
int(int rate));
MOCK_CONST_METHOD0(sample_rate_hz,
int());
MOCK_METHOD2(set_num_channels,
int(int input_channels, int output_channels));
MOCK_CONST_METHOD0(num_input_channels,
int());
MOCK_CONST_METHOD0(num_output_channels,
int());
MOCK_METHOD1(set_num_reverse_channels,
int(int channels));
MOCK_CONST_METHOD0(num_reverse_channels,
int());
MOCK_METHOD1(ProcessStream,
int(AudioFrame* frame));
MOCK_METHOD1(AnalyzeReverseStream,
int(AudioFrame* frame));
MOCK_METHOD1(set_stream_delay_ms,
int(int delay));
MOCK_CONST_METHOD0(stream_delay_ms,
int());
MOCK_METHOD1(set_delay_offset_ms,
void(int offset));
MOCK_CONST_METHOD0(delay_offset_ms,
int());
MOCK_METHOD1(StartDebugRecording,
int(const char filename[kMaxFilenameSize]));
MOCK_METHOD0(StopDebugRecording,
int());
virtual MockEchoCancellation* echo_cancellation() const {
return echo_cancellation_.get();
}
virtual MockEchoControlMobile* echo_control_mobile() const {
return echo_control_mobile_.get();
}
virtual MockGainControl* gain_control() const {
return gain_control_.get();
}
virtual MockHighPassFilter* high_pass_filter() const {
return high_pass_filter_.get();
};
virtual MockLevelEstimator* level_estimator() const {
return level_estimator_.get();
};
virtual MockNoiseSuppression* noise_suppression() const {
return noise_suppression_.get();
};
virtual MockVoiceDetection* voice_detection() const {
return voice_detection_.get();
};
MOCK_METHOD0(TimeUntilNextProcess,
WebRtc_Word32());
MOCK_METHOD0(Process,
WebRtc_Word32());
private:
scoped_ptr<MockEchoCancellation> echo_cancellation_;
scoped_ptr<MockEchoControlMobile> echo_control_mobile_;
scoped_ptr<MockGainControl> gain_control_;
scoped_ptr<MockHighPassFilter> high_pass_filter_;
scoped_ptr<MockLevelEstimator> level_estimator_;
scoped_ptr<MockNoiseSuppression> noise_suppression_;
scoped_ptr<MockVoiceDetection> voice_detection_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_

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webrtc/test/fake_common.h Normal file
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/*
* 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_TEST_FAKE_COMMON_H_
#define WEBRTC_TEST_FAKE_COMMON_H_
// Borrowed from libjingle's talk/media/webrtc/fakewebrtccommon.h.
#include "webrtc/typedefs.h"
#define WEBRTC_STUB(method, args) \
virtual int method args OVERRIDE { return 0; }
#define WEBRTC_STUB_CONST(method, args) \
virtual int method args const OVERRIDE { return 0; }
#define WEBRTC_BOOL_STUB(method, args) \
virtual bool method args OVERRIDE { return true; }
#define WEBRTC_VOID_STUB(method, args) \
virtual void method args OVERRIDE {}
#define WEBRTC_FUNC(method, args) \
virtual int method args OVERRIDE
#define WEBRTC_FUNC_CONST(method, args) \
virtual int method args const OVERRIDE
#define WEBRTC_BOOL_FUNC(method, args) \
virtual bool method args OVERRIDE
#define WEBRTC_VOID_FUNC(method, args) \
virtual void method args OVERRIDE
#define WEBRTC_CHECK_CHANNEL(channel) \
if (channels_.find(channel) == channels_.end()) return -1;
#define WEBRTC_ASSERT_CHANNEL(channel) \
ASSERT(channels_.find(channel) != channels_.end());
#endif // WEBRTC_TEST_FAKE_COMMON_H_

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@ -90,4 +90,17 @@ typedef uint16_t WebRtc_UWord16;
typedef uint32_t WebRtc_UWord32;
typedef uint64_t WebRtc_UWord64;
// Borrowed from Chromium's base/compiler_specific.h.
// Annotate a virtual method indicating it must be overriding a virtual
// method in the parent class.
// Use like:
// virtual void foo() OVERRIDE;
#if defined(_MSC_VER)
#define OVERRIDE override
#elif defined(__clang__)
#define OVERRIDE override
#else
#define OVERRIDE
#endif
#endif // WEBRTC_TYPEDEFS_H_

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/*
* 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_

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/*
* 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_