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platform-external-webrtc/webrtc/modules/audio_processing/include/mock_audio_processing.h
Bjorn Volcker 4e7aa43ea0 audio_processing: Adds two UMA histograms logging delay jumps in AEC 
We have two histograms today that trigger on large jumps in either platform reported stream delays (WebRTC.Audio.PlatformReportedStreamDelayJump) or the system delay in the AEC (WebRTC.Audio.AecSystemDelayJump). The latter is the internal buffer size in the AEC.
The sizes of such jumps are of relevance since it can harm the AEC and even put it in a complete failure state. It is hard, not to say impossible, to tell how frequent it is.
Therefore, two complementary histograms are added; number of jumps in each metric.
This way we get a quick way to determine how often a jump occurs in general and also how frequent it is within a call.

This is solved by adding a counter for each metric.
The counter is activated either upon an event trigger or if we know for sure when the AEC is running.
Unfortunately, we can't rely on the destructor at the end of a call so we add a public API for the user to take on the action of calling it at the end of a call.

Tested locally by building ToT chromium including changes and three triggered jumps (200, 50 and 60 ms).
The stats picked up the 60 and 200 ms jumps as expected.

BUG=488124
R=asapersson@webrtc.org, pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1229443003.

Cr-Commit-Position: refs/heads/master@{#9544}
2015-07-07 09:50:16 +00:00

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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/base/scoped_ptr.h"
#include "webrtc/modules/audio_processing/include/audio_processing.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_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_METHOD3(GetDelayMetrics,
int(int* median, int* std, float* fraction_poor_delays));
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_METHOD6(Initialize,
int(int sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
ChannelLayout input_layout,
ChannelLayout output_layout,
ChannelLayout reverse_layout));
MOCK_METHOD1(SetExtraOptions,
void(const Config& config));
MOCK_METHOD1(set_sample_rate_hz,
int(int rate));
MOCK_CONST_METHOD0(input_sample_rate_hz,
int());
MOCK_CONST_METHOD0(sample_rate_hz,
int());
MOCK_CONST_METHOD0(proc_sample_rate_hz,
int());
MOCK_CONST_METHOD0(proc_split_sample_rate_hz,
int());
MOCK_CONST_METHOD0(num_input_channels,
int());
MOCK_CONST_METHOD0(num_output_channels,
int());
MOCK_CONST_METHOD0(num_reverse_channels,
int());
MOCK_METHOD1(set_output_will_be_muted,
void(bool muted));
MOCK_CONST_METHOD0(output_will_be_muted,
bool());
MOCK_METHOD1(ProcessStream,
int(AudioFrame* frame));
MOCK_METHOD7(ProcessStream,
int(const float* const* src,
int samples_per_channel,
int input_sample_rate_hz,
ChannelLayout input_layout,
int output_sample_rate_hz,
ChannelLayout output_layout,
float* const* dest));
MOCK_METHOD1(AnalyzeReverseStream,
int(AudioFrame* frame));
MOCK_METHOD4(AnalyzeReverseStream,
int(const float* const* data, int frames, int sample_rate_hz,
ChannelLayout input_layout));
MOCK_METHOD1(set_stream_delay_ms,
int(int delay));
MOCK_CONST_METHOD0(stream_delay_ms,
int());
MOCK_CONST_METHOD0(was_stream_delay_set,
bool());
MOCK_METHOD1(set_stream_key_pressed,
void(bool key_pressed));
MOCK_CONST_METHOD0(stream_key_pressed,
bool());
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_METHOD1(StartDebugRecording,
int(FILE* handle));
MOCK_METHOD0(StopDebugRecording,
int());
MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
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();
}
private:
rtc::scoped_ptr<MockEchoCancellation> echo_cancellation_;
rtc::scoped_ptr<MockEchoControlMobile> echo_control_mobile_;
rtc::scoped_ptr<MockGainControl> gain_control_;
rtc::scoped_ptr<MockHighPassFilter> high_pass_filter_;
rtc::scoped_ptr<MockLevelEstimator> level_estimator_;
rtc::scoped_ptr<MockNoiseSuppression> noise_suppression_;
rtc::scoped_ptr<MockVoiceDetection> voice_detection_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_MOCK_AUDIO_PROCESSING_H_
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