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platform-external-webrtc/webrtc/modules/audio_processing/audio_processing_impl.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) 2012 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_AUDIO_PROCESSING_IMPL_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#include <list>
#include <string>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class AgcManagerDirect;
class AudioBuffer;
template<typename T>
class Beamformer;
class CriticalSectionWrapper;
class EchoCancellationImpl;
class EchoControlMobileImpl;
class FileWrapper;
class GainControlImpl;
class GainControlForNewAgc;
class HighPassFilterImpl;
class LevelEstimatorImpl;
class NoiseSuppressionImpl;
class ProcessingComponent;
class TransientSuppressor;
class VoiceDetectionImpl;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
namespace audioproc {
class Event;
} // namespace audioproc
#endif
class AudioRate {
public:
explicit AudioRate(int sample_rate_hz) { set(sample_rate_hz); }
virtual ~AudioRate() {}
void set(int rate) {
rate_ = rate;
samples_per_channel_ = AudioProcessing::kChunkSizeMs * rate_ / 1000;
}
int rate() const { return rate_; }
int samples_per_channel() const { return samples_per_channel_; }
private:
int rate_;
int samples_per_channel_;
};
class AudioFormat : public AudioRate {
public:
AudioFormat(int sample_rate_hz, int num_channels)
: AudioRate(sample_rate_hz),
num_channels_(num_channels) {}
virtual ~AudioFormat() {}
void set(int rate, int num_channels) {
AudioRate::set(rate);
num_channels_ = num_channels;
}
int num_channels() const { return num_channels_; }
private:
int num_channels_;
};
class AudioProcessingImpl : public AudioProcessing {
public:
explicit AudioProcessingImpl(const Config& config);
// AudioProcessingImpl takes ownership of beamformer.
AudioProcessingImpl(const Config& config, Beamformer<float>* beamformer);
virtual ~AudioProcessingImpl();
// AudioProcessing methods.
int Initialize() override;
int Initialize(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
ChannelLayout input_layout,
ChannelLayout output_layout,
ChannelLayout reverse_layout) override;
void SetExtraOptions(const Config& config) override;
int set_sample_rate_hz(int rate) override;
int input_sample_rate_hz() const override;
int sample_rate_hz() const override;
int proc_sample_rate_hz() const override;
int proc_split_sample_rate_hz() const override;
int num_input_channels() const override;
int num_output_channels() const override;
int num_reverse_channels() const override;
void set_output_will_be_muted(bool muted) override;
bool output_will_be_muted() const override;
int ProcessStream(AudioFrame* frame) override;
int ProcessStream(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) override;
int AnalyzeReverseStream(AudioFrame* frame) override;
int AnalyzeReverseStream(const float* const* data,
int samples_per_channel,
int sample_rate_hz,
ChannelLayout layout) override;
int set_stream_delay_ms(int delay) override;
int stream_delay_ms() const override;
bool was_stream_delay_set() const override;
void set_delay_offset_ms(int offset) override;
int delay_offset_ms() const override;
void set_stream_key_pressed(bool key_pressed) override;
bool stream_key_pressed() const override;
int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
int StartDebugRecording(FILE* handle) override;
int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
int StopDebugRecording() override;
void UpdateHistogramsOnCallEnd() override;
EchoCancellation* echo_cancellation() const override;
EchoControlMobile* echo_control_mobile() const override;
GainControl* gain_control() const override;
HighPassFilter* high_pass_filter() const override;
LevelEstimator* level_estimator() const override;
NoiseSuppression* noise_suppression() const override;
VoiceDetection* voice_detection() const override;
protected:
// Overridden in a mock.
virtual int InitializeLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
private:
int InitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
int MaybeInitializeLocked(int input_sample_rate_hz,
int output_sample_rate_hz,
int reverse_sample_rate_hz,
int num_input_channels,
int num_output_channels,
int num_reverse_channels)
EXCLUSIVE_LOCKS_REQUIRED(crit_);
int ProcessStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
int AnalyzeReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_);
bool is_data_processed() const;
bool output_copy_needed(bool is_data_processed) const;
bool synthesis_needed(bool is_data_processed) const;
bool analysis_needed(bool is_data_processed) const;
void InitializeExperimentalAgc() EXCLUSIVE_LOCKS_REQUIRED(crit_);
void InitializeTransient() EXCLUSIVE_LOCKS_REQUIRED(crit_);
void InitializeBeamformer() EXCLUSIVE_LOCKS_REQUIRED(crit_);
void MaybeUpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
EchoCancellationImpl* echo_cancellation_;
EchoControlMobileImpl* echo_control_mobile_;
GainControlImpl* gain_control_;
HighPassFilterImpl* high_pass_filter_;
LevelEstimatorImpl* level_estimator_;
NoiseSuppressionImpl* noise_suppression_;
VoiceDetectionImpl* voice_detection_;
rtc::scoped_ptr<GainControlForNewAgc> gain_control_for_new_agc_;
std::list<ProcessingComponent*> component_list_;
CriticalSectionWrapper* crit_;
rtc::scoped_ptr<AudioBuffer> render_audio_;
rtc::scoped_ptr<AudioBuffer> capture_audio_;
#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
// TODO(andrew): make this more graceful. Ideally we would split this stuff
// out into a separate class with an "enabled" and "disabled" implementation.
int WriteMessageToDebugFile();
int WriteInitMessage();
rtc::scoped_ptr<FileWrapper> debug_file_;
rtc::scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
std::string event_str_; // Memory for protobuf serialization.
#endif
AudioFormat fwd_in_format_;
// This one is an AudioRate, because the forward processing number of channels
// is mutable and is tracked by the capture_audio_.
AudioRate fwd_proc_format_;
AudioFormat fwd_out_format_;
AudioFormat rev_in_format_;
AudioFormat rev_proc_format_;
int split_rate_;
int stream_delay_ms_;
int delay_offset_ms_;
bool was_stream_delay_set_;
int last_stream_delay_ms_;
int last_aec_system_delay_ms_;
int stream_delay_jumps_;
int aec_system_delay_jumps_;
bool output_will_be_muted_ GUARDED_BY(crit_);
bool key_pressed_;
// Only set through the constructor's Config parameter.
const bool use_new_agc_;
rtc::scoped_ptr<AgcManagerDirect> agc_manager_ GUARDED_BY(crit_);
int agc_startup_min_volume_;
bool transient_suppressor_enabled_;
rtc::scoped_ptr<TransientSuppressor> transient_suppressor_;
const bool beamformer_enabled_;
rtc::scoped_ptr<Beamformer<float>> beamformer_;
const std::vector<Point> array_geometry_;
const bool supports_48kHz_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
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