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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 MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
#define MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
#include <memory>
#include "modules/audio_processing/agc/agc.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/gtest_prod_util.h"
namespace webrtc {
class AudioFrame;
class DebugFile;
class GainControl;
// Callbacks that need to be injected into AgcManagerDirect to read and control
// the volume values. This is done to remove the VoiceEngine dependency in
// AgcManagerDirect.
// TODO(aluebs): Remove VolumeCallbacks.
class VolumeCallbacks {
public:
virtual ~VolumeCallbacks() {}
virtual void SetMicVolume(int volume) = 0;
virtual int GetMicVolume() = 0;
};
// Direct interface to use AGC to set volume and compression values.
// AudioProcessing uses this interface directly to integrate the callback-less
// AGC.
//
// This class is not thread-safe.
class AgcManagerDirect final {
public:
// AgcManagerDirect will configure GainControl internally. The user is
// responsible for processing the audio using it after the call to Process.
// The operating range of startup_min_level is [12, 255] and any input value
// outside that range will be clamped.
AgcManagerDirect(GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level,
int clipped_level_min,
bool use_agc2_level_estimation,
bool disable_digital_adaptive);
~AgcManagerDirect();
int Initialize();
void AnalyzePreProcess(int16_t* audio,
int num_channels,
size_t samples_per_channel);
void Process(const int16_t* audio, size_t length, int sample_rate_hz);
// Call when the capture stream has been muted/unmuted. This causes the
// manager to disregard all incoming audio; chances are good it's background
// noise to which we'd like to avoid adapting.
void SetCaptureMuted(bool muted);
bool capture_muted() { return capture_muted_; }
float voice_probability();
private:
friend class AgcManagerDirectTest;
FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
DisableDigitalDisablesDigital);
// Dependency injection for testing. Don't delete |agc| as the memory is owned
// by the manager.
AgcManagerDirect(Agc* agc,
GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level,
int clipped_level_min);
// Most general c-tor.
AgcManagerDirect(Agc* agc,
GainControl* gctrl,
VolumeCallbacks* volume_callbacks,
int startup_min_level,
int clipped_level_min,
bool use_agc2_level_estimation,
bool disable_digital_adaptive);
// Sets a new microphone level, after first checking that it hasn't been
// updated by the user, in which case no action is taken.
void SetLevel(int new_level);
// Set the maximum level the AGC is allowed to apply. Also updates the
// maximum compression gain to compensate. The level must be at least
// |kClippedLevelMin|.
void SetMaxLevel(int level);
int CheckVolumeAndReset();
void UpdateGain();
void UpdateCompressor();
std::unique_ptr<ApmDataDumper> data_dumper_;
static int instance_counter_;
std::unique_ptr<Agc> agc_;
GainControl* gctrl_;
VolumeCallbacks* volume_callbacks_;
int frames_since_clipped_;
int level_;
int max_level_;
int max_compression_gain_;
int target_compression_;
int compression_;
float compression_accumulator_;
bool capture_muted_;
bool check_volume_on_next_process_;
bool startup_;
const bool use_agc2_level_estimation_;
const bool disable_digital_adaptive_;
int startup_min_level_;
const int clipped_level_min_;
int calls_since_last_gain_log_ = 0;
std::unique_ptr<DebugFile> file_preproc_;
std::unique_ptr<DebugFile> file_postproc_;
RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_
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