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/*
* Copyright (c) 2017 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.
*/
#include "modules/audio_processing/gain_controller2.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/atomicops.h"
#include "rtc_base/checks.h"
namespace webrtc {
int GainController2::instance_count_ = 0;
GainController2::GainController2()
: data_dumper_(
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
fixed_gain_controller_(data_dumper_.get()),
adaptive_agc_(data_dumper_.get()) {}
GainController2::~GainController2() = default;
void GainController2::Initialize(int sample_rate_hz) {
RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
sample_rate_hz == AudioProcessing::kSampleRate48kHz);
fixed_gain_controller_.SetSampleRate(sample_rate_hz);
data_dumper_->InitiateNewSetOfRecordings();
data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz);
}
void GainController2::Process(AudioBuffer* audio) {
AudioFrameView<float> float_frame(audio->channels_f(), audio->num_channels(),
audio->num_frames());
if (adaptive_digital_mode_) {
adaptive_agc_.Process(float_frame);
}
fixed_gain_controller_.Process(float_frame);
}
void GainController2::NotifyAnalogLevel(int level) {
if (analog_level_ != level && adaptive_digital_mode_) {
adaptive_agc_.Reset();
}
analog_level_ = level;
}
void GainController2::ApplyConfig(
const AudioProcessing::Config::GainController2& config) {
RTC_DCHECK(Validate(config));
config_ = config;
fixed_gain_controller_.SetGain(config_.fixed_gain_db);
adaptive_digital_mode_ = config_.adaptive_digital_mode;
}
bool GainController2::Validate(
const AudioProcessing::Config::GainController2& config) {
return config.fixed_gain_db >= 0.f;
}
std::string GainController2::ToString(
const AudioProcessing::Config::GainController2& config) {
std::stringstream ss;
ss << "{enabled: " << (config.enabled ? "true" : "false") << ", "
<< "fixed_gain_dB: " << config.fixed_gain_db << "}";
return ss.str();
}
} // namespace webrtc
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