3ec96df907
split domain and only implements floating point operations (to avoid spectral leakage issues and unnecessary complexity). The goal of this CL is adding the new sub-module into APM without providing an implementation that could replace the existing gain control modules. The focus is in fact on initialization, reset, and configuration of AGC2. The module itself only applies a hard-coded gain value. This behavior will change in the coming CLs. BUG=webrtc:7494 Review-Url: https://codereview.webrtc.org/2848593002 Cr-Commit-Position: refs/heads/master@{#18222}
66 lines
2.0 KiB
C++
66 lines
2.0 KiB
C++
/*
|
|
* 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 "webrtc/modules/audio_processing/agc2/gain_controller2.h"
|
|
|
|
#include "webrtc/base/atomicops.h"
|
|
#include "webrtc/base/checks.h"
|
|
#include "webrtc/modules/audio_processing/audio_buffer.h"
|
|
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
|
|
|
|
namespace webrtc {
|
|
|
|
namespace {
|
|
|
|
constexpr float kGain = 0.5f;
|
|
|
|
} // namespace
|
|
|
|
int GainController2::instance_count_ = 0;
|
|
|
|
GainController2::GainController2(int sample_rate_hz)
|
|
: sample_rate_hz_(sample_rate_hz),
|
|
data_dumper_(new ApmDataDumper(
|
|
rtc::AtomicOps::Increment(&instance_count_))),
|
|
digital_gain_applier_(),
|
|
gain_(kGain) {
|
|
RTC_DCHECK(sample_rate_hz_ == AudioProcessing::kSampleRate8kHz ||
|
|
sample_rate_hz_ == AudioProcessing::kSampleRate16kHz ||
|
|
sample_rate_hz_ == AudioProcessing::kSampleRate32kHz ||
|
|
sample_rate_hz_ == AudioProcessing::kSampleRate48kHz);
|
|
data_dumper_->InitiateNewSetOfRecordings();
|
|
data_dumper_->DumpRaw("gain_", 1, &gain_);
|
|
}
|
|
|
|
GainController2::~GainController2() = default;
|
|
|
|
void GainController2::Process(AudioBuffer* audio) {
|
|
for (size_t k = 0; k < audio->num_channels(); ++k) {
|
|
auto channel_view = rtc::ArrayView<float>(
|
|
audio->channels_f()[k], audio->num_frames());
|
|
digital_gain_applier_.Process(gain_, channel_view);
|
|
}
|
|
}
|
|
|
|
bool GainController2::Validate(
|
|
const AudioProcessing::Config::GainController2& config) {
|
|
return true;
|
|
}
|
|
|
|
std::string GainController2::ToString(
|
|
const AudioProcessing::Config::GainController2& config) {
|
|
std::stringstream ss;
|
|
ss << "{"
|
|
<< "enabled: " << (config.enabled ? "true" : "false") << "}";
|
|
return ss.str();
|
|
}
|
|
|
|
} // namespace webrtc
|