mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 14:20:45 +01:00
e36e8bbf6d
The FixedGainController (FGC) applies a fixed gain. It will also control the limiter. The limiter will be landed over the next several CLs. The GainController2 is a 'private submodule' of APM. It will control the new automatic gain controller (AGC). It controls the AGC through Initialize() and ApplyConfig(). This CL contains * build changes to make modules/audio_processing/agc2 an independent target * a new MutableFloatAudioFrame which is the audio interface between AGC2 and APM * move of the fixed gain application from GainController2 to FixedGainController. If you are a googler, there is more information in this doc: https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit# Bug: webrtc:7949 Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a Reviewed-on: https://webrtc-review.googlesource.com/50440 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22046}
68 lines
2.4 KiB
C++
68 lines
2.4 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 "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_))),
|
|
gain_controller_(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);
|
|
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());
|
|
gain_controller_.Process(float_frame);
|
|
}
|
|
|
|
void GainController2::ApplyConfig(
|
|
const AudioProcessing::Config::GainController2& config) {
|
|
RTC_DCHECK(Validate(config));
|
|
config_ = config;
|
|
gain_controller_.SetGain(config_.fixed_gain_db);
|
|
gain_controller_.EnableLimiter(config_.enable_limiter);
|
|
}
|
|
|
|
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 << ", "
|
|
<< "enable_limiter: " << (config.enable_limiter ? "true" : "false") << "}";
|
|
return ss.str();
|
|
}
|
|
|
|
} // namespace webrtc
|