blob: 308ee1224e885f6a127264d2a0cd1b1b3197b4a2 [file] [log] [blame]
/*
* Copyright (c) 2016 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/gain_control_for_experimental_agc.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/criticalsection.h"
namespace webrtc {
int GainControlForExperimentalAgc::instance_counter_ = 0;
GainControlForExperimentalAgc::GainControlForExperimentalAgc(
GainControl* gain_control,
rtc::CriticalSection* crit_capture)
: data_dumper_(new ApmDataDumper(instance_counter_)),
real_gain_control_(gain_control),
volume_(0),
crit_capture_(crit_capture) {
instance_counter_++;
}
GainControlForExperimentalAgc::~GainControlForExperimentalAgc() = default;
int GainControlForExperimentalAgc::Enable(bool enable) {
return real_gain_control_->Enable(enable);
}
bool GainControlForExperimentalAgc::is_enabled() const {
return real_gain_control_->is_enabled();
}
int GainControlForExperimentalAgc::set_stream_analog_level(int level) {
rtc::CritScope cs_capture(crit_capture_);
data_dumper_->DumpRaw("experimental_gain_control_set_stream_analog_level", 1,
&level);
volume_ = level;
return AudioProcessing::kNoError;
}
int GainControlForExperimentalAgc::stream_analog_level() {
rtc::CritScope cs_capture(crit_capture_);
data_dumper_->DumpRaw("experimental_gain_control_stream_analog_level", 1,
&volume_);
return volume_;
}
int GainControlForExperimentalAgc::set_mode(Mode mode) {
return AudioProcessing::kNoError;
}
GainControl::Mode GainControlForExperimentalAgc::mode() const {
return GainControl::kAdaptiveAnalog;
}
int GainControlForExperimentalAgc::set_target_level_dbfs(int level) {
return AudioProcessing::kNoError;
}
int GainControlForExperimentalAgc::target_level_dbfs() const {
return real_gain_control_->target_level_dbfs();
}
int GainControlForExperimentalAgc::set_compression_gain_db(int gain) {
return AudioProcessing::kNoError;
}
int GainControlForExperimentalAgc::compression_gain_db() const {
return real_gain_control_->compression_gain_db();
}
int GainControlForExperimentalAgc::enable_limiter(bool enable) {
return AudioProcessing::kNoError;
}
bool GainControlForExperimentalAgc::is_limiter_enabled() const {
return real_gain_control_->is_limiter_enabled();
}
int GainControlForExperimentalAgc::set_analog_level_limits(int minimum,
int maximum) {
return AudioProcessing::kNoError;
}
int GainControlForExperimentalAgc::analog_level_minimum() const {
return real_gain_control_->analog_level_minimum();
}
int GainControlForExperimentalAgc::analog_level_maximum() const {
return real_gain_control_->analog_level_maximum();
}
bool GainControlForExperimentalAgc::stream_is_saturated() const {
return real_gain_control_->stream_is_saturated();
}
void GainControlForExperimentalAgc::SetMicVolume(int volume) {
rtc::CritScope cs_capture(crit_capture_);
volume_ = volume;
}
int GainControlForExperimentalAgc::GetMicVolume() {
rtc::CritScope cs_capture(crit_capture_);
return volume_;
}
void GainControlForExperimentalAgc::Initialize() {
data_dumper_->InitiateNewSetOfRecordings();
}
} // namespace webrtc