blob: 164ca3c88ad23c5b22c2af5b144c7e11fae8a9cc [file] [log] [blame]
/*
* 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 "media/engine/apm_helpers.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/include/gain_control.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace apm_helpers {
void Init(AudioProcessing* apm) {
RTC_DCHECK(apm);
constexpr int kMinVolumeLevel = 0;
constexpr int kMaxVolumeLevel = 255;
// This is the initialization which used to happen in VoEBase::Init(), but
// which is not covered by the WVoE::ApplyOptions().
GainControl* gc = apm->gain_control();
if (gc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
RTC_DLOG(LS_ERROR) << "Failed to set analog level limits with minimum: "
<< kMinVolumeLevel
<< " and maximum: " << kMaxVolumeLevel;
}
}
AgcConfig GetAgcConfig(AudioProcessing* apm) {
RTC_DCHECK(apm);
AgcConfig result;
result.targetLeveldBOv = apm->gain_control()->target_level_dbfs();
result.digitalCompressionGaindB = apm->gain_control()->compression_gain_db();
result.limiterEnable = apm->gain_control()->is_limiter_enabled();
return result;
}
void SetAgcConfig(AudioProcessing* apm, const AgcConfig& config) {
RTC_DCHECK(apm);
GainControl* gc = apm->gain_control();
if (gc->set_target_level_dbfs(config.targetLeveldBOv) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set target level: "
<< config.targetLeveldBOv;
}
if (gc->set_compression_gain_db(config.digitalCompressionGaindB) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set compression gain: "
<< config.digitalCompressionGaindB;
}
if (gc->enable_limiter(config.limiterEnable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set limiter on/off: "
<< config.limiterEnable;
}
}
void SetAgcStatus(AudioProcessing* apm, bool enable) {
RTC_DCHECK(apm);
#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
GainControl::Mode agc_mode = GainControl::kFixedDigital;
#else
GainControl::Mode agc_mode = GainControl::kAdaptiveAnalog;
#endif
GainControl* gc = apm->gain_control();
if (gc->set_mode(agc_mode) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set AGC mode: " << agc_mode;
return;
}
if (gc->Enable(enable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable AGC: " << enable;
return;
}
RTC_LOG(LS_INFO) << "AGC set to " << enable << " with mode " << agc_mode;
}
void SetEcStatus(AudioProcessing* apm, bool enable, EcModes mode) {
RTC_DCHECK(apm);
RTC_DCHECK(mode == kEcConference || mode == kEcAecm) << "mode: " << mode;
AudioProcessing::Config apm_config = apm->GetConfig();
apm_config.echo_canceller.enabled = enable;
apm_config.echo_canceller.mobile_mode = (mode == kEcAecm);
apm_config.echo_canceller.legacy_moderate_suppression_level = false;
apm->ApplyConfig(apm_config);
RTC_LOG(LS_INFO) << "Echo control set to " << enable << " with mode " << mode;
}
void SetNsStatus(AudioProcessing* apm, bool enable) {
RTC_DCHECK(apm);
NoiseSuppression* ns = apm->noise_suppression();
if (ns->set_level(NoiseSuppression::kHigh) != 0) {
RTC_LOG(LS_ERROR) << "Failed to set high NS level.";
return;
}
if (ns->Enable(enable) != 0) {
RTC_LOG(LS_ERROR) << "Failed to enable/disable NS: " << enable;
return;
}
RTC_LOG(LS_INFO) << "NS set to " << enable;
}
} // namespace apm_helpers
} // namespace webrtc