7988e5cbbf
Some of the AEC settings in WebRtcVoiceEngine agree with just about everywhere else and have therefore been set in stone inside AEC2/AECM. A lot of routing for those settings disappears. The comfort noise setting is exposed in the API, so the flag for it will be removed a PSA later. Bug: webrtc:9535 Change-Id: I53816152415a9a069cea9520cec697b6bcfe0948 Reviewed-on: https://webrtc-review.googlesource.com/101622 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24863}
121 lines
4.2 KiB
C++
121 lines
4.2 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 "media/engine/apm_helpers.h"
|
|
|
|
#include "modules/audio_device/include/audio_device.h"
|
|
#include "modules/audio_processing/include/audio_processing.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;
|
|
}
|
|
|
|
void SetTypingDetectionStatus(AudioProcessing* apm, bool enable) {
|
|
RTC_DCHECK(apm);
|
|
VoiceDetection* vd = apm->voice_detection();
|
|
if (vd->Enable(enable)) {
|
|
RTC_LOG(LS_ERROR) << "Failed to enable/disable VAD: " << enable;
|
|
return;
|
|
}
|
|
if (vd->set_likelihood(VoiceDetection::kVeryLowLikelihood)) {
|
|
RTC_LOG(LS_ERROR) << "Failed to set low VAD likelihood.";
|
|
return;
|
|
}
|
|
RTC_LOG(LS_INFO) << "VAD set to " << enable << " for typing detection.";
|
|
}
|
|
} // namespace apm_helpers
|
|
} // namespace webrtc
|