ba50223363
Configuration for AudioProcessing::voice_detection() is moving into AudioProcessing::Config, to get rid of the pointer-to-submodule interfaces (such as voice_detection()). Bug: webrtc:9947 Change-Id: Ia64ae996a43d44423aa0d612a3f1185b52a3e534 Reviewed-on: https://webrtc-review.googlesource.com/c/116067 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26216}
108 lines
3.7 KiB
C++
108 lines
3.7 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "media/engine/apm_helpers.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "modules/audio_processing/include/gain_control.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace apm_helpers {
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void Init(AudioProcessing* apm) {
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RTC_DCHECK(apm);
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constexpr int kMinVolumeLevel = 0;
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constexpr int kMaxVolumeLevel = 255;
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// This is the initialization which used to happen in VoEBase::Init(), but
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// which is not covered by the WVoE::ApplyOptions().
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GainControl* gc = apm->gain_control();
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if (gc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
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RTC_DLOG(LS_ERROR) << "Failed to set analog level limits with minimum: "
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<< kMinVolumeLevel
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<< " and maximum: " << kMaxVolumeLevel;
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}
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}
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AgcConfig GetAgcConfig(AudioProcessing* apm) {
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RTC_DCHECK(apm);
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AgcConfig result;
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result.targetLeveldBOv = apm->gain_control()->target_level_dbfs();
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result.digitalCompressionGaindB = apm->gain_control()->compression_gain_db();
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result.limiterEnable = apm->gain_control()->is_limiter_enabled();
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return result;
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}
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void SetAgcConfig(AudioProcessing* apm, const AgcConfig& config) {
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RTC_DCHECK(apm);
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GainControl* gc = apm->gain_control();
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if (gc->set_target_level_dbfs(config.targetLeveldBOv) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set target level: "
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<< config.targetLeveldBOv;
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}
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if (gc->set_compression_gain_db(config.digitalCompressionGaindB) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set compression gain: "
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<< config.digitalCompressionGaindB;
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}
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if (gc->enable_limiter(config.limiterEnable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set limiter on/off: "
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<< config.limiterEnable;
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}
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}
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void SetAgcStatus(AudioProcessing* apm, bool enable) {
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RTC_DCHECK(apm);
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#if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
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GainControl::Mode agc_mode = GainControl::kFixedDigital;
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#else
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GainControl::Mode agc_mode = GainControl::kAdaptiveAnalog;
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#endif
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GainControl* gc = apm->gain_control();
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if (gc->set_mode(agc_mode) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set AGC mode: " << agc_mode;
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return;
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}
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if (gc->Enable(enable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable AGC: " << enable;
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return;
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}
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RTC_LOG(LS_INFO) << "AGC set to " << enable << " with mode " << agc_mode;
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}
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void SetEcStatus(AudioProcessing* apm, bool enable, EcModes mode) {
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RTC_DCHECK(apm);
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RTC_DCHECK(mode == kEcConference || mode == kEcAecm) << "mode: " << mode;
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AudioProcessing::Config apm_config = apm->GetConfig();
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apm_config.echo_canceller.enabled = enable;
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apm_config.echo_canceller.mobile_mode = (mode == kEcAecm);
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apm_config.echo_canceller.legacy_moderate_suppression_level = false;
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apm->ApplyConfig(apm_config);
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RTC_LOG(LS_INFO) << "Echo control set to " << enable << " with mode " << mode;
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}
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void SetNsStatus(AudioProcessing* apm, bool enable) {
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RTC_DCHECK(apm);
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NoiseSuppression* ns = apm->noise_suppression();
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if (ns->set_level(NoiseSuppression::kHigh) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to set high NS level.";
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return;
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}
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if (ns->Enable(enable) != 0) {
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RTC_LOG(LS_ERROR) << "Failed to enable/disable NS: " << enable;
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return;
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}
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RTC_LOG(LS_INFO) << "NS set to " << enable;
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}
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} // namespace apm_helpers
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} // namespace webrtc
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