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Updated WebRTC APM

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I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
This commit is contained in:
Grishka
2018-11-21 21:22:31 +03:00
parent cc0cf35725
commit 5caaaafa42
644 changed files with 90797 additions and 9287 deletions
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18
VoIPController.cpp
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@@ -54,6 +54,7 @@ bool VoIPController::didInitWin32TimeScale = false;
#ifdef __ANDROID__
#include "os/android/JNIUtilities.h"
#include "os/android/AudioInputAndroid.h"
extern jclass jniUtilitiesClass;
#endif
@@ -1042,6 +1043,20 @@ void VoIPController::InitializeAudio(){
audioIO=audio::AudioIO::Create();
audioInput=audioIO->GetInput();
audioOutput=audioIO->GetOutput();
#ifdef __ANDROID__
audio::AudioInputAndroid* androidInput=dynamic_cast<audio::AudioInputAndroid*>(audioInput);
if(androidInput){
unsigned int effects=androidInput->GetEnabledEffects();
if(!(effects & audio::AudioInputAndroid::EFFECT_AEC)){
config.enableAEC=true;
LOGI("Forcing software AEC because built-in is not good");
}
if(!(effects & audio::AudioInputAndroid::EFFECT_NS)){
config.enableNS=true;
LOGI("Forcing software NS because built-in is not good");
}
}
#endif
LOGI("AEC: %d NS: %d AGC: %d", config.enableAEC, config.enableNS, config.enableAGC);
echoCanceller=new EchoCanceller(config.enableAEC, config.enableNS, config.enableAGC);
encoder=new OpusEncoder(audioInput, true);
@@ -1056,7 +1071,7 @@ void VoIPController::InitializeAudio(){
#endif
if(!audioOutput->IsInitialized()){
LOGE("Erorr initializing audio playback");
LOGE("Error initializing audio playback");
lastError=ERROR_AUDIO_IO;
SetState(STATE_FAILED);
@@ -1102,6 +1117,7 @@ void VoIPController::UpdateAudioOutputState(){
areAnyAudioStreamsEnabled=true;
}
if(audioOutput){
LOGV("New audio output state: %d", areAnyAudioStreamsEnabled);
if(audioOutput->IsPlaying()!=areAnyAudioStreamsEnabled){
if(areAnyAudioStreamsEnabled)
audioOutput->Start();