Add UI for managing echo cancellation, AGC, and noise control.

Plumb audio settings through from webrtc to the sound preferences
UI (still needs some tweaking, of course.)

Also, choose stun servers based on grid.  Ultimately, the stun
stun servers will be passed up via login or something.

1 files changed, 71 insertions, 27 deletions

diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index 34d950b804..c51bcfcdd5 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -157,7 +157,6 @@ LLWebRTCImpl::LLWebRTCImpl() :
 
 void LLWebRTCImpl::init()
 {
-    RTC_DCHECK(mPeerConnectionFactory);
     mPlayoutDevice   = 0;
     mRecordingDevice = 0;
     rtc::InitializeSSL();
@@ -222,12 +221,10 @@ void LLWebRTCImpl::init()
     mPeerCustomProcessor = new LLCustomProcessor;
     webrtc::AudioProcessingBuilder apb;
     apb.SetCapturePostProcessing(std::unique_ptr<webrtc::CustomProcessing>(mPeerCustomProcessor));
-    rtc::scoped_refptr<webrtc::AudioProcessing> apm = apb.Create();
+    mAudioProcessingModule = apb.Create();
 
-    // TODO: wire some of these to the primary interface and ultimately
-    // to the UI to allow user config.
     webrtc::AudioProcessing::Config apm_config;
-    apm_config.echo_canceller.enabled         = true;
+    apm_config.echo_canceller.enabled         = false;
     apm_config.echo_canceller.mobile_mode     = false;
     apm_config.gain_controller1.enabled       = true;
     apm_config.gain_controller1.mode          = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
@@ -250,8 +247,8 @@ void LLWebRTCImpl::init()
     processing_config.reverse_output_stream().set_num_channels(2);
     processing_config.reverse_output_stream().set_sample_rate_hz(48000);
 
-    apm->Initialize(processing_config);
-    apm->ApplyConfig(apm_config);
+    mAudioProcessingModule->Initialize(processing_config);
+    mAudioProcessingModule->ApplyConfig(apm_config);
 
     mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(),
                                                                  mWorkerThread.get(),
@@ -262,7 +259,7 @@ void LLWebRTCImpl::init()
                                                                  nullptr /* video_encoder_factory */,
                                                                  nullptr /* video_decoder_factory */,
                                                                  nullptr /* audio_mixer */,
-                                                                 apm);
+                                                                 mAudioProcessingModule);
 
     mWorkerThread->BlockingCall([this]() { mPeerDeviceModule->StartPlayout(); });
 }
@@ -318,6 +315,49 @@ void LLWebRTCImpl::setRecording(bool recording)
         });
 }
 
+void LLWebRTCImpl::setAudioConfig(LLWebRTCDeviceInterface::AudioConfig config)
+{
+    webrtc::AudioProcessing::Config apm_config;
+    apm_config.echo_canceller.enabled         = config.mEchoCancellation;
+    apm_config.echo_canceller.mobile_mode     = false;
+    apm_config.gain_controller1.enabled       = true;
+    apm_config.gain_controller1.mode          = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+    apm_config.gain_controller2.enabled       = true;
+    apm_config.high_pass_filter.enabled       = true;
+    apm_config.transient_suppression.enabled  = true;
+    apm_config.pipeline.multi_channel_render  = true;
+    apm_config.pipeline.multi_channel_capture = true;
+    apm_config.pipeline.multi_channel_capture = true;
+
+    switch (config.mNoiseSuppressionLevel)
+    {
+        case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_NONE:
+            apm_config.noise_suppression.enabled = false;
+            apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kLow;
+            break;
+        case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_LOW:
+            apm_config.noise_suppression.enabled = true;
+            apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kLow;
+            break;
+        case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_MODERATE:
+            apm_config.noise_suppression.enabled = true;
+            apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kModerate;
+            break;
+        case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_HIGH:
+            apm_config.noise_suppression.enabled = true;
+            apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kHigh;
+            break;
+        case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_VERY_HIGH:
+            apm_config.noise_suppression.enabled = true;
+            apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kVeryHigh;
+            break;
+        default:
+            apm_config.noise_suppression.enabled = false;
+            apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kLow;
+    }
+    mAudioProcessingModule->ApplyConfig(apm_config);
+}
+
 void LLWebRTCImpl::refreshDevices()
 {
     mWorkerThread->PostTask([this]() { updateDevices(); });
@@ -616,32 +656,36 @@ void LLWebRTCPeerConnectionImpl::unsetSignalingObserver(LLWebRTCSignalingObserve
     }
 }
 
-// TODO: Add initialization structure through which
-// stun and turn servers may be passed in from
-// the sim or login.
-bool LLWebRTCPeerConnectionImpl::initializeConnection()
+bool LLWebRTCPeerConnectionImpl::initializeConnection(LLWebRTCPeerConnectionInterface::InitOptions options)
 {
     RTC_DCHECK(!mPeerConnection);
     mAnswerReceived = false;
 
     mWebRTCImpl->PostSignalingTask(
-        [this]()
+        [this, options]()
         {
+            std::vector<LLWebRTCPeerConnectionInterface::InitOptions::IceServers> servers = options.mServers;
+            if(servers.empty())
+            {
+                LLWebRTCPeerConnectionInterface::InitOptions::IceServers ice_servers;
+                ice_servers.mUrls.push_back("stun:stun.l.google.com:19302");
+                ice_servers.mUrls.push_back("stun1:stun.l.google.com:19302");
+                ice_servers.mUrls.push_back("stun2:stun.l.google.com:19302");
+                ice_servers.mUrls.push_back("stun3:stun.l.google.com:19302");
+                ice_servers.mUrls.push_back("stun4:stun.l.google.com:19302");
+            }
+
             webrtc::PeerConnectionInterface::RTCConfiguration config;
+            for (auto server : servers)
+            {
+                webrtc::PeerConnectionInterface::IceServer ice_server;
+                ice_server.urls = server.mUrls;
+                ice_server.username = server.mUserName;
+                ice_server.password = server.mPassword;
+                config.servers.push_back(ice_server);
+            }
+            
             config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-            webrtc::PeerConnectionInterface::IceServer server;
-            server.uri = "stun:roxie-turn.staging.secondlife.io:3478";
-            config.servers.push_back(server);
-            server.uri = "stun:stun.l.google.com:19302";
-            config.servers.push_back(server);
-            server.uri = "stun:stun1.l.google.com:19302";
-            config.servers.push_back(server);
-            server.uri = "stun:stun2.l.google.com:19302";
-            config.servers.push_back(server);
-            server.uri = "stun:stun3.l.google.com:19302";
-            config.servers.push_back(server);
-            server.uri = "stun:stun4.l.google.com:19302";
-            config.servers.push_back(server);
 
             config.set_min_port(60000);
             config.set_max_port(60100);
@@ -671,7 +715,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection()
 
             cricket::AudioOptions audioOptions;
             audioOptions.auto_gain_control = true;
-            audioOptions.echo_cancellation = true;  // incompatible with opus stereo
+            audioOptions.echo_cancellation = false;  // incompatible with opus stereo
             audioOptions.noise_suppression = true;
 
             mLocalStream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");