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.

3 files changed, 119 insertions, 33 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");
diff --git a/indra/llwebrtc/llwebrtc.h b/indra/llwebrtc/llwebrtc.h
index be2e5cdf68..f3a33435c9 100644
--- a/indra/llwebrtc/llwebrtc.h
+++ b/indra/llwebrtc/llwebrtc.h
@@ -99,10 +99,32 @@ class LLWebRTCDevicesObserver
 // to enumerate, set, and get notifications of changes
 // for both capture (microphone) and render (speaker)
 // devices.
+
 class LLWebRTCDeviceInterface
 {
   public:
-
+    struct AudioConfig {
+
+        bool mAGC { true };
+
+        bool mEchoCancellation { true };
+
+        // TODO: The various levels of noise suppression are configured
+        // on the APM which would require setting config on the APM.
+        // We should pipe the various values through
+        // later.
+        typedef enum {
+            NOISE_SUPPRESSION_LEVEL_NONE = 0,
+            NOISE_SUPPRESSION_LEVEL_LOW,
+            NOISE_SUPPRESSION_LEVEL_MODERATE,
+            NOISE_SUPPRESSION_LEVEL_HIGH,
+            NOISE_SUPPRESSION_LEVEL_VERY_HIGH
+        } ENoiseSuppressionLevel;
+        ENoiseSuppressionLevel mNoiseSuppressionLevel { NOISE_SUPPRESSION_LEVEL_VERY_HIGH };
+    };
+
+    virtual void setAudioConfig(AudioConfig config) = 0;
+    
     // instructs webrtc to refresh the device list.
     virtual void refreshDevices() = 0;
 
@@ -194,19 +216,35 @@ class LLWebRTCSignalingObserver
     virtual void OnDataChannelReady(LLWebRTCDataInterface *data_interface) = 0;
 };
 
-
 // LLWebRTCPeerConnectionInterface representsd a connection to a peer,
 // in most cases a Secondlife WebRTC server.  This interface
 // allows for management of this peer connection.
 class LLWebRTCPeerConnectionInterface
 {
   public:
+    
+    struct InitOptions
+    {
+        // equivalent of PeerConnectionInterface::IceServer
+        struct IceServers {
+            // Valid formats are described in RFC7064 and RFC7065.
+            // Urls should containe dns hostnames (not IP addresses)
+            // as the TLS certificate policy is 'secure.'
+            // and we do not currentply support TLS extensions.
+            std::vector<std::string> mUrls;
+            std::string mUserName;
+            std::string mPassword;
+        };
+
+        std::vector<IceServers> mServers;
+    };
+
+    virtual bool initializeConnection(InitOptions options = InitOptions()) = 0;
+    virtual bool shutdownConnection() = 0;
 
     virtual void setSignalingObserver(LLWebRTCSignalingObserver* observer) = 0;
     virtual void unsetSignalingObserver(LLWebRTCSignalingObserver* observer) = 0;
 
-    virtual bool initializeConnection() = 0;
-    virtual bool shutdownConnection() = 0;
     virtual void AnswerAvailable(const std::string &sdp) = 0;
 };
 
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
index 38810a29b5..32faf2516c 100644
--- a/indra/llwebrtc/llwebrtc_impl.h
+++ b/indra/llwebrtc/llwebrtc_impl.h
@@ -145,6 +145,8 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
     // LLWebRTCDeviceInterface
     //
 
+    void setAudioConfig(LLWebRTCDeviceInterface::AudioConfig config = LLWebRTCDeviceInterface::AudioConfig()) override;
+
     void refreshDevices() override;
 
     void setDevicesObserver(LLWebRTCDevicesObserver *observer) override;
@@ -227,6 +229,8 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
 
     // The factory that allows creation of native webrtc PeerConnections.
     rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> mPeerConnectionFactory;
+    
+    rtc::scoped_refptr<webrtc::AudioProcessing>                mAudioProcessingModule;
 
     // more native webrtc stuff
     std::unique_ptr<webrtc::TaskQueueFactory>                  mTaskQueueFactory;
@@ -278,11 +282,11 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnectionInterface,
     //
     // LLWebRTCPeerConnection
     //
+    bool initializeConnection(InitOptions options = InitOptions()) override;
+    bool shutdownConnection() override;
 
     void setSignalingObserver(LLWebRTCSignalingObserver *observer) override;
     void unsetSignalingObserver(LLWebRTCSignalingObserver *observer) override;
-    bool initializeConnection() override;
-    bool shutdownConnection() override;
     void AnswerAvailable(const std::string &sdp) override;
 
     //