Merge pull request #1837 from secondlife/roxie/webrtc-voice-gain

5 files changed, 22 insertions, 40 deletions

diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index 14fb63ec2a..d5bd913315 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -35,6 +35,7 @@
 #include "api/media_stream_interface.h"
 #include "api/media_stream_track.h"
 #include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
 
 namespace llwebrtc
 {
@@ -88,7 +89,7 @@ void LLAudioDeviceObserver::OnRenderData(const void    *audio_samples,
 {
 }
 
-LLCustomProcessor::LLCustomProcessor() : mSampleRateHz(0), mNumChannels(0), mMicrophoneEnergy(0.0)
+LLCustomProcessor::LLCustomProcessor() : mSampleRateHz(0), mNumChannels(0), mMicrophoneEnergy(0.0), mGain(1.0)
 {
     memset(mSumVector, 0, sizeof(mSumVector));
 }
@@ -128,9 +129,13 @@ void LLCustomProcessor::Process(webrtc::AudioBuffer *audio_in)
     for (size_t index = 0; index < stream_config.num_samples(); index++)
     {
         float sample = frame_samples[index];
+        sample       = sample * mGain; // apply gain
+        frame_samples[index] = sample; // write processed sample back to buffer.
         energy += sample * sample;
     }
 
+    audio_in->CopyFrom(&frame[0], stream_config);
+
     // smooth it.
     size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
     float  totalSum    = 0;
@@ -236,9 +241,9 @@ void LLWebRTCImpl::init()
     webrtc::AudioProcessing::Config apm_config;
     apm_config.echo_canceller.enabled         = false;
     apm_config.echo_canceller.mobile_mode     = false;
-    apm_config.gain_controller1.enabled       = true;
+    apm_config.gain_controller1.enabled       = false;
     apm_config.gain_controller1.mode          = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-    apm_config.gain_controller2.enabled       = true;
+    apm_config.gain_controller2.enabled       = false;
     apm_config.high_pass_filter.enabled       = true;
     apm_config.noise_suppression.enabled      = true;
     apm_config.noise_suppression.level        = webrtc::AudioProcessing::Config::NoiseSuppression::kVeryHigh;
@@ -260,6 +265,7 @@ void LLWebRTCImpl::init()
     mAudioProcessingModule->ApplyConfig(apm_config);
     mAudioProcessingModule->Initialize(processing_config);
 
+
     mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(),
                                                                  mWorkerThread.get(),
                                                                  mSignalingThread.get(),
@@ -336,9 +342,9 @@ 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.enabled       = config.mAGC;
     apm_config.gain_controller1.mode          = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
-    apm_config.gain_controller2.enabled       = true;
+    apm_config.gain_controller2.enabled       = false;
     apm_config.high_pass_filter.enabled       = true;
     apm_config.transient_suppression.enabled  = true;
     apm_config.pipeline.multi_channel_render  = true;
@@ -612,6 +618,8 @@ float LLWebRTCImpl::getTuningAudioLevel() { return -20 * log10f(mTuningAudioDevi
 
 float LLWebRTCImpl::getPeerConnectionAudioLevel() { return -20 * log10f(mPeerCustomProcessor->getMicrophoneEnergy()); }
 
+void LLWebRTCImpl::setPeerConnectionGain(float gain) { mPeerCustomProcessor->setGain(gain); }
+
 
 //
 // Peer Connection Helpers
@@ -937,7 +945,7 @@ void LLWebRTCPeerConnectionImpl::setSendVolume(float volume)
             {
                 for (auto &track : mLocalStream->GetAudioTracks())
                 {
-                    track->GetSource()->SetVolume(volume);
+                    track->GetSource()->SetVolume(volume*5.0);
                 }
             }
         });
diff --git a/indra/llwebrtc/llwebrtc.h b/indra/llwebrtc/llwebrtc.h
index ecbab81538..f447ea990a 100644
--- a/indra/llwebrtc/llwebrtc.h
+++ b/indra/llwebrtc/llwebrtc.h
@@ -145,6 +145,7 @@ class LLWebRTCDeviceInterface
     virtual void setTuningMode(bool enable) = 0;
     virtual float getTuningAudioLevel() = 0; // for use during tuning
     virtual float getPeerConnectionAudioLevel() = 0; // for use when not tuning
+    virtual void setPeerConnectionGain(float gain) = 0;
 };
 
 // LLWebRTCAudioInterface provides the viewer with a way
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
index 448d36e3ea..6672f8ce90 100644
--- a/indra/llwebrtc/llwebrtc_impl.h
+++ b/indra/llwebrtc/llwebrtc_impl.h
@@ -121,6 +121,8 @@ class LLCustomProcessor : public webrtc::CustomProcessing
 
     float getMicrophoneEnergy() { return mMicrophoneEnergy; }
 
+    void setGain(float gain) { mGain = gain; }
+
   protected:
     static const int NUM_PACKETS_TO_FILTER = 30;  // 300 ms of smoothing
     int              mSampleRateHz;
@@ -128,6 +130,7 @@ class LLCustomProcessor : public webrtc::CustomProcessing
 
     float mSumVector[NUM_PACKETS_TO_FILTER];
     float mMicrophoneEnergy;
+    float mGain;
 };
 
 
@@ -160,6 +163,8 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
     float getTuningAudioLevel() override;
     float getPeerConnectionAudioLevel() override;
 
+    void setPeerConnectionGain(float gain) override;
+
     //
     // AudioDeviceSink
     //
diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp
index ddad470149..36e998af89 100644
--- a/indra/newview/llvoicewebrtc.cpp
+++ b/indra/newview/llvoicewebrtc.cpp
@@ -87,7 +87,7 @@ namespace {
     const F32 VOLUME_SCALE_WEBRTC = 0.01f;
     const F32 LEVEL_SCALE_WEBRTC  = 0.008f;
 
-    const F32 SPEAKING_AUDIO_LEVEL = 0.35;
+    const F32 SPEAKING_AUDIO_LEVEL = 0.30;
 
     static const std::string REPORTED_VOICE_SERVER_TYPE = "Secondlife WebRTC Gateway";
 
@@ -1484,14 +1484,10 @@ void LLWebRTCVoiceClient::setMicGain(F32 gain)
     if (gain != mMicGain)
     {
         mMicGain = gain;
-        sessionState::for_each(boost::bind(predSetMicGain, _1, gain));
+        mWebRTCDeviceInterface->setPeerConnectionGain(gain);
     }
 }
 
-void LLWebRTCVoiceClient::predSetMicGain(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 gain)
-{
-    session->setMicGain(gain);
-}
 
 void LLWebRTCVoiceClient::setVoiceEnabled(bool enabled)
 {
@@ -1690,7 +1686,6 @@ std::map<std::string, LLWebRTCVoiceClient::sessionState::ptr_t> LLWebRTCVoiceCli
 LLWebRTCVoiceClient::sessionState::sessionState() :
     mHangupOnLastLeave(false),
     mNotifyOnFirstJoin(false),
-    mMicGain(1.0),
     mMuted(false),
     mSpeakerVolume(1.0),
     mShuttingDown(false)
@@ -1735,15 +1730,6 @@ void LLWebRTCVoiceClient::sessionState::setMuteMic(bool muted)
     }
 }
 
-void LLWebRTCVoiceClient::sessionState::setMicGain(F32 gain)
-{
-    mMicGain = gain;
-    for (auto &connection : mWebRTCConnections)
-    {
-        connection->setMicGain(gain);
-    }
-}
-
 void LLWebRTCVoiceClient::sessionState::setSpeakerVolume(F32 volume)
 {
     mSpeakerVolume = volume;
@@ -1848,7 +1834,6 @@ LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::addSession(const std
 
         LL_DEBUGS("Voice") << "adding new session with channel: " << channel_id << LL_ENDL;
         session->setMuteMic(mMuteMic);
-        session->setMicGain(mMicGain);
         session->setSpeakerVolume(mSpeakerVolume);
 
         sessionState::addSession(channel_id, session);
@@ -1974,7 +1959,6 @@ bool LLWebRTCVoiceClient::estateSessionState::processConnectionStates()
             connectionPtr_t connection(new LLVoiceWebRTCSpatialConnection(neighbor, INVALID_PARCEL_ID, mChannelID));
 
             mWebRTCConnections.push_back(connection);
-            connection->setMicGain(mMicGain);
             connection->setMuteMic(mMuted);
             connection->setSpeakerVolume(mSpeakerVolume);
         }
@@ -2104,7 +2088,6 @@ LLVoiceWebRTCConnection::LLVoiceWebRTCConnection(const LLUUID &regionID, const s
     mShutDown(false),
     mIceCompleted(false),
     mSpeakerVolume(0.0),
-    mMicGain(0.0),
     mOutstandingRequests(0),
     mChannelID(channelID),
     mRegionID(regionID),
@@ -2367,15 +2350,6 @@ void LLVoiceWebRTCConnection::setMuteMic(bool muted)
     }
 }
 
-void LLVoiceWebRTCConnection::setMicGain(F32 gain)
-{
-    mMicGain = gain;
-    if (mWebRTCAudioInterface)
-    {
-        mWebRTCAudioInterface->setSendVolume(gain);
-    }
-}
-
 void LLVoiceWebRTCConnection::setSpeakerVolume(F32 volume)
 {
     mSpeakerVolume = volume;
@@ -2683,7 +2657,6 @@ bool LLVoiceWebRTCConnection::connectionStateMachine()
             // this connection.
             mWebRTCAudioInterface->setMute(mMuted);
             mWebRTCAudioInterface->setReceiveVolume(mSpeakerVolume);
-            mWebRTCAudioInterface->setSendVolume(mMicGain);
             LLWebRTCVoiceClient::getInstance()->OnConnectionEstablished(mChannelID, mRegionID);
             setVoiceConnectionState(VOICE_STATE_WAIT_FOR_DATA_CHANNEL);
             break;
diff --git a/indra/newview/llvoicewebrtc.h b/indra/newview/llvoicewebrtc.h
index 60724a4a2a..7042bbae00 100644
--- a/indra/newview/llvoicewebrtc.h
+++ b/indra/newview/llvoicewebrtc.h
@@ -282,7 +282,6 @@ public:
         virtual void sendData(const std::string &data);
 
         void setMuteMic(bool muted);
-        void setMicGain(F32 volume);
         void setSpeakerVolume(F32 volume);
         void setUserVolume(const LLUUID& id, F32 volume);
 
@@ -303,7 +302,6 @@ public:
         std::string mName;
 
         bool    mMuted;          // this session is muted.
-        F32     mMicGain;        // gain for this session.
         F32     mSpeakerVolume;  // volume for this session.
 
         bool        mShuttingDown;
@@ -382,7 +380,6 @@ public:
     static void predSendData(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const std::string& spatial_data);
     static void predUpdateOwnVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 audio_level);
     static void predSetMuteMic(const LLWebRTCVoiceClient::sessionStatePtr_t &session, bool mute);
-    static void predSetMicGain(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 volume);
     static void predSetSpeakerVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 volume);
     static void predShutdownSession(const LLWebRTCVoiceClient::sessionStatePtr_t &session);
     static void predSetUserMute(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const LLUUID& id, bool mute);
@@ -607,7 +604,6 @@ class LLVoiceWebRTCConnection :
     void processIceUpdatesCoro();
 
     virtual void setMuteMic(bool muted);
-    virtual void setMicGain(F32 volume);
     virtual void setSpeakerVolume(F32 volume);
 
     void setUserVolume(const LLUUID& id, F32 volume);
@@ -686,7 +682,6 @@ class LLVoiceWebRTCConnection :
     std::string mRemoteChannelSDP;
 
     bool mMuted;
-    F32  mMicGain;
     F32  mSpeakerVolume;
 
     bool mShutDown;