Microphone was being prematurely enabled on login for a short period.

The microphone issue was causing a short moment of sound, and was
causing bluetooth headsets to switch to hands-free/one channel mode
which is disruptive.

Also, update webrtc to deal with issue where airpods were garbled
after coming out of hands-free mode.

2 files changed, 62 insertions, 60 deletions

diff --git a/autobuild.xml b/autobuild.xml
index c2d063763d..3d34c78c01 100644
--- a/autobuild.xml
+++ b/autobuild.xml
@@ -745,18 +745,6 @@
       </map>
       <key>glm</key>
       <map>
-        <key>canonical_repo</key>
-        <string>https://github.com/secondlife/3p-glm</string>
-        <key>copyright</key>
-        <string>Copyright (c) 2005 - G-Truc Creation</string>
-        <key>description</key>
-        <string>OpenGL Mathematics</string>
-        <key>license</key>
-        <string>MIT</string>
-        <key>license_file</key>
-        <string>LICENSES/glm_license.txt</string>
-        <key>name</key>
-        <string>glm</string>
         <key>platforms</key>
         <map>
           <key>common</key>
@@ -774,16 +762,28 @@
             <string>common</string>
           </map>
         </map>
-        <key>source_type</key>
-        <string>git</string>
+        <key>license</key>
+        <string>MIT</string>
+        <key>license_file</key>
+        <string>LICENSES/glm_license.txt</string>
+        <key>copyright</key>
+        <string>Copyright (c) 2005 - G-Truc Creation</string>
+        <key>version</key>
+        <string>v1.0.1</string>
+        <key>name</key>
+        <string>glm</string>
         <key>vcs_branch</key>
         <string>refs/tags/v1.0.1-r1</string>
         <key>vcs_revision</key>
         <string>399cd5ba57a9267a560ce07e50a0f8c5fe3dc66f</string>
         <key>vcs_url</key>
         <string>git://github.com/secondlife/3p-glm.git</string>
-        <key>version</key>
-        <string>v1.0.1</string>
+        <key>canonical_repo</key>
+        <string>https://github.com/secondlife/3p-glm</string>
+        <key>description</key>
+        <string>OpenGL Mathematics</string>
+        <key>source_type</key>
+        <string>git</string>
       </map>
       <key>gstreamer</key>
       <map>
@@ -1418,14 +1418,6 @@
       </map>
       <key>llphysicsextensions_source</key>
       <map>
-        <key>copyright</key>
-        <string>Copyright (c) 2010, Linden Research, Inc.</string>
-        <key>license</key>
-        <string>internal</string>
-        <key>license_file</key>
-        <string>LICENSES/llphysicsextensions.txt</string>
-        <key>name</key>
-        <string>llphysicsextensions_source</string>
         <key>platforms</key>
         <map>
           <key>darwin64</key>
@@ -1477,8 +1469,16 @@
             <string>windows64</string>
           </map>
         </map>
+        <key>license</key>
+        <string>internal</string>
+        <key>license_file</key>
+        <string>LICENSES/llphysicsextensions.txt</string>
+        <key>copyright</key>
+        <string>Copyright (c) 2010, Linden Research, Inc.</string>
         <key>version</key>
         <string>1.0.b8b1f73</string>
+        <key>name</key>
+        <string>llphysicsextensions_source</string>
       </map>
       <key>llphysicsextensions_stub</key>
       <map>
@@ -2008,16 +2008,6 @@ Copyright (c) 2012, 2014, 2015, 2016 nghttp2 contributors</string>
       </map>
       <key>openal</key>
       <map>
-        <key>copyright</key>
-        <string>Copyright (C) 1999-2007 by authors.</string>
-        <key>description</key>
-        <string>OpenAL Soft is a software implementation of the OpenAL 3D audio API.</string>
-        <key>license</key>
-        <string>LGPL2</string>
-        <key>license_file</key>
-        <string>LICENSES/openal-soft.txt</string>
-        <key>name</key>
-        <string>openal</string>
         <key>platforms</key>
         <map>
           <key>darwin64</key>
@@ -2063,8 +2053,18 @@ Copyright (c) 2012, 2014, 2015, 2016 nghttp2 contributors</string>
             <string>windows64</string>
           </map>
         </map>
+        <key>license</key>
+        <string>LGPL2</string>
+        <key>license_file</key>
+        <string>LICENSES/openal-soft.txt</string>
+        <key>copyright</key>
+        <string>Copyright (C) 1999-2007 by authors.</string>
         <key>version</key>
         <string>1.23.1</string>
+        <key>name</key>
+        <string>openal</string>
+        <key>description</key>
+        <string>OpenAL Soft is a software implementation of the OpenAL 3D audio API.</string>
       </map>
       <key>openjpeg</key>
       <map>
@@ -2793,11 +2793,11 @@ Copyright (c) 2012, 2014, 2015, 2016 nghttp2 contributors</string>
             <key>archive</key>
             <map>
               <key>hash</key>
-              <string>194b4f5957c9f003c46e61a434e23a7c3d1180d6</string>
+              <string>8987b409ab7254ed22dab6dfbaefa059e7ab000e</string>
               <key>hash_algorithm</key>
               <string>sha1</string>
               <key>url</key>
-              <string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.70-debug/webrtc-m114.5735.08.70-debug.10377605436-darwin64-10377605436.tar.zst</string>
+              <string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.71-debug/webrtc-m114.5735.08.71-debug.10436964656-darwin64-10436964656.tar.zst</string>
             </map>
             <key>name</key>
             <string>darwin64</string>
@@ -2807,11 +2807,11 @@ Copyright (c) 2012, 2014, 2015, 2016 nghttp2 contributors</string>
             <key>archive</key>
             <map>
               <key>hash</key>
-              <string>38e0c7d30b4c40eb04e60ab199440b847cc7c6cf</string>
+              <string>153a220a138f9abe0e35fd63ea7f114b1117b4d6</string>
               <key>hash_algorithm</key>
               <string>sha1</string>
               <key>url</key>
-              <string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.70-debug/webrtc-m114.5735.08.70-debug.10377605436-linux64-10377605436.tar.zst</string>
+              <string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.71-debug/webrtc-m114.5735.08.71-debug.10436964656-linux64-10436964656.tar.zst</string>
             </map>
             <key>name</key>
             <string>linux64</string>
@@ -2821,11 +2821,11 @@ Copyright (c) 2012, 2014, 2015, 2016 nghttp2 contributors</string>
             <key>archive</key>
             <map>
               <key>hash</key>
-              <string>053fb5c873df9192e34cddcf2db1c5fdcff76ba1</string>
+              <string>709f23421d9de07fddd37ac4fd2cc4bb723bb4d7</string>
               <key>hash_algorithm</key>
               <string>sha1</string>
               <key>url</key>
-              <string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.70-debug/webrtc-m114.5735.08.70-debug.10377605436-windows64-10377605436.tar.zst</string>
+              <string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.71-debug/webrtc-m114.5735.08.71-debug.10436964656-windows64-10436964656.tar.zst</string>
             </map>
             <key>name</key>
             <string>windows64</string>
@@ -2838,7 +2838,7 @@ Copyright (c) 2012, 2014, 2015, 2016 nghttp2 contributors</string>
         <key>copyright</key>
         <string>Copyright (c) 2011, The WebRTC project authors. All rights reserved.</string>
         <key>version</key>
-        <string>m114.5735.08.70-debug.10377605436</string>
+        <string>m114.5735.08.71-debug.10436964656</string>
         <key>name</key>
         <string>webrtc</string>
         <key>vcs_branch</key>
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index d154bfb8eb..dd7883f973 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -206,10 +206,10 @@ void LLWebRTCImpl::init()
             mTuningDeviceModule->SetAudioDeviceSink(this);
             mTuningDeviceModule->InitMicrophone();
             mTuningDeviceModule->InitSpeaker();
+            mTuningDeviceModule->SetStereoRecording(false);
+            mTuningDeviceModule->SetStereoPlayout(true);
             mTuningDeviceModule->InitRecording();
             mTuningDeviceModule->InitPlayout();
-            mTuningDeviceModule->SetStereoRecording(true);
-            mTuningDeviceModule->SetStereoPlayout(true);
             updateDevices();
         });
 
@@ -227,10 +227,6 @@ void LLWebRTCImpl::init()
             mPeerDeviceModule->EnableBuiltInAEC(false);
             mPeerDeviceModule->InitMicrophone();
             mPeerDeviceModule->InitSpeaker();
-            mPeerDeviceModule->InitRecording();
-            mPeerDeviceModule->InitPlayout();
-            mPeerDeviceModule->SetStereoRecording(true);
-            mPeerDeviceModule->SetStereoPlayout(true);
         });
 
     // The custom processor allows us to retrieve audio data (and levels)
@@ -253,6 +249,8 @@ void LLWebRTCImpl::init()
     apm_config.pipeline.multi_channel_render  = true;
     apm_config.pipeline.multi_channel_capture = false;
 
+    mAudioProcessingModule->ApplyConfig(apm_config);
+
     webrtc::ProcessingConfig processing_config;
     processing_config.input_stream().set_num_channels(2);
     processing_config.input_stream().set_sample_rate_hz(48000);
@@ -263,10 +261,8 @@ void LLWebRTCImpl::init()
     processing_config.reverse_output_stream().set_num_channels(2);
     processing_config.reverse_output_stream().set_sample_rate_hz(48000);
 
-    mAudioProcessingModule->ApplyConfig(apm_config);
     mAudioProcessingModule->Initialize(processing_config);
 
-
     mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(),
                                                                  mWorkerThread.get(),
                                                                  mSignalingThread.get(),
@@ -329,6 +325,8 @@ void LLWebRTCImpl::setRecording(bool recording)
         {
             if (recording)
             {
+                mPeerDeviceModule->SetStereoRecording(false);
+                mPeerDeviceModule->InitRecording();
                 mPeerDeviceModule->StartRecording();
             }
             else
@@ -345,6 +343,8 @@ void LLWebRTCImpl::setPlayout(bool playing)
         {
             if (playing)
             {
+                mPeerDeviceModule->SetStereoPlayout(true);
+                mPeerDeviceModule->InitPlayout();
                 mPeerDeviceModule->StartPlayout();
             }
             else
@@ -430,9 +430,9 @@ void ll_set_device_module_capture_device(rtc::scoped_refptr<webrtc::AudioDeviceM
     // has it at 0
     device_module->SetRecordingDevice(device + 1);
 #endif
+    device_module->SetStereoRecording(false);
     device_module->InitMicrophone();
     device_module->InitRecording();
-    device_module->SetStereoRecording(false);
 }
 
 void LLWebRTCImpl::setCaptureDevice(const std::string &id)
@@ -494,9 +494,9 @@ void ll_set_device_module_render_device(rtc::scoped_refptr<webrtc::AudioDeviceMo
 #else
     device_module->SetPlayoutDevice(device + 1);
 #endif
+    device_module->SetStereoPlayout(true);
     device_module->InitSpeaker();
     device_module->InitPlayout();
-    device_module->SetStereoPlayout(true);
 }
 
 void LLWebRTCImpl::setRenderDevice(const std::string &id)
@@ -626,6 +626,8 @@ void LLWebRTCImpl::setTuningMode(bool enable)
                 //mTuningDeviceModule->StopPlayout();
                 ll_set_device_module_render_device(mPeerDeviceModule, mPlayoutDevice);
                 ll_set_device_module_capture_device(mPeerDeviceModule, mRecordingDevice);
+                mPeerDeviceModule->SetStereoPlayout(true);
+                mPeerDeviceModule->SetStereoRecording(false);
                 mPeerDeviceModule->InitPlayout();
                 mPeerDeviceModule->InitRecording();
                 mPeerDeviceModule->StartPlayout();
@@ -667,13 +669,13 @@ LLWebRTCPeerConnectionInterface *LLWebRTCImpl::newPeerConnection()
     rtc::scoped_refptr<LLWebRTCPeerConnectionImpl> peerConnection = rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>(new rtc::RefCountedObject<LLWebRTCPeerConnectionImpl>());
     peerConnection->init(this);
 
+    mPeerConnections.emplace_back(peerConnection);
+    peerConnection->enableSenderTracks(!mMute);
     if (mPeerConnections.empty())
     {
         setRecording(true);
         setPlayout(true);
     }
-    mPeerConnections.emplace_back(peerConnection);
-    peerConnection->enableSenderTracks(!mMute);
     return peerConnection.get();
 }
 
@@ -702,7 +704,7 @@ void LLWebRTCImpl::freePeerConnection(LLWebRTCPeerConnectionInterface* peer_conn
 LLWebRTCPeerConnectionImpl::LLWebRTCPeerConnectionImpl() :
     mWebRTCImpl(nullptr),
     mPeerConnection(nullptr),
-    mMute(false),
+    mMute(true),
     mAnswerReceived(false)
 {
 }
@@ -724,8 +726,8 @@ void LLWebRTCPeerConnectionImpl::init(LLWebRTCImpl * webrtc_impl)
 }
 void LLWebRTCPeerConnectionImpl::terminate()
 {
-    mWebRTCImpl->PostSignalingTask(
-        [=]()
+    mWebRTCImpl->SignalingBlockingCall(
+        [this]()
         {
             if (mPeerConnection)
             {
@@ -847,7 +849,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection(const LLWebRTCPeerConnecti
                 codecparam.clock_rate                 = 48000;
                 codecparam.num_channels               = 2;
                 codecparam.parameters["stereo"]       = "1";
-                codecparam.parameters["sprop-stereo"] = "1";
+                codecparam.parameters["sprop-stereo"] = "0";
                 params.codecs.push_back(codecparam);
                 sender->SetParameters(params);
             }
@@ -862,7 +864,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection(const LLWebRTCPeerConnecti
                 codecparam.clock_rate                 = 48000;
                 codecparam.num_channels               = 2;
                 codecparam.parameters["stereo"]       = "1";
-                codecparam.parameters["sprop-stereo"] = "1";
+                codecparam.parameters["sprop-stereo"] = "0";
                 params.codecs.push_back(codecparam);
                 receiver->SetParameters(params);
             }
@@ -1009,7 +1011,7 @@ void LLWebRTCPeerConnectionImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiv
     codecparam.clock_rate                 = 48000;
     codecparam.num_channels               = 2;
     codecparam.parameters["stereo"]       = "1";
-    codecparam.parameters["sprop-stereo"] = "1";
+    codecparam.parameters["sprop-stereo"] = "0";
     params.codecs.push_back(codecparam);
     receiver->SetParameters(params);
 }
@@ -1200,7 +1202,7 @@ void LLWebRTCPeerConnectionImpl::OnSuccess(webrtc::SessionDescriptionInterface *
         else if (sdp_line.find("a=fmtp:" + opus_payload) == 0)
         {
             sdp_mangled_stream << sdp_line << "a=fmtp:" << opus_payload
-            << " minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1;maxplaybackrate=48000;sprop-maxplaybackrate=48000;sprop-maxcapturerate=48000\n";
+            << " minptime=10;useinbandfec=1;stereo=1;sprop-stereo=0;maxplaybackrate=48000;sprop-maxplaybackrate=48000;sprop-maxcapturerate=48000\n";
         }
         else
         {