[WebRTC] control microphone gain via custom audio processor.

Previously, there were two places audio gain could be controlled:
- the device manager
- the audio track

The device manager audio gain control sets the system gain for all applications,
not just the webrtc application.

The audio track gain happens well after the audio processing where we want it to happen.

So, gain control was added to the existing custom audio processor, which previously only
handled calculating and retrieving the audio levels.

After these changes, the microphone gain slider does impact the audio volume heard by peers.

1 files changed, 14 insertions, 6 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);
                 }
             }
         });