From 23b065b0f1f8620e91a41f5538dfc1ff3077037e Mon Sep 17 00:00:00 2001
From: Roxie Linden <roxie@lindenlab.com>
Date: Thu, 4 Jan 2024 13:02:55 -0800
Subject: Enable AEC
---
indra/llwebrtc/llwebrtc.cpp | 10 +++++-----
1 file changed, 5 insertions(+), 5 deletions(-)
(limited to 'indra')
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index 9b3ec2889b..8e56f9c222 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -110,7 +110,7 @@ void LLWebRTCImpl::init()
mTaskQueueFactory.get(),
std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
mTuningDeviceModule->Init();
- mTuningDeviceModule->SetStereoRecording(false);
+ mTuningDeviceModule->SetStereoRecording(true);
mTuningDeviceModule->SetStereoPlayout(true);
mTuningDeviceModule->EnableBuiltInAEC(false);
updateDevices();
@@ -118,7 +118,7 @@ void LLWebRTCImpl::init()
rtc::scoped_refptr<webrtc::AudioProcessing> apm = webrtc::AudioProcessingBuilder().Create();
webrtc::AudioProcessing::Config apm_config;
- apm_config.echo_canceller.enabled = false;
+ apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = false;
apm_config.gain_controller1.enabled = true;
apm_config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
@@ -141,7 +141,7 @@ void LLWebRTCImpl::init()
mPeerDeviceModule->Init();
mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
- mPeerDeviceModule->SetStereoRecording(false);
+ mPeerDeviceModule->SetStereoRecording(true);
mPeerDeviceModule->SetStereoPlayout(true);
mPeerDeviceModule->EnableBuiltInAEC(false);
mPeerDeviceModule->InitMicrophone();
@@ -515,7 +515,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection()
cricket::AudioOptions audioOptions;
audioOptions.auto_gain_control = true;
- audioOptions.echo_cancellation = false; // incompatible with opus stereo
+ audioOptions.echo_cancellation = true; // incompatible with opus stereo
audioOptions.noise_suppression = true;
mLocalStream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");
@@ -878,7 +878,7 @@ 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\n";
+ << " minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1;maxplaybackrate=48000;sprop-maxcapturerate=48000;sprop-maxplaybackrate=48000\n";
}
else
{
--
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