diff options
Diffstat (limited to 'indra/llwebrtc')
| -rw-r--r-- | indra/llwebrtc/llwebrtc.cpp | 10 |
1 files changed, 5 insertions, 5 deletions
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 { |