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Merge branch 'roxie/webrtc-airpod-fixes' of github.com:secondlife/viewer into roxie/webrtc-airpod-fixes

meow-7.2.2
Roxie Linden 2024-08-18 23:10:00 -07:00
parent 226f7bc5b7 28fdd6e786
commit de0b5afa25
2 changed files with 24 additions and 22 deletions
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14
autobuild.xml
View File

@ -2793,11 +2793,11 @@ Copyright (c) 2012, 2014, 2015, 2016 nghttp2 contributors</string>
<key>archive</key>
<map>
<key>hash</key>
<string>f8a58d9b5d18810189c5b09ca5c5d7227346ac8d</string>
<string>cbf9fae6cdc6983456eec601a596ce27c961f05f</string>
<key>hash_algorithm</key>
<string>sha1</string>
<key>url</key>
<string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.72/webrtc-m114.5735.08.72.10447328796-darwin64-10447328796.tar.zst</string>
<string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.72-debug/webrtc-m114.5735.08.72-debug.10438479298-darwin64-10438479298.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>0037f70b29f6c85eb7ee2f030f466d774793bf41</string>
<string>8a874c6bd96cdcce8b6fe744f235dc60f1bb4279</string>
<key>hash_algorithm</key>
<string>sha1</string>
<key>url</key>
<string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.72/webrtc-m114.5735.08.72.10447328796-linux64-10447328796.tar.zst</string>
<string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.72-debug/webrtc-m114.5735.08.72-debug.10438479298-linux64-10438479298.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>744ca0f034f73a10fc40182f6c099a5952cb42a6</string>
<string>379e29faa67a826729179f7228146e2812a6088a</string>
<key>hash_algorithm</key>
<string>sha1</string>
<key>url</key>
<string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.72/webrtc-m114.5735.08.72.10447328796-windows64-10447328796.tar.zst</string>
<string>https://github.com/secondlife/3p-webrtc-build/releases/download/m114.5735.08.72-debug/webrtc-m114.5735.08.72-debug.10438479298-windows64-10438479298.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>m114.5735.08.72.10447328796</string>
<string>m114.5735.08.72-debug.10438479298</string>
<key>name</key>
<string>webrtc</string>
<key>vcs_branch</key>

32
indra/llwebrtc/llwebrtc.cpp
View File

@ -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)
{