From d302d89891475d1431099deac99bc52a2c3046a1 Mon Sep 17 00:00:00 2001
From: Roxie Linden <roxie@lindenlab.com>
Date: Thu, 30 Nov 2023 13:14:07 -0800
Subject: Refactor/clean-up WebRTC voice to handle multiple voice streams
This is useful for cross-region voice, quick voice switching, etc.
---
indra/llwebrtc/llwebrtc.cpp | 79 +++++++++++++++++++++++++++---------------
indra/llwebrtc/llwebrtc.h | 8 +----
indra/llwebrtc/llwebrtc_impl.h | 7 ++--
3 files changed, 54 insertions(+), 40 deletions(-)
(limited to 'indra/llwebrtc')
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index bf1e04c2f2..b2f5e0212e 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -216,6 +216,7 @@ void LLWebRTCImpl::setCaptureDevice(const std::string &id)
mWorkerThread->PostTask(
[this, id]()
{
+ int16_t tuningRecordingDevice = 0;
int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
for (int16_t i = 0; i < captureDeviceCount; i++)
{
@@ -225,18 +226,32 @@ void LLWebRTCImpl::setCaptureDevice(const std::string &id)
if (id == guid || id == "Default") // first one in list is default
{
RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
- mRecordingDevice = i;
+ tuningRecordingDevice = i;
break;
}
}
mTuningDeviceModule->StopRecording();
- mTuningDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mTuningDeviceModule->SetRecordingDevice(tuningRecordingDevice);
mTuningDeviceModule->InitMicrophone();
mTuningDeviceModule->InitRecording();
mTuningDeviceModule->StartRecording();
bool was_peer_recording = false;
if (mPeerDeviceModule)
{
+ int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
+ for (int16_t i = 0; i < captureDeviceCount; i++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mTuningDeviceModule->RecordingDeviceName(i, name, guid);
+ if (id == guid || id == "Default") // first one in list is default
+ {
+ RTC_LOG(LS_INFO)
+ << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
+ mRecordingDevice = i;
+ break;
+ }
+ }
was_peer_recording = mPeerDeviceModule->Recording();
if (was_peer_recording)
{
@@ -259,6 +274,7 @@ void LLWebRTCImpl::setRenderDevice(const std::string &id)
[this, id]()
{
int16_t renderDeviceCount = mTuningDeviceModule->PlayoutDevices();
+ int16_t tuningPlayoutDevice = 0;
for (int16_t i = 0; i < renderDeviceCount; i++)
{
char name[webrtc::kAdmMaxDeviceNameSize];
@@ -267,7 +283,7 @@ void LLWebRTCImpl::setRenderDevice(const std::string &id)
if (id == guid || id == "Default")
{
RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
- mPlayoutDevice = i;
+ tuningPlayoutDevice = i;
break;
}
}
@@ -277,30 +293,40 @@ void LLWebRTCImpl::setRenderDevice(const std::string &id)
{
mTuningDeviceModule->StopPlayout();
}
- bool was_peer_mute = false;
if (mPeerDeviceModule)
{
- mPeerDeviceModule->SpeakerMute(&was_peer_mute);
- if (!was_peer_mute)
- {
- mPeerDeviceModule->SetSpeakerMute(true);
- }
+ mPeerDeviceModule->SetSpeakerMute(true);
}
- mTuningDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mTuningDeviceModule->SetPlayoutDevice(tuningPlayoutDevice);
mTuningDeviceModule->InitSpeaker();
mTuningDeviceModule->InitPlayout();
if (was_tuning_playing)
{
mTuningDeviceModule->StartPlayout();
}
+ renderDeviceCount = mPeerDeviceModule->PlayoutDevices();
+
if (mPeerDeviceModule)
{
+ for (int16_t i = 0; i < renderDeviceCount; i++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mPeerDeviceModule->PlayoutDeviceName(i, name, guid);
+ if (id == guid || id == "Default")
+ {
+ RTC_LOG(LS_INFO)
+ << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
+ mPlayoutDevice = i;
+ break;
+ }
+ }
mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
mPeerDeviceModule->InitSpeaker();
mPeerDeviceModule->InitPlayout();
mPeerDeviceModule->StartPlayout();
- mPeerDeviceModule->SetSpeakerMute(was_peer_mute);
+ mPeerDeviceModule->SetSpeakerMute(false);
}
mTuningDeviceModule->SetSpeakerMute(false);
@@ -376,6 +402,8 @@ float LLWebRTCImpl::getPeerAudioLevel() { return 20 * mPeerAudioDeviceObserver->
void LLWebRTCImpl::setSpeakerVolume(float volume) { mPeerDeviceModule->SetSpeakerVolume( (uint32_t)(volume * VOLUME_SCALE_WEBRTC));}
void LLWebRTCImpl::setMicrophoneVolume(float volume) { mPeerDeviceModule->SetMicrophoneVolume((uint32_t)(volume * VOLUME_SCALE_WEBRTC));}
+void LLWebRTCImpl::setMute(bool mute) { mPeerDeviceModule->SetMicrophoneMute(mute); }
+
//
// Helpers
//
@@ -566,9 +594,12 @@ void LLWebRTCPeerConnectionImpl::AnswerAvailable(const std::string &sdp)
mWebRTCImpl->PostSignalingTask(
[this, sdp]()
{
- RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
- mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
- rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
+ if (mPeerConnection)
+ {
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
+ mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
+ rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
+ }
});
}
@@ -844,17 +875,6 @@ void LLWebRTCPeerConnectionImpl::OnSetLocalDescriptionComplete(webrtc::RTCError
}
-void LLWebRTCPeerConnectionImpl::setAudioObserver(LLWebRTCAudioObserver *observer) { mAudioObserverList.emplace_back(observer); }
-
-void LLWebRTCPeerConnectionImpl::unsetAudioObserver(LLWebRTCAudioObserver *observer)
-{
- std::vector<LLWebRTCAudioObserver *>::iterator it = std::find(mAudioObserverList.begin(), mAudioObserverList.end(), observer);
- if (it != mAudioObserverList.end())
- {
- mAudioObserverList.erase(it);
- }
-}
-
//
// DataChannelObserver implementation
//
@@ -897,9 +917,12 @@ void LLWebRTCPeerConnectionImpl::OnMessage(const webrtc::DataBuffer& buffer)
void LLWebRTCPeerConnectionImpl::sendData(const std::string& data, bool binary)
{
- rtc::CopyOnWriteBuffer cowBuffer(data.data(), data.length());
- webrtc::DataBuffer buffer(cowBuffer, binary);
- mDataChannel->Send(buffer);
+ if (mDataChannel)
+ {
+ rtc::CopyOnWriteBuffer cowBuffer(data.data(), data.length());
+ webrtc::DataBuffer buffer(cowBuffer, binary);
+ mDataChannel->Send(buffer);
+ }
}
void LLWebRTCPeerConnectionImpl::setDataObserver(LLWebRTCDataObserver* observer) { mDataObserverList.emplace_back(observer); }
diff --git a/indra/llwebrtc/llwebrtc.h b/indra/llwebrtc/llwebrtc.h
index 753fe6a983..ed80fa5648 100644
--- a/indra/llwebrtc/llwebrtc.h
+++ b/indra/llwebrtc/llwebrtc.h
@@ -88,19 +88,13 @@ class LLWebRTCDeviceInterface
virtual void setSpeakerVolume(float volume) = 0; // volume between 0.0 and 1.0
virtual void setMicrophoneVolume(float volume) = 0; // volume between 0.0 and 1.0
+ virtual void setMute(bool mute) = 0;
virtual float getPeerAudioLevel() = 0;
};
-class LLWebRTCAudioObserver
-{
- public:
-};
-
class LLWebRTCAudioInterface
{
public:
- virtual void setAudioObserver(LLWebRTCAudioObserver *observer) = 0;
- virtual void unsetAudioObserver(LLWebRTCAudioObserver *observer) = 0;
virtual void setMute(bool mute) = 0;
};
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
index eb439b5a46..76e29c63fb 100644
--- a/indra/llwebrtc/llwebrtc_impl.h
+++ b/indra/llwebrtc/llwebrtc_impl.h
@@ -118,7 +118,8 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface
float getPeerAudioLevel() override;
void setSpeakerVolume(float volume) override; // range 0.0-1.0
- void setMicrophoneVolume(float volume) override; // range 0.0-1.0
+ void setMicrophoneVolume(float volume) override; // range 0.0-1.0
+ void setMute(bool mute) override;
//
// Helpers
@@ -227,8 +228,6 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnection,
//
// LLWebRTCAudioInterface
//
- void setAudioObserver(LLWebRTCAudioObserver *observer) override;
- void unsetAudioObserver(LLWebRTCAudioObserver *observer) override;
void setMute(bool mute) override;
//
@@ -292,8 +291,6 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnection,
bool mAnswerReceived;
rtc::scoped_refptr<webrtc::PeerConnectionInterface> mPeerConnection;
-
- std::vector<LLWebRTCAudioObserver *> mAudioObserverList;
std::vector<LLWebRTCDataObserver *> mDataObserverList;
rtc::scoped_refptr<webrtc::DataChannelInterface> mDataChannel;
--
cgit v1.2.3