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Merge pull request #1837 from secondlife/roxie/webrtc-voice-gain

master
Roxanne Skelly 2024-06-25 09:37:22 -07:00 committed by GitHub
parent 35148b8a84 34a2fd525f
commit 370e6154c4
No known key found for this signature in database
GPG Key ID: B5690EEEBB952194
5 changed files with 22 additions and 40 deletions
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20
indra/llwebrtc/llwebrtc.cpp
View File

@ -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);
}
}
});

1
indra/llwebrtc/llwebrtc.h
View File

@ -145,6 +145,7 @@ class LLWebRTCDeviceInterface
virtual void setTuningMode(bool enable) = 0;
virtual float getTuningAudioLevel() = 0; // for use during tuning
virtual float getPeerConnectionAudioLevel() = 0; // for use when not tuning
virtual void setPeerConnectionGain(float gain) = 0;
};
// LLWebRTCAudioInterface provides the viewer with a way

5
indra/llwebrtc/llwebrtc_impl.h
View File

@ -121,6 +121,8 @@ class LLCustomProcessor : public webrtc::CustomProcessing
float getMicrophoneEnergy() { return mMicrophoneEnergy; }
void setGain(float gain) { mGain = gain; }
protected:
static const int NUM_PACKETS_TO_FILTER = 30; // 300 ms of smoothing
int mSampleRateHz;
@ -128,6 +130,7 @@ class LLCustomProcessor : public webrtc::CustomProcessing
float mSumVector[NUM_PACKETS_TO_FILTER];
float mMicrophoneEnergy;
float mGain;
};
@ -160,6 +163,8 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
float getTuningAudioLevel() override;
float getPeerConnectionAudioLevel() override;
void setPeerConnectionGain(float gain) override;
//
// AudioDeviceSink
//

31
indra/newview/llvoicewebrtc.cpp
View File

@ -87,7 +87,7 @@ namespace {
const F32 VOLUME_SCALE_WEBRTC = 0.01f;
const F32 LEVEL_SCALE_WEBRTC = 0.008f;
const F32 SPEAKING_AUDIO_LEVEL = 0.35;
const F32 SPEAKING_AUDIO_LEVEL = 0.30;
static const std::string REPORTED_VOICE_SERVER_TYPE = "Secondlife WebRTC Gateway";
@ -1484,14 +1484,10 @@ void LLWebRTCVoiceClient::setMicGain(F32 gain)
if (gain != mMicGain)
{
mMicGain = gain;
sessionState::for_each(boost::bind(predSetMicGain, _1, gain));
mWebRTCDeviceInterface->setPeerConnectionGain(gain);
}
}
void LLWebRTCVoiceClient::predSetMicGain(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 gain)
{
session->setMicGain(gain);
}
void LLWebRTCVoiceClient::setVoiceEnabled(bool enabled)
{
@ -1690,7 +1686,6 @@ std::map<std::string, LLWebRTCVoiceClient::sessionState::ptr_t> LLWebRTCVoiceCli
LLWebRTCVoiceClient::sessionState::sessionState() :
mHangupOnLastLeave(false),
mNotifyOnFirstJoin(false),
mMicGain(1.0),
mMuted(false),
mSpeakerVolume(1.0),
mShuttingDown(false)
@ -1735,15 +1730,6 @@ void LLWebRTCVoiceClient::sessionState::setMuteMic(bool muted)
}
}
void LLWebRTCVoiceClient::sessionState::setMicGain(F32 gain)
{
mMicGain = gain;
for (auto &connection : mWebRTCConnections)
{
connection->setMicGain(gain);
}
}
void LLWebRTCVoiceClient::sessionState::setSpeakerVolume(F32 volume)
{
mSpeakerVolume = volume;
@ -1848,7 +1834,6 @@ LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::addSession(const std
LL_DEBUGS("Voice") << "adding new session with channel: " << channel_id << LL_ENDL;
session->setMuteMic(mMuteMic);
session->setMicGain(mMicGain);
session->setSpeakerVolume(mSpeakerVolume);
sessionState::addSession(channel_id, session);
@ -1974,7 +1959,6 @@ bool LLWebRTCVoiceClient::estateSessionState::processConnectionStates()
connectionPtr_t connection(new LLVoiceWebRTCSpatialConnection(neighbor, INVALID_PARCEL_ID, mChannelID));
mWebRTCConnections.push_back(connection);
connection->setMicGain(mMicGain);
connection->setMuteMic(mMuted);
connection->setSpeakerVolume(mSpeakerVolume);
}
@ -2104,7 +2088,6 @@ LLVoiceWebRTCConnection::LLVoiceWebRTCConnection(const LLUUID &regionID, const s
mShutDown(false),
mIceCompleted(false),
mSpeakerVolume(0.0),
mMicGain(0.0),
mOutstandingRequests(0),
mChannelID(channelID),
mRegionID(regionID),
@ -2367,15 +2350,6 @@ void LLVoiceWebRTCConnection::setMuteMic(bool muted)
}
}
void LLVoiceWebRTCConnection::setMicGain(F32 gain)
{
mMicGain = gain;
if (mWebRTCAudioInterface)
{
mWebRTCAudioInterface->setSendVolume(gain);
}
}
void LLVoiceWebRTCConnection::setSpeakerVolume(F32 volume)
{
mSpeakerVolume = volume;
@ -2683,7 +2657,6 @@ bool LLVoiceWebRTCConnection::connectionStateMachine()
// this connection.
mWebRTCAudioInterface->setMute(mMuted);
mWebRTCAudioInterface->setReceiveVolume(mSpeakerVolume);
mWebRTCAudioInterface->setSendVolume(mMicGain);
LLWebRTCVoiceClient::getInstance()->OnConnectionEstablished(mChannelID, mRegionID);
setVoiceConnectionState(VOICE_STATE_WAIT_FOR_DATA_CHANNEL);
break;

5
indra/newview/llvoicewebrtc.h
View File

@ -282,7 +282,6 @@ public:
virtual void sendData(const std::string &data);
void setMuteMic(bool muted);
void setMicGain(F32 volume);
void setSpeakerVolume(F32 volume);
void setUserVolume(const LLUUID& id, F32 volume);
@ -303,7 +302,6 @@ public:
std::string mName;
bool mMuted; // this session is muted.
F32 mMicGain; // gain for this session.
F32 mSpeakerVolume; // volume for this session.
bool mShuttingDown;
@ -382,7 +380,6 @@ public:
static void predSendData(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const std::string& spatial_data);
static void predUpdateOwnVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 audio_level);
static void predSetMuteMic(const LLWebRTCVoiceClient::sessionStatePtr_t &session, bool mute);
static void predSetMicGain(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 volume);
static void predSetSpeakerVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 volume);
static void predShutdownSession(const LLWebRTCVoiceClient::sessionStatePtr_t &session);
static void predSetUserMute(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const LLUUID& id, bool mute);
@ -607,7 +604,6 @@ class LLVoiceWebRTCConnection :
void processIceUpdatesCoro();
virtual void setMuteMic(bool muted);
virtual void setMicGain(F32 volume);
virtual void setSpeakerVolume(F32 volume);
void setUserVolume(const LLUUID& id, F32 volume);
@ -686,7 +682,6 @@ class LLVoiceWebRTCConnection :
std::string mRemoteChannelSDP;
bool mMuted;
F32 mMicGain;
F32 mSpeakerVolume;
bool mShutDown;