diff options
Diffstat (limited to 'indra/llwebrtc/llwebrtc.cpp')
| -rw-r--r-- | indra/llwebrtc/llwebrtc.cpp | 20 |
1 files changed, 14 insertions, 6 deletions
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp index 14fb63ec2a..d5bd913315 100644 --- a/indra/llwebrtc/llwebrtc.cpp +++ b/indra/llwebrtc/llwebrtc.cpp @@ -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); } } }); |