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authorRoxie Linden <roxie@lindenlab.com>2024-02-05 16:47:47 -0800
committerRoxie Linden <roxie@lindenlab.com>2024-02-08 18:37:09 -0800
commitdc1858a32a71e4e944009b206c32112ec846880b (patch)
tree196034409fbb238c768e57103b24884731e5a84c
parent65b5a6e12d32602ae5b0ed3982dc2c74057c7ab7 (diff)
put observer-based tuning audio level calculation back
-rw-r--r--indra/llwebrtc/llwebrtc.cpp53
-rw-r--r--indra/llwebrtc/llwebrtc_impl.h29
2 files changed, 74 insertions, 8 deletions
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index f0d7f6ad8f..3273786242 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -40,6 +40,46 @@
namespace llwebrtc
{
+LLAudioDeviceObserver::LLAudioDeviceObserver() : mMicrophoneEnergy(0.0), mSumVector {0} {}
+
+float LLAudioDeviceObserver::getMicrophoneEnergy() { return mMicrophoneEnergy; }
+
+void LLAudioDeviceObserver::OnCaptureData(const void *audio_samples,
+ const size_t num_samples,
+ const size_t bytes_per_sample,
+ const size_t num_channels,
+ const uint32_t samples_per_sec)
+{
+ float energy = 0;
+ const short *samples = (const short *) audio_samples;
+ for (size_t index = 0; index < num_samples * num_channels; index++)
+ {
+ float sample = (static_cast<float>(samples[index]) / (float) 32767);
+ energy += sample * sample;
+ }
+
+ // smooth it.
+ size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
+ float totalSum = 0;
+ int i;
+ for (i = 0; i < (buffer_size - 1); i++)
+ {
+ mSumVector[i] = mSumVector[i + 1];
+ totalSum += mSumVector[i];
+ }
+ mSumVector[i] = energy;
+ totalSum += energy;
+ mMicrophoneEnergy = std::sqrt(totalSum / (num_samples * buffer_size));
+}
+
+void LLAudioDeviceObserver::OnRenderData(const void *audio_samples,
+ const size_t num_samples,
+ const size_t bytes_per_sample,
+ const size_t num_channels,
+ const uint32_t samples_per_sec)
+{
+}
+
LLCustomProcessor::LLCustomProcessor() :
mSampleRateHz(0),
mNumChannels(0)
@@ -118,12 +158,13 @@ void LLWebRTCImpl::init()
mSignalingThread->SetName("WebRTCSignalingThread", nullptr);
mSignalingThread->Start();
+ mTuningAudioDeviceObserver = new LLAudioDeviceObserver;
mWorkerThread->PostTask(
[this]()
{
mTuningDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(
- webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
- mTaskQueueFactory.get(), nullptr);
+ webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio, mTaskQueueFactory.get(),
+ std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
mTuningDeviceModule->Init();
mTuningDeviceModule->SetStereoRecording(true);
mTuningDeviceModule->SetStereoPlayout(true);
@@ -164,9 +205,9 @@ void LLWebRTCImpl::init()
mPeerDeviceModule->InitPlayout();
});
- mCustomProcessor = new LLCustomProcessor;
+ mPeerCustomProcessor = new LLCustomProcessor;
webrtc::AudioProcessingBuilder apb;
- apb.SetCapturePostProcessing(std::unique_ptr<webrtc::CustomProcessing>(mCustomProcessor));
+ apb.SetCapturePostProcessing(std::unique_ptr<webrtc::CustomProcessing>(mPeerCustomProcessor));
rtc::scoped_refptr<webrtc::AudioProcessing> apm = apb.Create();
webrtc::AudioProcessing::Config apm_config;
@@ -471,9 +512,9 @@ void LLWebRTCImpl::setTuningMode(bool enable)
}
}
-float LLWebRTCImpl::getTuningAudioLevel() { return -20 * log10f(mCustomProcessor->getMicrophoneEnergy()); }
+float LLWebRTCImpl::getTuningAudioLevel() { return -20 * log10f(mTuningAudioDeviceObserver->getMicrophoneEnergy()); }
-float LLWebRTCImpl::getPeerAudioLevel() { return -20 * log10f(mCustomProcessor->getMicrophoneEnergy()); }
+float LLWebRTCImpl::getPeerAudioLevel() { return -20 * log10f(mPeerCustomProcessor->getMicrophoneEnergy()); }
//
// Helpers
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
index 419482b751..7146267379 100644
--- a/indra/llwebrtc/llwebrtc_impl.h
+++ b/indra/llwebrtc/llwebrtc_impl.h
@@ -65,6 +65,30 @@ namespace llwebrtc
class LLWebRTCPeerConnectionImpl;
+class LLAudioDeviceObserver : public webrtc::AudioDeviceDataObserver
+{
+ public:
+ LLAudioDeviceObserver();
+
+ float getMicrophoneEnergy();
+
+ void OnCaptureData(const void *audio_samples,
+ const size_t num_samples,
+ const size_t bytes_per_sample,
+ const size_t num_channels,
+ const uint32_t samples_per_sec) override;
+
+ void OnRenderData(const void *audio_samples,
+ const size_t num_samples,
+ const size_t bytes_per_sample,
+ const size_t num_channels,
+ const uint32_t samples_per_sec) override;
+
+ protected:
+ float mSumVector[30]; // 300 ms of smoothing
+ float mMicrophoneEnergy;
+};
+
class LLCustomProcessor : public webrtc::CustomProcessing
{
public:
@@ -93,7 +117,7 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
{
public:
LLWebRTCImpl() :
- mCustomProcessor(nullptr), mMute(true)
+ mPeerCustomProcessor(nullptr), mMute(true)
{
}
~LLWebRTCImpl() {}
@@ -190,7 +214,8 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
int32_t mRecordingDevice;
bool mMute;
- LLCustomProcessor * mCustomProcessor;
+ LLAudioDeviceObserver * mTuningAudioDeviceObserver;
+ LLCustomProcessor * mPeerCustomProcessor;
// peer connections
std::vector<rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>> mPeerConnections;