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-rw-r--r--indra/llwebrtc/llwebrtc.cpp83
-rw-r--r--indra/llwebrtc/llwebrtc_impl.h37
2 files changed, 68 insertions, 52 deletions
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index d6eff5e1f2..f0d7f6ad8f 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -27,6 +27,7 @@
#include "llwebrtc_impl.h"
#include <algorithm>
#include <format>
+#include <string.h>
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
@@ -34,30 +35,54 @@
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/media_stream_interface.h"
#include "api/media_stream_track.h"
+#include "modules/audio_processing/audio_buffer.h"
namespace llwebrtc
{
-const float VOLUME_SCALE_WEBRTC = 100;
-
-LLAudioDeviceObserver::LLAudioDeviceObserver() : mMicrophoneEnergy(0.0), mSumVector {0} {}
+LLCustomProcessor::LLCustomProcessor() :
+ mSampleRateHz(0),
+ mNumChannels(0)
+{
+ memset(mSumVector, sizeof(mSumVector), 0);
+}
-float LLAudioDeviceObserver::getMicrophoneEnergy() { return mMicrophoneEnergy; }
+void LLCustomProcessor::Initialize(int sample_rate_hz, int num_channels)
+{
+ mSampleRateHz = sample_rate_hz;
+ mNumChannels = num_channels;
+ memset(mSumVector, sizeof(mSumVector), 0);
+}
-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)
+void LLCustomProcessor::Process(webrtc::AudioBuffer *audio_in)
{
+ webrtc::StreamConfig stream_config;
+ stream_config.set_sample_rate_hz(mSampleRateHz);
+ stream_config.set_num_channels(mNumChannels);
+ std::vector<float *> frame;
+ std::vector<float> frame_samples;
+
+ if (audio_in->num_channels() < 1 || audio_in->num_frames() < 480)
+ {
+ return;
+ }
+
+ frame_samples.resize(stream_config.num_samples());
+ frame.resize(stream_config.num_channels());
+ for (size_t ch = 0; ch < stream_config.num_channels(); ++ch)
+ {
+ frame[ch] = &(frame_samples)[ch * stream_config.num_frames()];
+ }
+
+ audio_in->CopyTo(stream_config, &frame[0]);
+
float energy = 0;
- const short *samples = (const short *) audio_samples;
- for (size_t index = 0; index < num_samples * num_channels; index++)
+ for (size_t index = 0; index < stream_config.num_samples(); index++)
{
- float sample = (static_cast<float>(samples[index]) / (float) 32767);
+ float sample = frame_samples[index];
energy += sample * sample;
}
-
+
// smooth it.
size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
float totalSum = 0;
@@ -69,15 +94,7 @@ void LLAudioDeviceObserver::OnCaptureData(const void *audio_samples,
}
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)
-{
+ mMicrophoneEnergy = std::sqrt(totalSum / (stream_config.num_samples() * buffer_size));
}
void LLWebRTCImpl::init()
@@ -104,11 +121,9 @@ void LLWebRTCImpl::init()
mWorkerThread->PostTask(
[this]()
{
- mTuningAudioDeviceObserver = new LLAudioDeviceObserver;
- mTuningDeviceModule =
- webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
- mTaskQueueFactory.get(),
- std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
+ mTuningDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(
+ webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
+ mTaskQueueFactory.get(), nullptr);
mTuningDeviceModule->Init();
mTuningDeviceModule->SetStereoRecording(true);
mTuningDeviceModule->SetStereoPlayout(true);
@@ -134,11 +149,9 @@ void LLWebRTCImpl::init()
mWorkerThread->BlockingCall(
[this]()
{
- mPeerAudioDeviceObserver = new LLAudioDeviceObserver;
mPeerDeviceModule =
webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
- mTaskQueueFactory.get(),
- std::unique_ptr<webrtc::AudioDeviceDataObserver>(mPeerAudioDeviceObserver));
+ mTaskQueueFactory.get(), nullptr);
mPeerDeviceModule->Init();
mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
@@ -151,7 +164,11 @@ void LLWebRTCImpl::init()
mPeerDeviceModule->InitPlayout();
});
- rtc::scoped_refptr<webrtc::AudioProcessing> apm = webrtc::AudioProcessingBuilder().Create();
+ mCustomProcessor = new LLCustomProcessor;
+ webrtc::AudioProcessingBuilder apb;
+ apb.SetCapturePostProcessing(std::unique_ptr<webrtc::CustomProcessing>(mCustomProcessor));
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm = apb.Create();
+
webrtc::AudioProcessing::Config apm_config;
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = false;
@@ -454,9 +471,9 @@ void LLWebRTCImpl::setTuningMode(bool enable)
}
}
-float LLWebRTCImpl::getTuningAudioLevel() { return -20 * log10f(mTuningAudioDeviceObserver->getMicrophoneEnergy()); }
+float LLWebRTCImpl::getTuningAudioLevel() { return -20 * log10f(mCustomProcessor->getMicrophoneEnergy()); }
-float LLWebRTCImpl::getPeerAudioLevel() { return -20 * log10f(mPeerAudioDeviceObserver->getMicrophoneEnergy()); }
+float LLWebRTCImpl::getPeerAudioLevel() { return -20 * log10f(mCustomProcessor->getMicrophoneEnergy()); }
//
// Helpers
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
index 3ccb6058ad..419482b751 100644
--- a/indra/llwebrtc/llwebrtc_impl.h
+++ b/indra/llwebrtc/llwebrtc_impl.h
@@ -65,27 +65,27 @@ namespace llwebrtc
class LLWebRTCPeerConnectionImpl;
-class LLAudioDeviceObserver : public webrtc::AudioDeviceDataObserver
+class LLCustomProcessor : public webrtc::CustomProcessing
{
public:
- LLAudioDeviceObserver();
+ LLCustomProcessor();
+ ~LLCustomProcessor() override {}
- float getMicrophoneEnergy();
+ // (Re-) Initializes the submodule.
+ void Initialize(int sample_rate_hz, int num_channels) override;
+ // Analyzes the given capture or render signal.
+ void Process(webrtc::AudioBuffer *audio) override;
+ // Returns a string representation of the module state.
+ std::string ToString() const override { return ""; }
- 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;
+ float getMicrophoneEnergy() { return mMicrophoneEnergy; }
protected:
- float mSumVector[30]; // 300 ms of smoothing
+ static const int NUM_PACKETS_TO_FILTER = 30; // 300 ms of smoothing
+ int mSampleRateHz;
+ int mNumChannels;
+
+ float mSumVector[NUM_PACKETS_TO_FILTER];
float mMicrophoneEnergy;
};
@@ -93,7 +93,7 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
{
public:
LLWebRTCImpl() :
- mTuningAudioDeviceObserver(nullptr), mPeerAudioDeviceObserver(nullptr), mMute(true)
+ mCustomProcessor(nullptr), mMute(true)
{
}
~LLWebRTCImpl() {}
@@ -189,9 +189,8 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS
int32_t mPlayoutDevice;
int32_t mRecordingDevice;
bool mMute;
-
- LLAudioDeviceObserver * mTuningAudioDeviceObserver;
- LLAudioDeviceObserver * mPeerAudioDeviceObserver;
+
+ LLCustomProcessor * mCustomProcessor;
// peer connections
std::vector<rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>> mPeerConnections;
generated by cgit v1.2.3 (git 2.25.1) at 2025-11-04 02:43:30 +0000