diff options
Diffstat (limited to 'indra/llwebrtc/llwebrtc.cpp')
| -rw-r--r-- | indra/llwebrtc/llwebrtc.cpp | 83 |
1 files changed, 50 insertions, 33 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 |