From e6e6664cc95685fc8dc43d02a5ae4e343a319c92 Mon Sep 17 00:00:00 2001
From: Jordan <rollersteaam@gmail.com>
Date: Thu, 19 Mar 2026 16:00:03 +0000
Subject: [PATCH] fix: noise problems when using android microphone

---
 Plugin~/WebRTCPlugin/Context.cpp              |  5 ++-
 .../WebRTCPlugin/UnityAudioTrackSource.cpp    | 35 +++++++++++++++++++
 Plugin~/WebRTCPlugin/UnityAudioTrackSource.h  |  2 ++
 3 files changed, 39 insertions(+), 3 deletions(-)

diff --git a/Plugin~/WebRTCPlugin/Context.cpp b/Plugin~/WebRTCPlugin/Context.cpp
index 5ed9572c45..7b5b0d2b44 100644
--- a/Plugin~/WebRTCPlugin/Context.cpp
+++ b/Plugin~/WebRTCPlugin/Context.cpp
@@ -244,11 +244,10 @@ namespace webrtc
 
     rtc::scoped_refptr<AudioSourceInterface> Context::CreateAudioSource()
     {
-        // avoid optimization specially for voice
         cricket::AudioOptions audioOptions;
-        audioOptions.auto_gain_control = false;
-        audioOptions.noise_suppression = false;
+        audioOptions.noise_suppression = true;
         audioOptions.highpass_filter = false;
+        audioOptions.auto_gain_control = true;
         return UnityAudioTrackSource::Create(audioOptions);
     }
 
diff --git a/Plugin~/WebRTCPlugin/UnityAudioTrackSource.cpp b/Plugin~/WebRTCPlugin/UnityAudioTrackSource.cpp
index 926def49d8..70aa88fd3d 100644
--- a/Plugin~/WebRTCPlugin/UnityAudioTrackSource.cpp
+++ b/Plugin~/WebRTCPlugin/UnityAudioTrackSource.cpp
@@ -1,6 +1,7 @@
 #include "pch.h"
 
 #include <common_audio/include/audio_util.h>
+#include <modules/audio_processing/include/audio_processing.h>
 #include <rtc_base/ref_counted_object.h>
 
 #include "UnityAudioTrackSource.h"
@@ -68,6 +69,16 @@ namespace webrtc
 
         while (_convertedAudioData.size() >= nNumSamplesFor10ms)
         {
+            if (_audioProcessing)
+            {
+                StreamConfig streamConfig(nSampleRate, nNumChannels);
+                _audioProcessing->ProcessStream(
+                    _convertedAudioData.data(),
+                    streamConfig,
+                    streamConfig,
+                    _convertedAudioData.data());
+            }
+
             for (auto sink : _arrSink)
                 sink->OnData(_convertedAudioData.data(), nBitPerSample, nSampleRate, nNumChannels, nNumFramesFor10ms);
             _convertedAudioData.erase(_convertedAudioData.begin(), _convertedAudioData.begin() + nNumSamplesFor10ms);
@@ -78,6 +89,30 @@ namespace webrtc
     UnityAudioTrackSource::UnityAudioTrackSource(const cricket::AudioOptions& audio_options)
         : _options(audio_options)
     {
+        // Build and configure the APM so that the options are actually applied.
+        // cricket::AudioOptions are stored as metadata by LocalAudioSource but are
+        // never wired into any real processing — PushAudioData() bypasses the WebRTC
+        // voice engine entirely, so we must drive the APM ourselves here.
+        AudioProcessing::Config apmConfig;
+
+        apmConfig.noise_suppression.enabled =
+            audio_options.noise_suppression.value_or(true);
+        apmConfig.noise_suppression.level =
+            AudioProcessing::Config::NoiseSuppression::kHigh;
+
+        apmConfig.gain_controller1.enabled =
+            audio_options.auto_gain_control.value_or(true);
+        apmConfig.gain_controller1.mode =
+            AudioProcessing::Config::GainController1::kAdaptiveDigital;
+
+        apmConfig.high_pass_filter.enabled =
+            audio_options.highpass_filter.value_or(false);
+
+        apmConfig.echo_canceller.enabled =
+            audio_options.echo_cancellation.value_or(false);
+
+        _audioProcessing = AudioProcessingBuilder().Create();
+        _audioProcessing->ApplyConfig(apmConfig);
     }
 
     UnityAudioTrackSource::~UnityAudioTrackSource() { }
diff --git a/Plugin~/WebRTCPlugin/UnityAudioTrackSource.h b/Plugin~/WebRTCPlugin/UnityAudioTrackSource.h
index 7fc4a874eb..803118b092 100644
--- a/Plugin~/WebRTCPlugin/UnityAudioTrackSource.h
+++ b/Plugin~/WebRTCPlugin/UnityAudioTrackSource.h
@@ -3,6 +3,7 @@
 #include <mutex>
 
 #include <api/media_stream_interface.h>
+#include <modules/audio_processing/include/audio_processing.h>
 #include <pc/local_audio_source.h>
 
 namespace unity
@@ -34,6 +35,7 @@ namespace webrtc
         std::vector<AudioTrackSinkInterface*> _arrSink;
         std::mutex _mutex;
         cricket::AudioOptions _options;
+        rtc::scoped_refptr<AudioProcessing> _audioProcessing;
         int _sampleRate = 0;
         size_t _numChannels = 0;
         size_t _numFrames = 0;