fix: synchronize CustomAudioSource's sinks_ against AddSink/RemoveSink

webrtc-jni/src/main/cpp/include/media/audio/CustomAudioSource.h

@@ -19,6 +19,7 @@
 
 #include "api/media_stream_interface.h"
 #include "rtc_base/ref_counted_object.h"
+#include "rtc_base/synchronization/mutex.h"
 
 #include "media/SyncClock.h"
 
@@ -50,9 +51,13 @@ namespace jni
             void SetAudioCaptureDelay(int64_t delay_us);
 
         private:
+            // Guards sinks_ and audio_capture_delay_us_ against concurrent
+            // AddSink()/RemoveSink() (called by WebRTC's own signaling/worker
+            // thread as tracks attach/detach) racing with PushAudioData() and
+            // SetAudioCaptureDelay() (called by the application's own threads).
+            webrtc::Mutex mutex_;
             std::vector<webrtc::AudioTrackSinkInterface*> sinks_;
             std::shared_ptr<SyncClock> clock_;
-            //webrtc::CriticalSection crit_;
             std::atomic<int64_t> total_samples_captured_;
             int64_t audio_capture_delay_us_;
     };

webrtc-jni/src/main/cpp/src/media/audio/CustomAudioSource.cpp

@@ -24,14 +24,14 @@ namespace jni
 
     void CustomAudioSource::AddSink(webrtc::AudioTrackSinkInterface * sink)
     {
-        //webrtc::CritScope lock(&crit_);
+        webrtc::MutexLock lock(&mutex_);
 
         sinks_.push_back(sink);
     }
 
     void CustomAudioSource::RemoveSink(webrtc::AudioTrackSinkInterface * sink)
     {
-        //webrtc::CritScope lock(&crit_);
+        webrtc::MutexLock lock(&mutex_);
 
         sinks_.erase(std::remove(sinks_.begin(), sinks_.end(), sink), sinks_.end());
     }
@@ -50,7 +50,7 @@ namespace jni
                                          int sample_rate, size_t number_of_channels,
                                          size_t number_of_frames)
     {
-        //webrtc::CritScope lock(&crit_);
+        webrtc::MutexLock lock(&mutex_);
 
         // Calculate absolute capture time
         int64_t timestamp_us = clock_->GetTimestampUs();
@@ -77,6 +77,8 @@ namespace jni
 
     void CustomAudioSource::SetAudioCaptureDelay(int64_t delay_us)
     {
+        webrtc::MutexLock lock(&mutex_);
+
         audio_capture_delay_us_ = delay_us;
     }
 }

webrtc/src/test/java/dev/onvoid/webrtc/media/audio/CustomAudioSourceTest.java

@@ -83,6 +83,37 @@ void pushAudioData() {
         testAudioFormat(16, 48000, 2, 480);
     }
 
+    @Test
+    void concurrentAddRemoveSinkDoesNotCrash() throws InterruptedException {
+        // Regression test: AddSink()/RemoveSink() (called here from this thread,
+        // mirroring the internal thread WebRTC uses as tracks attach/detach) used
+        // to race unsynchronized with PushAudioData() (the application's capture
+        // thread) over the native sinks_ vector.
+        AudioTrack audioTrack = factory.createAudioTrack("audioTrack", customAudioSource);
+        AudioTrackSink sink = (data, bitsPerSample, sampleRate, channels, frames) -> { };
+
+        byte[] audioData = new byte[480 * 2 * 2]; // 10ms of 48kHz stereo 16-bit audio
+
+        AtomicBoolean running = new AtomicBoolean(true);
+
+        Thread pushThread = new Thread(() -> {
+            while (running.get()) {
+                customAudioSource.pushAudio(audioData, 16, 48000, 2, 480);
+            }
+        });
+        pushThread.start();
+
+        for (int i = 0; i < 5000; i++) {
+            audioTrack.addSink(sink);
+            audioTrack.removeSink(sink);
+        }
+
+        running.set(false);
+        pushThread.join(5000);
+
+        audioTrack.dispose();
+    }
+
     @Test
     void pushAudioWithDifferentFormats() {
         testAudioFormat(8, 8000, 1, 80);    // 8-bit, 8kHz, mono, 10ms