fix the 440->441 samples issue and pass a noop callback for release

libwebrtc/src/native/audio_source.rs

@@ -61,7 +61,7 @@ impl NativeAudioSource {
             queue_size_ms.try_into().unwrap(),
         );
 
-        let queue_size_samples = queue_size_ms * (sample_rate / 1000) * num_channels;
+        let queue_size_samples = (queue_size_ms * sample_rate * num_channels) / 1000;
         Self { sys_handle, sample_rate, num_channels, queue_size_samples }
     }
 
@@ -118,16 +118,24 @@ impl NativeAudioSource {
                 });
             }
 
+            // Define a no-op callback for fast path (queue_size_ms=0)
+            // This is safer than passing null, which can cause UB in release mode optimizations
+            extern "C" fn noop_complete_callback(_ctx: *const sys_at::SourceContext) {
+                // No-op: fast path completes synchronously, no callback needed
+            }
+
             unsafe {
-                // Pass null ctx + null callback; C++ ignores them in direct mode
                 let data: &[i16] = frame.data.as_ref();
+                // Use a valid no-op callback instead of null for safety
+                // In release mode, transmuting null pointers can cause UB
+                let noop_callback = sys_at::CompleteCallback(noop_complete_callback);
                 let ok = self.sys_handle.capture_frame(
                     data,
                     self.sample_rate,
                     self.num_channels,
                     nb_frames,
-                    std::ptr::null(),
-                    std::mem::zeroed::<sys_at::CompleteCallback>(),
+                    std::ptr::null(), // Context is still null - callback won't use it
+                    noop_callback,
                 );
                 if !ok {
                     return Err(RtcError {

webrtc-sys/include/livekit/audio_track.h

@@ -140,10 +140,10 @@ class AudioTrackSource {
     int missed_frames_ RTC_GUARDED_BY(mutex_) = 0;
     std::vector<int16_t> silence_buffer_;
 
-    int sample_rate_;
-    int num_channels_;
-    int queue_size_samples_;
-    int notify_threshold_samples_;
+    int sample_rate_ = 0;
+    int num_channels_ = 0;
+    int queue_size_samples_ = 0;
+    int notify_threshold_samples_ = 0;
 
     cricket::AudioOptions options_{};
   };

webrtc-sys/src/audio_track.cpp

@@ -141,8 +141,11 @@ AudioTrackSource::InternalSource::InternalSource(
       num_channels_(num_channels),
       capture_userdata_(nullptr),
       on_complete_(nullptr) {
-  if (!queue_size_ms)
+  if (!queue_size_ms) {
+    // Set queue_size_samples_ to 0 so that capture_frame() will get to the fast path.
+    queue_size_samples_ = 0;
     return;  // no audio queue
+  }
 
   // start sending silence when there is nothing on the queue for 10 frames
   // (100ms)