diff --git a/examples/playogg_from_disk/src/main.rs b/examples/playogg_from_disk/src/main.rs
index 6dc27f7d..0353843d 100644
--- a/examples/playogg_from_disk/src/main.rs
+++ b/examples/playogg_from_disk/src/main.rs
@@ -74,33 +74,28 @@ async fn main() -> Result<(), Box<dyn Error>> {
     tokio::spawn({
         let room = room.clone();
         async move {
-            const FRAME_DURATION: Duration = Duration::from_millis(1000); // Write 1s of audio at a time
-            let ms = FRAME_DURATION.as_millis() as u32;
-            let num_samples = (header.sample_rate / 1000 * ms) as usize;
-
-
             log::info!("sample_rate: {}", header.sample_rate);
             log::info!("num_channels: {}", header.channels);
-            log::info!("chunk size: {}ms - {} samples", ms, num_samples);
-
-            let mut last_granule: u64 = 0;
 
             while let Ok((page_data, page_header)) = reader.parse_next_page() {
-                let sample_count = page_header.granule_position - last_granule;
-                let frame_size = page_data.len();
-                last_granule = page_header.granule_position;
 
+                let frame_size = page_data.len();
                 let mut audio_frame = AudioFrame_u8 {
                     data: vec![0u8; 0].into(),
                     num_channels: header.channels as u32,
                     sample_rate: header.sample_rate,
-                    samples_per_channel: sample_count as u32,
+                    samples_per_channel: (frame_size / header.channels as usize) as u32,
                 };
 
+                let mut bytes = 0;
                 for d in page_data.iter() {
+                    bytes += 1;
                     audio_frame.data.push(*d);
                 }
 
+                log::info!("samples_per_channel: {}, bytes: {}, frame_size: {}",
+                    audio_frame.samples_per_channel, bytes, frame_size);
+
                 source.capture_frame(&audio_frame).await.unwrap();
             }
 
diff --git a/webrtc-sys/src/audio_track.cpp b/webrtc-sys/src/audio_track.cpp
index f99ac92d..a0075411 100644
--- a/webrtc-sys/src/audio_track.cpp
+++ b/webrtc-sys/src/audio_track.cpp
@@ -150,7 +150,7 @@ void AudioTrackSource::InternalSource::on_captured_frame_u8(
     size_t number_of_frames) {
   webrtc::MutexLock lock(&mutex_);
   for (auto sink : sinks_) {
-    sink->OnData(data.data(), 16, sample_rate, number_of_channels,
+    sink->OnData(data.data(), 8, 2 * sample_rate, number_of_channels,
                  number_of_frames);
   }
 }