fix: different audio frame size than 10ms (#72)

examples/wgpu_room/src/sine_track.rs

@@ -12,6 +12,7 @@ struct FrameData {
     pub sample_rate: u32,
     pub freq: f64,
     pub amplitude: f64,
+    pub phase: u64,
 }
 
 impl Default for FrameData {
@@ -20,6 +21,7 @@ impl Default for FrameData {
             sample_rate: 48000,
             freq: 440.0,
             amplitude: 1.0,
+            phase: 0,
         }
     }
 }
@@ -98,7 +100,7 @@ impl SineTrack {
         loop {
             interval.tick().await;
 
-            let data = frame_options.lock();
+            let mut data = frame_options.lock();
             let samples_count_10ms = (data.sample_rate / 100) as usize;
 
             if samples_10ms.capacity() != samples_count_10ms {
@@ -108,10 +110,13 @@ impl SineTrack {
             for i in 0..samples_count_10ms {
                 let val = data.amplitude
                     * f64::sin(
-                        std::f64::consts::PI * 2.0 * data.freq * i as f64
-                            / samples_count_10ms as f64,
+                        std::f64::consts::PI
+                            * 2.0
+                            * data.freq
+                            * (data.phase as f64 / data.sample_rate as f64),
                     );
 
+                data.phase += 1;
                 // WebRTC uses 16-bit signed PCM
                 samples_10ms[i] = (val * 32768.0) as i16;
             }

livekit-webrtc/src/native/audio_source.rs

@@ -1,16 +1,27 @@
 use crate::audio_frame::AudioFrame;
 use cxx::SharedPtr;
+use std::sync::{Arc, Mutex};
 use webrtc_sys::media_stream as sys_ms;
 
 #[derive(Clone)]
 pub struct NativeAudioSource {
     sys_handle: SharedPtr<sys_ms::ffi::AudioTrackSource>,
+    inner: Arc<Mutex<AudioSourceInner>>,
+}
+
+#[derive(Default)]
+struct AudioSourceInner {
+    buf: Vec<i16>,
+    offset: usize,
+    sample_rate: u32,
+    num_channels: u32,
 }
 
 impl Default for NativeAudioSource {
     fn default() -> Self {
         Self {
             sys_handle: sys_ms::ffi::new_audio_track_source(),
+            inner: Default::default(),
         }
     }
 }
@@ -21,14 +32,51 @@ impl NativeAudioSource {
     }
 
     pub fn capture_frame(&self, frame: &AudioFrame) {
-        // TODO(theomonnom): Should we check for 10ms worth of data here?
-        unsafe {
-            self.sys_handle.on_captured_frame(
-                frame.data.as_ptr(),
-                frame.sample_rate as i32,
-                frame.num_channels as usize,
-                frame.samples_per_channel as usize,
-            )
+        let mut inner = self.inner.lock().unwrap();
+        let samples_10ms = (frame.sample_rate / 100 * frame.num_channels) as usize;
+        if inner.sample_rate != frame.sample_rate || inner.num_channels != frame.num_channels {
+            inner.buf.resize(samples_10ms as usize, 0);
+            inner.offset = 0;
+        }
+
+        // Split the frame into 10ms chunks
+        let mut i = 0;
+        loop {
+            let buf_offset = inner.offset;
+            let remaining_data = frame.data.len() - i; // Remaining data to read inside the frame
+            let needed_data = samples_10ms - buf_offset; // Needed data of "frame.data" to make a complete 10ms from inner.buf
+            if remaining_data < needed_data {
+                if remaining_data > 0 {
+                    // Not enough data to make a complete 10ms frame, store the remaining data inside inner.buf
+                    // It'll be used on the next capture.
+                    inner.buf[buf_offset..buf_offset + remaining_data]
+                        .copy_from_slice(&frame.data[i..]);
+                    inner.offset += remaining_data;
+                }
+
+                break;
+            }
+
+            let data = if inner.offset != 0 {
+                // Use the data from the previous capture
+                let data = &mut inner.buf[buf_offset..];
+                data.copy_from_slice(&frame.data[i..i + needed_data]);
+                inner.offset = 0;
+                &inner.buf
+            } else {
+                &frame.data[i..i + samples_10ms]
+            };
+
+            unsafe {
+                self.sys_handle.on_captured_frame(
+                    data.as_ptr(),
+                    frame.sample_rate as i32,
+                    frame.num_channels as usize,
+                    samples_10ms,
+                )
+            }
+
+            i += needed_data;
         }
     }
 }