fix(smallwebrtc): respect audio_out_10ms_chunks parameter in RawAudioTrack

The RawAudioTrack class was hardcoded to always produce 10ms audio frames
regardless of the audio_out_10ms_chunks transport parameter. This caused
firmware clients to receive 20ms chunks even when 40ms was configured.

Changes:
- Add num_10ms_chunks parameter to RawAudioTrack constructor
- Update add_audio_bytes to chunk audio based on configured size
- Update recv() to produce frames of the configured size
- Pass audio_out_10ms_chunks from TransportParams when creating track

src/pipecat/transports/smallwebrtc/transport.py

@@ -78,19 +78,24 @@ class RawAudioTrack(AudioStreamTrack):
     supporting queued audio data with proper synchronization.
     """
 
-    def __init__(self, sample_rate):
+    def __init__(self, sample_rate: int, num_10ms_chunks: int = 1):
         """Initialize the raw audio track.
 
         Args:
             sample_rate: The audio sample rate in Hz.
+            num_10ms_chunks: Number of 10ms chunks per output frame (default 1).
         """
         super().__init__()
         self._sample_rate = sample_rate
+        self._num_10ms_chunks = num_10ms_chunks
         self._samples_per_10ms = sample_rate * 10 // 1000
         self._bytes_per_10ms = self._samples_per_10ms * 2  # 16-bit (2 bytes per sample)
+        # Calculate chunk size based on num_10ms_chunks
+        self._samples_per_chunk = self._samples_per_10ms * num_10ms_chunks
+        self._bytes_per_chunk = self._bytes_per_10ms * num_10ms_chunks
         self._timestamp = 0
         self._start = time.time()
-        # Queue of (bytes, future), broken into 10ms sub chunks as needed
+        # Queue of (bytes, future), broken into configured chunk sizes as needed
         self._chunk_queue = deque()
 
     def add_audio_bytes(self, audio_bytes: bytes):
@@ -103,17 +108,20 @@ class RawAudioTrack(AudioStreamTrack):
             A Future that completes when the data is processed.
 
         Raises:
-            ValueError: If audio bytes are not a multiple of 10ms size.
+            ValueError: If audio bytes are not a multiple of the configured chunk size.
         """
-        if len(audio_bytes) % self._bytes_per_10ms != 0:
-            raise ValueError("Audio bytes must be a multiple of 10ms size.")
+        if len(audio_bytes) % self._bytes_per_chunk != 0:
+            raise ValueError(
+                f"Audio bytes must be a multiple of {self._num_10ms_chunks * 10}ms size "
+                f"({self._bytes_per_chunk} bytes)."
+            )
         future = asyncio.get_running_loop().create_future()
 
-        # Break input into 10ms chunks
-        for i in range(0, len(audio_bytes), self._bytes_per_10ms):
-            chunk = audio_bytes[i : i + self._bytes_per_10ms]
+        # Break input into configured chunk sizes
+        for i in range(0, len(audio_bytes), self._bytes_per_chunk):
+            chunk = audio_bytes[i : i + self._bytes_per_chunk]
             # Only the last chunk carries the future to be resolved once fully consumed
-            fut = future if i + self._bytes_per_10ms >= len(audio_bytes) else None
+            fut = future if i + self._bytes_per_chunk >= len(audio_bytes) else None
             self._chunk_queue.append((chunk, fut))
 
         return future
@@ -135,7 +143,7 @@ class RawAudioTrack(AudioStreamTrack):
             if future and not future.done():
                 future.set_result(True)
         else:
-            chunk = bytes(self._bytes_per_10ms)  # silence
+            chunk = bytes(self._bytes_per_chunk)  # silence
 
         # Convert the byte data to an ndarray of int16 samples
         samples = np.frombuffer(chunk, dtype=np.int16)
@@ -145,7 +153,7 @@ class RawAudioTrack(AudioStreamTrack):
         frame.sample_rate = self._sample_rate
         frame.pts = self._timestamp
         frame.time_base = fractions.Fraction(1, self._sample_rate)
-        self._timestamp += self._samples_per_10ms
+        self._timestamp += self._samples_per_chunk
         return frame
 
 
@@ -493,7 +501,10 @@ class SmallWebRTCClient:
         self._video_input_track = self._webrtc_connection.video_input_track()
         self._screen_video_track = self._webrtc_connection.screen_video_input_track()
         if self._params.audio_out_enabled:
-            self._audio_output_track = RawAudioTrack(sample_rate=self._out_sample_rate)
+            self._audio_output_track = RawAudioTrack(
+                sample_rate=self._out_sample_rate,
+                num_10ms_chunks=self._params.audio_out_10ms_chunks,
+            )
             self._webrtc_connection.replace_audio_track(self._audio_output_track)
 
         if self._params.video_out_enabled: