Latency lowered.

src/auracast/multicast.py

@@ -140,7 +140,7 @@ class AlsaArecordAudioInput(audio_io.AudioInput):
 
 
 class PyAlsaAudioInput(audio_io.ThreadedAudioInput):
-    """PyALSA audio input with callback thread and ring buffer."""
+    """PyALSA audio input with callback thread and ring buffer - supports mono/stereo."""
 
     def __init__(self, device, pcm_format: audio_io.PcmFormat):
         super().__init__()
@@ -153,10 +153,22 @@ class PyAlsaAudioInput(audio_io.ThreadedAudioInput):
         self._ring_lock = threading.Lock()
         self._running = False
         self._callback_thread = None
-        self._max_buffer_bytes = int(self._pcm_format.sample_rate * 0.1 * 2)
+        self._actual_channels = None
+        self._periodsize = None
 
     def _open(self) -> audio_io.PcmFormat:
+        # ========== LATENCY CONFIGURATION ==========
+        # Adjust these parameters to tune latency vs stability
+        ALSA_PERIODSIZE = 120      # Samples per ALSA read (240@48kHz = 5ms, 120 = 2.5ms, 96 = 2ms)
+        ALSA_PERIODS = 2           # Number of periods in ALSA buffer (lower = less latency, more risk of underrun)
+        # Ring buffer: keep only 3 periods max to minimize latency (safety margin only)
+        # ===========================================
+        
         requested_rate = int(self._pcm_format.sample_rate)
+        requested_channels = int(self._pcm_format.channels)
+        self._periodsize = ALSA_PERIODSIZE
+        # Max ring buffer = 3 periods worth of data (tight coupling, minimal latency)
+        self._max_buffer_bytes = ALSA_PERIODSIZE * 3 * 2 * requested_channels
         
         self._pcm = alsaaudio.PCM(
             type=alsaaudio.PCM_CAPTURE,
@@ -164,17 +176,25 @@ class PyAlsaAudioInput(audio_io.ThreadedAudioInput):
             device=self._device,
         )
         
-        self._pcm.setchannels(1)
+        self._pcm.setchannels(requested_channels)
         self._pcm.setformat(alsaaudio.PCM_FORMAT_S16_LE)
         actual_rate = self._pcm.setrate(requested_rate)
-        self._pcm.setperiodsize(240)
+        self._pcm.setperiodsize(ALSA_PERIODSIZE)
+        try:
+            self._pcm.setperiods(ALSA_PERIODS)
+        except AttributeError:
+            pass  # Some pyalsaaudio versions don't have setperiods()
         
-        logging.info("PyALSA: device=%s requested=%d actual=%d periodsize=240 (5ms)", 
+        ring_buf_samples = self._max_buffer_bytes // (2 * requested_channels)
-                     self._device, requested_rate, actual_rate)
+        ring_buf_ms = (ring_buf_samples / actual_rate) * 1000
+        logging.info("PyALSA: device=%s rate=%d ch=%d periodsize=%d (%.1fms) periods=%d ring_buf=%d samples (%.1fms)", 
+                     self._device, actual_rate, requested_channels, ALSA_PERIODSIZE, 
+                     (ALSA_PERIODSIZE / actual_rate) * 1000, ALSA_PERIODS, ring_buf_samples, ring_buf_ms)
         
         if actual_rate != requested_rate:
             logging.warning("PyALSA: Sample rate mismatch! requested=%d actual=%d", requested_rate, actual_rate)
         
+        self._actual_channels = requested_channels
         self._running = True
         self._callback_thread = threading.Thread(target=self._capture_loop, daemon=True)
         self._callback_thread.start()
@@ -183,25 +203,28 @@ class PyAlsaAudioInput(audio_io.ThreadedAudioInput):
             audio_io.PcmFormat.Endianness.LITTLE,
             audio_io.PcmFormat.SampleType.INT16,
             actual_rate,
-            1,
+            requested_channels,
         )
 
     def _capture_loop(self):
         first_read = True
+        hw_channels = None
         while self._running:
             try:
                 length, data = self._pcm.read()
                 if length > 0:
                     if first_read:
-                        expected_bytes = 240 * 2  # 240 frames * 2 bytes/sample for mono
+                        expected_mono = self._periodsize * 2
-                        logging.info("PyALSA first capture: length=%d bytes=%d expected=%d", length, len(data), expected_bytes)
+                        expected_stereo = self._periodsize * 2 * 2
+                        hw_channels = 2 if len(data) == expected_stereo else 1
+                        logging.info("PyALSA first capture: bytes=%d detected_hw_channels=%d requested_channels=%d", 
+                                    len(data), hw_channels, self._actual_channels)
                         first_read = False
                     
-                    # If we got stereo data (480 bytes instead of 240), downsample to mono
+                    # Convert stereo hardware to mono if needed
-                    if len(data) == 960:  # 240 frames * 2 channels * 2 bytes = stereo
+                    if hw_channels == 2 and self._actual_channels == 1:
-                        logging.warning("PyALSA: Got stereo data, converting to mono")
                         pcm_stereo = np.frombuffer(data, dtype=np.int16)
-                        pcm_mono = pcm_stereo[::2]  # Take only left channel
+                        pcm_mono = pcm_stereo[::2]
                         data = pcm_mono.tobytes()
                     
                     with self._ring_lock:
@@ -229,10 +252,13 @@ class PyAlsaAudioInput(audio_io.ThreadedAudioInput):
                         result += chunk[:needed]
                         self._ring_buffer.appendleft(chunk[needed:])
                 else:
-                    break
+                    # Ring buffer empty - release lock and wait a bit
-        
+                    pass
-        if len(result) < bytes_needed:
+            
-            result += b'\x00' * (bytes_needed - len(result))
+            if len(result) < bytes_needed:
+                # Don't busy-wait - sleep briefly to let capture thread fill buffer
+                import time
+                time.sleep(0.0001)  # 0.1ms
         
         return result