Latency lowered.

pyproject.toml

@@ -17,7 +17,8 @@ dependencies = [
     "sounddevice (>=0.5.2,<0.6.0)",
     "python-dotenv (>=1.1.1,<2.0.0)",
     "smbus2 (>=0.5.0,<0.6.0)",
-    "samplerate (>=0.2.2,<0.3.0)"
+    "samplerate (>=0.2.2,<0.3.0)",
+    "pyalsaaudio (>=0.9.0,<1.0.0)"
 ]
 
 [project.optional-dependencies]

src/auracast/multicast.py

@@ -56,7 +56,7 @@ from auracast.utils.webrtc_audio_input import WebRTCAudioInput
 
 
 # Patch sounddevice.InputStream globally to use low-latency settings
-import sounddevice as sd
+import alsaaudio
 from collections import deque
 
 
@@ -139,96 +139,138 @@ class AlsaArecordAudioInput(audio_io.AudioInput):
         self._proc = None
 
 
-class ModSoundDeviceAudioInput(audio_io.SoundDeviceAudioInput):
-    """Patched SoundDeviceAudioInput with low-latency capture and adaptive resampling."""
+class PyAlsaAudioInput(audio_io.ThreadedAudioInput):
+    """PyALSA audio input with callback thread and ring buffer - supports mono/stereo."""
 
-    def _open(self):
-        """Create RawInputStream with low-latency parameters and initialize ring buffer."""
-        dev_info = sd.query_devices(self._device)
-        hostapis = sd.query_hostapis()
-        api_index = dev_info.get('hostapi')
-        api_name = hostapis[api_index]['name'] if isinstance(api_index, int) and 0 <= api_index < len(hostapis) else 'unknown'
-        pa_ver = sd.get_portaudio_version()
+    def __init__(self, device, pcm_format: audio_io.PcmFormat):
+        super().__init__()
+        self._device = str(device) if not isinstance(device, str) else device
+        if self._device.isdigit():
+            self._device = 'default' if self._device == '0' else f'hw:{self._device}'
+        self._pcm_format = pcm_format
+        self._pcm = None
+        self._ring_buffer = deque()
+        self._ring_lock = threading.Lock()
+        self._running = False
+        self._callback_thread = None
+        self._actual_channels = None
+        self._periodsize = None
 
-        logging.info(
-            "SoundDevice backend=%s device='%s' (id=%s) ch=%s default_low_input_latency=%.4f default_high_input_latency=%.4f portaudio=%s",
-            api_name,
-            dev_info.get('name'),
-            self._device,
-            dev_info.get('max_input_channels'),
-            float(dev_info.get('default_low_input_latency') or 0.0),
-            float(dev_info.get('default_high_input_latency') or 0.0),
-            pa_ver[1] if isinstance(pa_ver, tuple) and len(pa_ver) >= 2 else pa_ver,
-        )
-        # Create RawInputStream with injected low-latency parameters
-        # Target ~2 ms blocksize (48 kHz -> 96 frames). For other rates, keep ~2 ms.
-        _sr = int(self._pcm_format.sample_rate)
-
-        self.counter=0
-        self.max_avail=0
-        self.logfile_name="available_samples.txt"
-        self.blocksize = 120
-
-        if os.path.exists(self.logfile_name):
-            os.remove(self.logfile_name)
-
-        self._stream = sd.RawInputStream(
-            samplerate=self._pcm_format.sample_rate,
+    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,
+            mode=alsaaudio.PCM_NORMAL,
             device=self._device,
-            channels=self._pcm_format.channels,
-            dtype='int16',
-            blocksize=self.blocksize,
-            latency=0.004,
         )
-        self._stream.start()
-
+        
+        self._pcm.setchannels(requested_channels)
+        self._pcm.setformat(alsaaudio.PCM_FORMAT_S16_LE)
+        actual_rate = self._pcm.setrate(requested_rate)
+        self._pcm.setperiodsize(ALSA_PERIODSIZE)
+        try:
+            self._pcm.setperiods(ALSA_PERIODS)
+        except AttributeError:
+            pass  # Some pyalsaaudio versions don't have setperiods()
+        
+        ring_buf_samples = self._max_buffer_bytes // (2 * requested_channels)
+        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()
+        
         return audio_io.PcmFormat(
             audio_io.PcmFormat.Endianness.LITTLE,
             audio_io.PcmFormat.SampleType.INT16,
-            self._pcm_format.sample_rate,
-            1,
+            actual_rate,
+            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_mono = self._periodsize * 2
+                        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
+                    
+                    # Convert stereo hardware to mono if needed
+                    if hw_channels == 2 and self._actual_channels == 1:
+                        pcm_stereo = np.frombuffer(data, dtype=np.int16)
+                        pcm_mono = pcm_stereo[::2]
+                        data = pcm_mono.tobytes()
+                    
+                    with self._ring_lock:
+                        self._ring_buffer.append(data)
+                        total_bytes = sum(len(chunk) for chunk in self._ring_buffer)
+                        while total_bytes > self._max_buffer_bytes:
+                            self._ring_buffer.popleft()
+                            total_bytes = sum(len(chunk) for chunk in self._ring_buffer)
+            except:
+                if self._running:
+                    break
+
     def _read(self, frame_size: int) -> bytes:
-        """Read PCM samples from the stream."""
-
-        #if self.counter % 50 == 0:
-        frame_size = frame_size + 1 # consume samples a little faster to avoid latency akkumulation
-
-        pcm_buffer, overflowed = self._stream.read(frame_size)
-        if overflowed:
-            logging.warning("SoundDeviceAudioInput: overflowed")
-
-        n_available = self._stream.read_available
-
-        # adapt = n_available > 20
-        # if adapt:
-        #     pcm_extra, overflowed = self._stream.read(3)
-        #     logging.info('consuming extra samples, available was %d', n_available)
-        #     if overflowed:
-        #         logging.warning("SoundDeviceAudioInput: overflowed")
+        bytes_needed = frame_size * 2
+        result = b''
+        
+        while len(result) < bytes_needed:
+            with self._ring_lock:
+                if self._ring_buffer:
+                    chunk = self._ring_buffer.popleft()
+                    needed = bytes_needed - len(result)
+                    if len(chunk) <= needed:
+                        result += chunk
+                    else:
+                        result += chunk[:needed]
+                        self._ring_buffer.appendleft(chunk[needed:])
+                else:
+                    # Ring buffer empty - release lock and wait a bit
+                    pass
             
-        #     out = bytes(pcm_buffer) +  bytes(pcm_extra)
-        # else:
-        out = bytes(pcm_buffer)     
+            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
 
-        self.max_avail = max(self.max_avail, n_available)
+    def _close(self) -> None:
+        self._running = False
+        if self._callback_thread:
+            self._callback_thread.join(timeout=1.0)
+        if self._pcm:
+            self._pcm.close()
+            self._pcm = None
 
-        #Diagnostics 
-        #with open(self.logfile_name, "a", encoding="utf-8") as f:
-        #    f.write(f"{n_available}, {adapt}, {round(self._runavg, 2)}, {overflowed}\n")
-
-        if self.counter % 500 == 0:
-            logging.info(
-                "read available=%d, max=%d, latency:%d", 
-                n_available, self.max_avail, self._stream.latency
-            )
-            self.max_avail = 0
-
-        self.counter += 1
-        return out
-
-audio_io.SoundDeviceAudioInput = ModSoundDeviceAudioInput
+audio_io.SoundDeviceAudioInput = PyAlsaAudioInput
 
 # modified from bumble
 class ModWaveAudioInput(audio_io.ThreadedAudioInput):