refactor: simplify ALSA audio configuration and improve device detection

- Replaced complex multi-PCM asound.conf with streamlined dsnoop-based ch1/ch2 devices using matched 5ms periods
- Updated sounddevice utilities to recognize dsnoop and hw:X,Y patterns for better device filtering
- Adjusted audio input latency parameters to align with new ALSA period configuration

.gitignore

@@ -45,7 +45,5 @@ src/auracast/.env
 src/auracast/server/certs/ca/ca_cert.srl
 src/auracast/server/credentials.json
 pcm1862-i2s.dtbo
-test.wav
-both.wav
-vin1.wav
-vin2.wav
+ch1.wav
+ch2.wav

README.md

@@ -218,6 +218,10 @@ sudo ldconfig                        # refresh linker cache
 	- echo i2c-dev     | sudo tee -a /etc/modules
 - read temp /src/scripts/temp 
 
+# test recording
+ arecord -f cd -c 1 -D record_left left.wav -r48000
+ arecord -f cd -c 1 -D record_right right.wav -r48000
+
 # Known issues:
 - When running on a laptop there might be issues switching between usb and browser audio input since they use the same audio device
 

src/auracast/multicast.py

@@ -84,17 +84,26 @@ class ModSoundDeviceAudioInput(audio_io.SoundDeviceAudioInput):
             logging.warning("Failed to query sounddevice backend/device info: %s", e)
 
         # Create RawInputStream with injected low-latency parameters
+        # Match dsnoop period (5 ms @ 48 kHz -> 240 frames). For other rates, keep 5 ms.
+        _sr = int(self._pcm_format.sample_rate)
+        _block = max(24, _sr // 200)
+        _extra = None
+        try:
+            _extra = sd.AlsaSettings(period_size=_block, periods=2)
+        except Exception:
+            _extra = None
         self._stream = sd.RawInputStream(
             samplerate=self._pcm_format.sample_rate,
             device=self._device,
             channels=self._pcm_format.channels,
             dtype='int16',
-            blocksize=240,  # Match frame size
-            latency=0.010,  
+            blocksize=_block,
+            latency=0.005,
+            extra_settings=_extra,
         )
         self._stream.start()
         
-        logging.info(f"SoundDeviceAudioInput: Opened with blocksize=240, latency=0.010 (10ms)")
+        logging.info(f"SoundDeviceAudioInput: Opened with blocksize={_block}, latency=0.010 (~10ms target)")
         
         return audio_io.PcmFormat(
             audio_io.PcmFormat.Endianness.LITTLE,
@@ -686,9 +695,9 @@ class Streamer():
         enable_drift_compensation = getattr(global_config, 'enable_adaptive_frame_dropping', False)
         # Hardcoded parameters (unit: milliseconds)
         drift_threshold_ms = 2.0 if enable_drift_compensation else 0.0
-        static_drop_ms = 1 if enable_drift_compensation else 0.0
+        static_drop_ms =  drift_threshold_ms if enable_drift_compensation else 0.0
         # Guard interval measured in LC3 frames (10 ms each); 50 => 500 ms cooldown
-        discard_guard_frames = int(2*sample_rate / 1000) if enable_drift_compensation else 0
+        discard_guard_frames = int(sample_rate / 1000) // 2  if enable_drift_compensation else 0
         # Derived sample counts
         drop_threshold_samples = int(sample_rate * drift_threshold_ms / 1000.0)
         static_drop_samples = int(sample_rate * static_drop_ms / 1000.0)
@@ -1126,7 +1135,6 @@ if __name__ == "__main__":
     #config.debug = True
     
     # Enable clock drift compensation to prevent latency accumulation
-    # With ~43 samples/sec drift (0.89ms/sec), threshold of 2ms will trigger every ~2.2 seconds
 
     run_async(
         broadcast(

src/auracast/utils/sounddevice_utils.py

@@ -232,13 +232,19 @@ def get_alsa_usb_inputs():
         name = dev.get('name', '').lower()
         # Filter for USB devices based on common patterns:
         # - Contains 'usb' in the name
-        # - hw:X,Y pattern (ALSA hardware devices)
+        # - hw:X or hw:X,Y pattern present anywhere in name (ALSA hardware devices)
+        # - dsnoop/ch1/ch2 convenience entries from asound.conf
         # Exclude: default, dmix, pulse, pipewire, sysdefault
         if any(exclude in name for exclude in ['default', 'dmix', 'pulse', 'pipewire', 'sysdefault']):
             continue
         
-        # Include if it has 'usb' in name or matches hw:X pattern
-        if 'usb' in name or re.match(r'hw:\d+', name):
+        # Include if it has 'usb' or contains an hw:* token, or matches common dsnoop/mono aliases
+        if (
+            'usb' in name or
+            re.search(r'hw:\d+(?:,\d+)?', name) or
+            name.startswith('dsnoop') or
+            name in ('ch1', 'ch2')
+        ):
             usb_inputs.append((idx, dev))
     
     return usb_inputs

src/misc/asound.conf

@@ -1,39 +1,27 @@
-# --- raw HW by card ID (no sharing) ---
-pcm.adc_hw_stereo {
-  type hw
-  card "i2s"   # from `arecord -l`: card 2: i2s [pcm1862 on i2s]
-  device 0
+pcm.ch1 {
+    type dsnoop
+    ipc_key 234884
+    slave {
+        pcm "hw:CARD=i2s,DEV=0"
+        channels 2
+        rate 48000
+        format S16_LE
+        period_size 240    # 5 ms @ 48 kHz
+        buffer_size 480    # 2 periods (≈10 ms total)
+    }
+    bindings.0  0
 }
 
-# --- mono splits (do the routing here) ---
-pcm.adc_in1 {
-  type route
-  slave.pcm "adc_hw_stereo"
-  slave.channels 2
-  ttable.0.0 1.0   # left -> mono
-  ttable.1.0 0.0
-}
-
-pcm.adc_in2 {
-  type route
-  slave.pcm "adc_hw_stereo"
-  slave.channels 2
-  ttable.0.0 0.0
-  ttable.1.0 1.0   # right -> mono
-}
-
-# --- diagnostics: exact slot taps (bypass any sharing/conversion) ---
-pcm.adc_ch0 {
-  type route
-  slave.pcm "adc_hw_stereo"
-  slave.channels 2
-  ttable.0.0 1.0
-  ttable.1.0 0.0
-}
-pcm.adc_ch1 {
-  type route
-  slave.pcm "adc_hw_stereo"
-  slave.channels 2
-  ttable.0.0 0.0
-  ttable.1.0 1.0
-}
+pcm.ch2 {
+    type dsnoop
+    ipc_key 2241234
+    slave {
+        pcm "hw:CARD=i2s,DEV=0"
+        channels 2
+        rate 48000
+        format S16_LE
+        period_size 240    # 5 ms @ 48 kHz
+        buffer_size 480    # 2 periods (≈10 ms total)
+    }    
+    bindings.0  1
+}

src/scripts/list_sd_nodes.py

@@ -1,16 +1,47 @@
 import sounddevice as sd, pprint
-from auracast.utils.sounddevice_utils import devices_by_backend
+from auracast.utils.sounddevice_utils import (
+    devices_by_backend,
+    get_alsa_inputs,
+    get_alsa_usb_inputs,
+    get_network_pw_inputs,
+    refresh_pw_cache,
+)
 
 print("PortAudio library:", sd._libname)
 print("PortAudio version:", sd.get_portaudio_version())
+
 print("\nHost APIs:")
-pprint.pprint(sd.query_hostapis())
-print("\nDevices:")
-pprint.pprint(sd.query_devices())
+apis = sd.query_hostapis()
+pprint.pprint(apis)
 
-# Example: only PulseAudio devices on Linux
-print("\nOnly PulseAudio devices:")
-for i, d in devices_by_backend("PulseAudio"):
-    print(f"{i}: {d['name']}  in={d['max_input_channels']} out={d['max_output_channels']}")
+print("\nAll Devices (with host API name):")
+devs = sd.query_devices()
+for i, d in enumerate(devs):
+    ha_name = apis[d['hostapi']]['name'] if isinstance(d.get('hostapi'), int) and d['hostapi'] < len(apis) else '?'
+    if d.get('max_input_channels', 0) > 0:
+        print(f"IN  {i:>3}: {d['name']}  api={ha_name}  in={d['max_input_channels']}")
+    elif d.get('max_output_channels', 0) > 0:
+        print(f"OUT {i:>3}: {d['name']}  api={ha_name}  out={d['max_output_channels']}")
+    else:
+        print(f"DEV {i:>3}: {d['name']}  api={ha_name}  (no I/O)")
+
+print("\nALSA input devices (PortAudio ALSA host):")
+for i, d in devices_by_backend('ALSA'):
+    if d.get('max_input_channels', 0) > 0:
+        print(f"ALSA {i:>3}: {d['name']}  in={d['max_input_channels']}")
+
+print("\nALSA USB-filtered inputs:")
+for i, d in get_alsa_usb_inputs():
+    print(f"USB  {i:>3}: {d['name']}  in={d['max_input_channels']}")
+
+print("\nRefreshing PipeWire caches...")
+try:
+    refresh_pw_cache()
+except Exception:
+    pass
+
+print("PipeWire Network inputs (from cache):")
+for i, d in get_network_pw_inputs():
+    print(f"NET  {i:>3}: {d['name']}  in={d.get('max_input_channels', 0)}")