Add new reset mechanism.

Co-authored-by: Pbopbo <p.obernesser@freenet.de>
Reviewed-on: #26

.gitignore

@@ -51,4 +51,5 @@ src/auracast/available_samples.txt
 src/auracast/server/stream_settings2.json
 src/scripts/temperature_log*
 
-src/auracast/server/recordings/
+src/auracast/server/recordings/
+src/auracast/server/led_settings.json

poetry.lock

@@ -270,51 +270,6 @@ optional = false
 python-versions = ">=3.9"
 groups = ["main"]
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 ]
 
@@ -1849,6 +1804,22 @@ files = [
     {file = "protobuf-6.30.2.tar.gz", hash = "sha256:35c859ae076d8c56054c25b59e5e59638d86545ed6e2b6efac6be0b6ea3ba048"},
 ]
 
+[[package]]
+name = "pyalsaaudio"
+version = "0.11.0"
+description = "ALSA bindings"
+optional = false
+python-versions = "*"
+groups = ["main"]
+files = []
+develop = false
+
+[package.source]
+type = "git"
+url = "ssh://git@gitea.summitwave.work:222/auracaster/sw_pyalsaaudio.git"
+reference = "b3d11582e03df6929b2e7acbaa1306afc7b8a6bc"
+resolved_reference = "b3d11582e03df6929b2e7acbaa1306afc7b8a6bc"
+
 [[package]]
 name = "pyarrow"
 version = "20.0.0"
@@ -2443,6 +2414,22 @@ files = [
     {file = "rpds_py-0.25.1.tar.gz", hash = "sha256:8960b6dac09b62dac26e75d7e2c4a22efb835d827a7278c34f72b2b84fa160e3"},
 ]
 
+[[package]]
+name = "rpi-gpio"
+version = "0.7.1"
+description = "A module to control Raspberry Pi GPIO channels"
+optional = false
+python-versions = "*"
+groups = ["main"]
+files = [
+    {file = "RPi.GPIO-0.7.1-cp27-cp27mu-linux_armv6l.whl", hash = "sha256:b86b66dc02faa5461b443a1e1f0c1d209d64ab5229696f32fb3b0215e0600c8c"},
+    {file = "RPi.GPIO-0.7.1-cp310-cp310-linux_armv6l.whl", hash = "sha256:57b6c044ef5375a78c8dda27cdfadf329e76aa6943cd6cffbbbd345a9adf9ca5"},
+    {file = "RPi.GPIO-0.7.1-cp37-cp37m-linux_armv6l.whl", hash = "sha256:77afb817b81331ce3049a4b8f94a85e41b7c404d8e56b61ac0f1eb75c3120868"},
+    {file = "RPi.GPIO-0.7.1-cp38-cp38-linux_armv6l.whl", hash = "sha256:29226823da8b5ccb9001d795a944f2e00924eeae583490f0bc7317581172c624"},
+    {file = "RPi.GPIO-0.7.1-cp39-cp39-linux_armv6l.whl", hash = "sha256:15311d3b063b71dee738cd26570effc9985a952454d162937c34e08c0fc99902"},
+    {file = "RPi.GPIO-0.7.1.tar.gz", hash = "sha256:cd61c4b03c37b62bba4a5acfea9862749c33c618e0295e7e90aa4713fb373b70"},
+]
+
 [[package]]
 name = "samplerate"
 version = "0.2.2"
@@ -2976,4 +2963,4 @@ test = ["pytest", "pytest-asyncio"]
 [metadata]
 lock-version = "2.1"
 python-versions = ">=3.11"
-content-hash = "3c9f92c7a5af40f98da9c7824d9c2a6f7eb809e91e43cfef4995761b2e887256"
+content-hash = "7bccf2978170ead195e1e8cff151823a5276823195a239622186fcec830154d9"

pyproject.toml

@@ -17,7 +17,9 @@ 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)",
+    "rpi-gpio (>=0.7.1,<0.8.0)",
+    "pyalsaaudio @ git+ssh://git@gitea.summitwave.work:222/auracaster/sw_pyalsaaudio.git@b3d11582e03df6929b2e7acbaa1306afc7b8a6bc"
 ]
 
 [project.optional-dependencies]

src/auracast/auracast_config.py

@@ -55,7 +55,6 @@ class AuracastBigConfig(BaseModel):
     precode_wav: bool = False
     iso_que_len: int = 64
     num_bis: int = 1  # 1 = mono (FRONT_LEFT), 2 = stereo (FRONT_LEFT + FRONT_RIGHT)
-    input_gain_db: float = 0.0  # Software gain boost in dB applied before LC3 encoding (0 = off, max 20)
 
 class AuracastBigConfigDeu(AuracastBigConfig):
     id: int = 12
@@ -112,4 +111,5 @@ class AuracastConfigGroup(AuracastGlobalConfig):
     bigs: List[AuracastBigConfig] = [
         AuracastBigConfigDeu(),
     ]
-    analog_gain: int = 50  # ADC gain level for analog mode (10-60%)
+    analog_gain_db_left: float = 0.0  # ADC gain level for analog mode left channel (-12 to 18 dB)
+    analog_gain_db_right: float = 0.0  # ADC gain level for analog mode right channel (-12 to 18 dB)

src/auracast/multicast.py

@@ -30,6 +30,7 @@ import time
 import threading
 
 import numpy as np  # for audio down-mix
+import samplerate
 import os
 
 import lc3  # type: ignore  # pylint: disable=E0401
@@ -56,7 +57,7 @@ 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 +140,146 @@ 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 non-blocking reads - 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__()
+        logging.info("PyALSA: device = %s", device)
+        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._actual_channels = None
+        self._periodsize = None
+        self._hw_channels = None
+        self._first_read = True
+        self._resampler = None
+        self._resampler_buffer = np.empty(0, dtype=np.float32)
 
-        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:
+        ALSA_PERIODSIZE = 240
+        ALSA_PERIODS = 4
+        ALSA_MODE = alsaaudio.PCM_NONBLOCK
+        
+        requested_rate = int(self._pcm_format.sample_rate)
+        requested_channels = int(self._pcm_format.channels)
+        self._periodsize = ALSA_PERIODSIZE
+        
+        self._pcm = alsaaudio.PCM(
+            type=alsaaudio.PCM_CAPTURE,
+            mode=ALSA_MODE,
             device=self._device,
-            channels=self._pcm_format.channels,
-            dtype='int16',
-            blocksize=self.blocksize,
-            latency=0.004,
+            periods=ALSA_PERIODS,
         )
-        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)
+        
+        logging.info("PyALSA: device=%s rate=%d ch=%d periodsize=%d (%.1fms) periods=%d mode=%s", 
+                     self._device, actual_rate, requested_channels, ALSA_PERIODSIZE, 
+                     (ALSA_PERIODSIZE / actual_rate) * 1000, ALSA_PERIODS, ALSA_MODE)
+        
+        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._resampler = samplerate.Resampler('sinc_fastest', channels=requested_channels)
+        self._resampler_buffer = np.empty(0, dtype=np.float32)
 
+        self._bang_bang = 0
+        
         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 _read(self, frame_size: int) -> bytes:
-        """Read PCM samples from the stream."""
+        try:
+            avail = self._pcm.avail()
+            logging.debug("PyALSA: avail before read: %d", avail)
+            length, data = self._pcm.read_sw(frame_size + self._bang_bang)
+            avail = self._pcm.avail()
+            SETPOINT = 120 
+            TOLERANCE = 40 
+            if avail < SETPOINT - TOLERANCE:
+                self._bang_bang = -1 
+            elif avail > SETPOINT + TOLERANCE:
+                self._bang_bang = 1
+            else:
+                self._bang_bang = 0
+                
+                
+            logging.debug("PyALSA: read length=%d, data length=%d, avail=%d, bang_bang=%d", length, len(data), avail, self._bang_bang)
 
-        #if self.counter % 50 == 0:
-        frame_size = frame_size + 1 # consume samples a little faster to avoid latency akkumulation
+            if length > 0:
+                if self._first_read:
+                    expected_mono = self._periodsize * 2
+                    expected_stereo = self._periodsize * 2 * 2
+                    # self._hw_channels = 2 if len(data) == expected_stereo else 1
+                    self._hw_channels = self._actual_channels
+                    logging.info("PyALSA first read: bytes=%d detected_hw_channels=%d requested_channels=%d", 
+                                len(data), self._hw_channels, self._actual_channels)
+                    self._first_read = False
+                
+                if self._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()
+                
+                actual_samples = len(data) // (2 * self._actual_channels)
+                ratio = frame_size / actual_samples
+                pcm_f32 = np.frombuffer(data, dtype=np.int16).astype(np.float32) / 32768.0
+                if self._actual_channels > 1:
+                    pcm_f32 = pcm_f32.reshape(-1, self._actual_channels)
+                resampled = self._resampler.process(pcm_f32, ratio, end_of_input=False)
+                if self._actual_channels > 1:
+                    resampled = resampled.reshape(-1)
+                self._resampler_buffer = np.concatenate([self._resampler_buffer, resampled])
+            else:
+                logging.warning("PyALSA: No data read from ALSA")
+                self._resampler_buffer = np.concatenate([
+                    self._resampler_buffer,
+                    np.zeros(frame_size * self._actual_channels, dtype=np.float32),
+                ])
+        except alsaaudio.ALSAAudioError as e:
+            logging.error("PyALSA: ALSA read error: %s", e)
+            self._resampler_buffer = np.concatenate([
+                self._resampler_buffer,
+                np.zeros(frame_size * self._actual_channels, dtype=np.float32),
+            ])
+        except Exception as e:
+            logging.error("PyALSA: Unexpected error in _read: %s", e, exc_info=True)
+            self._resampler_buffer = np.concatenate([
+                self._resampler_buffer,
+                np.zeros(frame_size * self._actual_channels, dtype=np.float32),
+            ])
 
-        pcm_buffer, overflowed = self._stream.read(frame_size)
-        if overflowed:
-            logging.warning("SoundDeviceAudioInput: overflowed")
+        needed = frame_size * self._actual_channels
+        if len(self._resampler_buffer) < needed:
+            pad = np.zeros(needed - len(self._resampler_buffer), dtype=np.float32)
+            self._resampler_buffer = np.concatenate([self._resampler_buffer, pad])
+            logging.debug("PyALSA: padded buffer with %d samples", needed - len(self._resampler_buffer))
 
-        n_available = self._stream.read_available
+        output = self._resampler_buffer[:needed]
+        self._resampler_buffer = self._resampler_buffer[needed:]
 
-        # 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")
-            
-        #     out = bytes(pcm_buffer) +  bytes(pcm_extra)
-        # else:
-        out = bytes(pcm_buffer)     
+        logging.debug("PyALSA: resampler_buffer remaining=%d", len(self._resampler_buffer))
+        return np.clip(output * 32767.0, -32768, 32767).astype(np.int16).tobytes()
 
-        self.max_avail = max(self.max_avail, n_available)
+  
 
-        #Diagnostics 
-        #with open(self.logfile_name, "a", encoding="utf-8") as f:
-        #    f.write(f"{n_available}, {adapt}, {round(self._runavg, 2)}, {overflowed}\n")
+    def _close(self) -> None:
+        if self._pcm:
+            self._pcm.close()
+            self._pcm = None
 
-        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):
@@ -538,7 +589,7 @@ async def init_broadcast(
 
     def on_flow():
         data_packet_queue = iso_queue.data_packet_queue
-        print(
+        logging.info(
             f'\rPACKETS: pending={data_packet_queue.pending}, '
             f'queued={data_packet_queue.queued}, '
             f'completed={data_packet_queue.completed}',
@@ -811,13 +862,6 @@ class Streamer():
                 big['encoder'] = encoders[0]
                 big['precoded'] = False
 
-            # Pre-compute software gain multiplier from dB config (0 dB = 1.0 = no change)
-            gain_db = getattr(big_config[i], 'input_gain_db', 0.0)
-            gain_db = max(0.0, min(20.0, float(gain_db)))
-            big['_gain_linear'] = 10.0 ** (gain_db / 20.0) if gain_db > 0 else 0.0
-            if big['_gain_linear'] > 0.0:
-                logging.info("Software gain for BIG %d: +%.1f dB (linear %.3f)", i, gain_db, big['_gain_linear'])
-
         logging.info("Streaming audio...")
         bigs = self.bigs
         self.is_streaming = True
@@ -865,14 +909,6 @@ class Streamer():
                         stream_finished[i] = True
                         continue
 
-                    # Apply software gain boost if configured (> 0 dB)
-                    gain_lin = big.get('_gain_linear', 0.0)
-                    if gain_lin > 0.0:
-                        pcm_arr = np.frombuffer(pcm_frame, dtype=np.int16).astype(np.float32)
-                        pcm_arr *= gain_lin
-                        np.clip(pcm_arr, -32768, 32767, out=pcm_arr)
-                        pcm_frame = pcm_arr.astype(np.int16).tobytes()
-
                     # Compute RMS audio level (normalized 0.0-1.0) for level monitoring
                     pcm_samples = np.frombuffer(pcm_frame, dtype=np.int16).astype(np.float32)
                     rms = np.sqrt(np.mean(pcm_samples ** 2)) / 32768.0 if len(pcm_samples) > 0 else 0.0

src/auracast/multicast_control.py

@@ -143,6 +143,10 @@ async def main():
         level=os.environ.get('LOG_LEVEL', logging.DEBUG),
         format='%(module)s.py:%(lineno)d %(levelname)s: %(message)s'
     )
+    
+    # Enable debug logging for bumble
+    # logging.getLogger('bumble').setLevel(logging.DEBUG)
+    
     os.chdir(os.path.dirname(__file__))
 
     global_conf = auracast_config.AuracastGlobalConfig(

src/auracast/server/multicast_frontend.py

@@ -198,35 +198,64 @@ else:
     start_stream, stop_stream = render_stream_controls(is_streaming, "Start Auracast", "Stop Auracast", running_mode, secondary_is_streaming)
 
 # Analog gain control (only for Analog mode, placed below start button)
-analog_gain_value = 50  # default (ALSA 10-60 range)
-software_boost_db = 0  # default
+analog_gain_db_left = 0  # default (dB)
+analog_gain_db_right = 0  # default (dB)
 if audio_mode == "Analog":
-    saved_analog_gain = saved_settings.get('analog_gain', 50)
-    # Convert persisted ALSA value (10-60) to display value (0-100)
-    saved_display = int(round((saved_analog_gain - 10) * 100 / 50))
-    saved_display = max(0, min(100, saved_display))
-    analog_gain_display = st.slider(
-        "Analog Input Gain",
-        min_value=0,
-        max_value=100,
-        value=saved_display,
-        step=5,
-        disabled=is_streaming,
-        format="%d%%",
-        help="ADC gain level for both analog inputs. Default is 80%."
-    )
-    # Map display value (0-100) back to ALSA range (10-60)
-    analog_gain_value = int(round(10 + analog_gain_display * 50 / 100))
-    saved_boost = saved_settings.get('software_boost_db', 0)
-    software_boost_db = st.slider(
-        "Boost",
-        min_value=0,
-        max_value=20,
-        value=min(int(saved_boost), 20),
-        step=1,
-        disabled=is_streaming,
-        help="Digital gain boost applied before encoding (0-20 dB). Use this when the line-level signal is too quiet even at max ADC gain. Higher values may cause clipping on loud signals."
+    if '_analog_gain_db_left' not in st.session_state:
+        st.session_state['_analog_gain_db_left'] = max(-12, min(18, int(saved_settings.get('analog_gain_db_left', 0))))
+    if '_analog_gain_db_right' not in st.session_state:
+        st.session_state['_analog_gain_db_right'] = max(-12, min(18, int(saved_settings.get('analog_gain_db_right', 0))))
+    if '_gain_link_channels' not in st.session_state:
+        st.session_state['_gain_link_channels'] = True
+
+    link_channels = st.checkbox(
+        "Link audio channel gain",
+        key='_gain_link_channels',
+        help="When enabled, Ch 2 mirrors Ch 1."
     )
+    _gain_col1, _gain_col2 = st.columns(2)
+    with _gain_col1:
+        analog_gain_db_left = st.slider(
+            "Ch 1 Input Gain",
+            min_value=-12,
+            max_value=18,
+            key='_analog_gain_db_left',
+            step=1,
+            format="%d dB",
+            help="ADC gain for channel 1 (-12 to 18 dB). Default is 0 dB."
+        )
+    with _gain_col2:
+        if link_channels:
+            st.session_state['_analog_gain_db_right'] = analog_gain_db_left
+        elif st.session_state.get('_prev_gain_link_channels', True):
+            # Transition: just unlinked — seed Ch 2 from Ch 1 so it doesn't jump to min
+            st.session_state['_analog_gain_db_right'] = analog_gain_db_left
+        st.session_state['_prev_gain_link_channels'] = link_channels
+        analog_gain_db_right = st.slider(
+            "Ch 2 Input Gain",
+            min_value=-12,
+            max_value=18,
+            key='_analog_gain_db_right',
+            step=1,
+            format="%d dB",
+            disabled=link_channels,
+            help="Uncheck 'Link audio channel gain' to adjust Ch 2 independently." if link_channels else "ADC gain for channel 2 (-12 to 18 dB). Default is 0 dB."
+        )
+    # Apply gain live while streaming whenever either slider value changes
+    if is_streaming:
+        prev_left = st.session_state.get('_prev_analog_gain_db_left')
+        prev_right = st.session_state.get('_prev_analog_gain_db_right')
+        if prev_left != analog_gain_db_left or prev_right != analog_gain_db_right:
+            try:
+                requests.post(
+                    f"{BACKEND_URL}/adc_gain",
+                    json={"gain_db_left": analog_gain_db_left, "gain_db_right": analog_gain_db_right},
+                    timeout=1,
+                )
+            except Exception:
+                pass
+        st.session_state['_prev_analog_gain_db_left'] = analog_gain_db_left
+        st.session_state['_prev_analog_gain_db_right'] = analog_gain_db_right
 
 # Audio level monitor (checkbox, not persisted across reloads)
 show_level_monitor = st.checkbox("Audio level monitor", value=False, disabled=not is_streaming,
@@ -684,8 +713,8 @@ else:
                     'immediate_rendering': immediate_rendering2,
                     'presentation_delay_ms': presentation_delay_ms2,
                     'qos_preset': qos_preset2,
-                    'analog_gain': analog_gain_value,
-                    'software_boost_db': software_boost_db,
+                    'analog_gain_db_left': analog_gain_db_left,
+                    'analog_gain_db_right': analog_gain_db_right,
                 }
 
         radio1_cfg = {
@@ -701,8 +730,8 @@ else:
             'presentation_delay_ms': presentation_delay_ms1,
             'qos_preset': qos_preset1,
             'stereo_mode': stereo_enabled,
-            'analog_gain': analog_gain_value,
-            'software_boost_db': software_boost_db,
+            'analog_gain_db_left': analog_gain_db_left,
+            'analog_gain_db_right': analog_gain_db_right,
         }
 
     if audio_mode == "Network - Dante":
@@ -1585,10 +1614,10 @@ if start_stream:
                 immediate_rendering=bool(cfg['immediate_rendering']),
                 presentation_delay_us=int(cfg['presentation_delay_ms'] * 1000),
                 qos_config=QOS_PRESET_MAP[cfg['qos_preset']],
-                analog_gain=cfg.get('analog_gain', 50),
+                analog_gain_db_left=cfg.get('analog_gain_db_left', 0.0),
+                analog_gain_db_right=cfg.get('analog_gain_db_right', 0.0),
                 bigs=[
                     auracast_config.AuracastBigConfig(
-                        id=cfg.get('id', 123456),
                         code=(cfg['stream_passwort'].strip() or None),
                         name=cfg['name'],
                         program_info=cfg['program_info'],
@@ -1599,7 +1628,6 @@ if start_stream:
                         sampling_frequency=q['rate'],
                         octets_per_frame=q['octets'],
                         num_bis=channels,  # 1=mono, 2=stereo - this determines the behavior
-                        input_gain_db=float(cfg.get('software_boost_db', 0)),
                     )
                 ],
             )
@@ -1643,8 +1671,6 @@ if start_stream:
                 if not stream.get('input_device'):
                     continue
                 
-                stream_id = radio_id * 1000 + i + 1  # Unique ID per stream
-                
                 # Check if this specific stream uses stereo (dante_stereo_X_Y device)
                 input_device = stream['input_device']
                 stream_is_stereo = is_stereo_mode or input_device.startswith('dante_stereo_')
@@ -1652,7 +1678,6 @@ if start_stream:
                 num_channels = 2 if stream_is_stereo else 1
                 
                 bigs.append(auracast_config.AuracastBigConfig(
-                    id=stream_id,
                     code=(stream.get('stream_password', '').strip() or None),
                     name=stream['name'],
                     program_info=stream['program_info'],
@@ -1755,7 +1780,21 @@ if is_started or is_stopped:
 # System expander (collapsed)
 ############################
 with st.expander("System control", expanded=False):
-    
+
+    st.subheader("Status LED")
+    led_enabled_current = bool(saved_settings.get("led_enabled", True))
+    led_enabled = st.checkbox(
+        "Blue LED on while transmitting",
+        value=led_enabled_current,
+        help="When enabled, the blue LED on GPIO pin 12 lights up while the stream is active."
+    )
+    if led_enabled != led_enabled_current:
+        try:
+            requests.post(f"{BACKEND_URL}/set_led_enabled", json={"led_enabled": led_enabled}, timeout=2)
+        except Exception as e:
+            st.error(f"Failed to update LED setting: {e}")
+        st.rerun()
+
     st.subheader("System temperatures")
     temp_col1, temp_col2, temp_col3 = st.columns([1, 1, 1])
     with temp_col1:

src/auracast/server/multicast_server.py

@@ -3,6 +3,7 @@ TODO: in the future the multicaster objects should run in their own threads or e
 
 """
 import os
+import re
 import logging as log
 import json
 from datetime import datetime
@@ -26,6 +27,57 @@ from auracast.utils.sounddevice_utils import (
 )
 
 load_dotenv()
+
+# Blue LED on GPIO pin 12 (BCM) – turns on while transmitting
+LED_PIN = 12
+try:
+    import RPi.GPIO as _GPIO
+    _GPIO.setmode(_GPIO.BCM)
+    _GPIO.setup(LED_PIN, _GPIO.OUT)
+    _GPIO_AVAILABLE = True
+except Exception:
+    _GPIO_AVAILABLE = False
+    _GPIO = None  # type: ignore
+
+_LED_ENABLED: bool = True  # toggled via /set_led_enabled
+_LED_SETTINGS_FILE = os.path.join(os.path.dirname(__file__), 'led_settings.json')
+
+def _load_led_settings() -> None:
+    global _LED_ENABLED
+    try:
+        if os.path.exists(_LED_SETTINGS_FILE):
+            with open(_LED_SETTINGS_FILE, 'r', encoding='utf-8') as f:
+                data = json.load(f)
+            _LED_ENABLED = bool(data.get('led_enabled', True))
+    except Exception:
+        _LED_ENABLED = True
+
+def _save_led_settings() -> None:
+    try:
+        os.makedirs(os.path.dirname(_LED_SETTINGS_FILE), exist_ok=True)
+        with open(_LED_SETTINGS_FILE, 'w', encoding='utf-8') as f:
+            json.dump({'led_enabled': _LED_ENABLED}, f)
+    except Exception:
+        pass
+
+def _led_on():
+    if _GPIO_AVAILABLE and _LED_ENABLED:
+        try:
+            _GPIO.output(LED_PIN, _GPIO.LOW)
+        except Exception:
+            pass
+
+def _led_off():
+    if _GPIO_AVAILABLE:
+        try:
+            _GPIO.output(LED_PIN, _GPIO.HIGH)
+        except Exception:
+            pass
+
+
+# Configure bumble debug logging
+# log.getLogger('bumble').setLevel(log.DEBUG)
+
 # make sure pipewire sets latency
 # Primary and secondary persisted settings files
 STREAM_SETTINGS_FILE1 = os.path.join(os.path.dirname(__file__), 'stream_settings.json')
@@ -133,6 +185,10 @@ def save_settings(persisted: dict, secondary: bool = False) -> None:
 def gen_random_add() -> str:
     return ':'.join(['%02X' % random.randint(0, 255) for _ in range(6)])
 
+def gen_random_broadcast_id() -> int:
+    """Generate a random 24-bit Broadcast ID (1..0xFFFFFF)."""
+    return random.randint(1, 0xFFFFFF)
+
 app = FastAPI()
 
 # Allow CORS for frontend on localhost
@@ -238,16 +294,18 @@ async def _init_i2c_on_startup() -> None:
             log.warning("Exception running i2cget (%s): %s", " ".join(read_cmd), e, exc_info=True)
 
 
-async def _set_adc_level(gain_percent: int = 50) -> None:
-    """Set ADC mixer level.
+async def _set_adc_level(gain_db_left: float = 0.0, gain_db_right: float = 0.0) -> None:
+    """Set ADC mixer gain in dB for left and right channels independently.
 
-    Runs: amixer -c 1 set 'ADC' x%
+    Runs: amixer -c 2 sset ADC {gain_db_left}dB,{gain_db_right}dB
     
     Args:
-        gain_percent: Gain level 10-60, default 50 (above 60% causes clipping/silence)
+        gain_db_left: Left channel gain in dB (-12 to 18), default 0
+        gain_db_right: Right channel gain in dB (-12 to 18), default 0
     """
-    gain_percent = max(10, min(60, gain_percent))
-    cmd = ["amixer", "-c", "1", "set", "ADC", f"{gain_percent}%"]
+    gain_db_left = max(-12.0, min(18.0, gain_db_left))
+    gain_db_right = max(-12.0, min(18.0, gain_db_right))
+    cmd = ["amixer", "-c", "2", "sset", "ADC", "--", f"{int(gain_db_left)}dB,{int(gain_db_right)}dB"]
     try:
         proc = await asyncio.create_subprocess_exec(
             *cmd,
@@ -256,15 +314,46 @@ async def _set_adc_level(gain_percent: int = 50) -> None:
         )
         stdout, stderr = await proc.communicate()
         if proc.returncode != 0:
-            log.warning(
+            log.error(
                 "amixer ADC level command failed (rc=%s): %s",
                 proc.returncode,
                 (stderr or b"" ).decode(errors="ignore").strip(),
             )
         else:
             log.info("amixer ADC level set successfully: %s", (stdout or b"" ).decode(errors="ignore").strip())
+            read_proc = await asyncio.create_subprocess_exec(
+                "amixer", "-c", "2", "sget", "ADC",
+                stdout=asyncio.subprocess.PIPE,
+                stderr=asyncio.subprocess.PIPE,
+            )
+            read_stdout, read_stderr = await read_proc.communicate()
+            if read_proc.returncode != 0:
+                log.error(
+                    "amixer ADC sget failed (rc=%s): %s",
+                    read_proc.returncode,
+                    (read_stderr or b"").decode(errors="ignore").strip(),
+                )
+            else:
+                sget_output = (read_stdout or b"").decode(errors="ignore")
+                actual = {}
+                for line in sget_output.splitlines():
+                    for ch_key, ch_name in (("left", "Front Left"), ("right", "Front Right")):
+                        if ch_name in line:
+                            m = re.search(r'\[(-?\d+(?:\.\d+)?)dB\]', line)
+                            if m:
+                                actual[ch_key] = round(float(m.group(1)))
+                expected_left = int(gain_db_left)
+                expected_right = int(gain_db_right)
+                if actual.get("left") != expected_left or actual.get("right") != expected_right:
+                    mismatch = (
+                        f"ADC level mismatch after set: expected L={expected_left}dB R={expected_right}dB, "
+                        f"got L={actual.get('left')}dB R={actual.get('right')}dB"
+                    )
+                    log.error(mismatch)
+                else:
+                    log.info("ADC level set successfully: L=%sdB R=%sdB", expected_left, expected_right)
     except Exception as e:
-        log.warning("Exception running amixer ADC level command: %s", e, exc_info=True)
+        log.error("Exception running amixer ADC level command: %s", e, exc_info=True)
 
 
 async def _stop_all() -> bool:
@@ -284,6 +373,7 @@ async def _stop_all() -> bool:
             was_running = True
         finally:
             multicaster2 = None
+    _led_off()
     return was_running
 
 async def _status_primary() -> dict:
@@ -343,8 +433,9 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
             if input_device_name in ('ch1', 'ch2'):
                 audio_mode_persist = 'Analog'
                 # Set ADC gain level for analog mode
-                analog_gain = getattr(conf, 'analog_gain', 50)
-                await _set_adc_level(analog_gain)
+                analog_gain_db_left = getattr(conf, 'analog_gain_db_left', 0.0)
+                analog_gain_db_right = getattr(conf, 'analog_gain_db_right', 0.0)
+                await _set_adc_level(analog_gain_db_left, analog_gain_db_right)
             elif input_device_name in dante_channels:
                 audio_mode_persist = 'Network - Dante'
             else:
@@ -387,18 +478,15 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
                     if is_stereo and sel == 'ch1':
                         # Stereo mode: use ALSA directly to capture both channels from hardware
                         # ch1=left (channel 0), ch2=right (channel 1)
-                        big.audio_source = 'alsa:hw:CARD=i2s,DEV=0'
+                        big.audio_source = 'device:hw:2'
                         big.input_format = f"int16le,{hardware_capture_rate},2"
                         log.info("Configured analog stereo input: using ALSA hw:CARD=i2s,DEV=0 with ch1=left, ch2=right")
                     elif is_stereo and sel == 'ch2':
                         # Skip ch2 in stereo mode as it's already captured as part of stereo pair
                         continue
                     else:
-                        # Mono mode: individual channel capture
-                        device_index = resolve_input_device_index(sel)
-                        if device_index is None:
-                            raise HTTPException(status_code=400, detail=f"Audio device '{sel}' not found.")
-                        big.audio_source = f'device:{device_index}'
+                        # Mono mode: use dsnoop virtual device directly (ch1=left, ch2=right)
+                        big.audio_source = f'device:{sel}'
                         big.input_format = f"int16le,{hardware_capture_rate},1"
                     continue
 
@@ -454,10 +542,12 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
 
         conf.qos_config.max_transport_latency_ms = int(conf.qos_config.number_of_retransmissions) * 10 + 3
 
-        # Only generate a new random_address if the BIG is still at the model default.
+        # Generate fresh random_address and broadcast ID for any BIG still at model defaults.
         for big in conf.bigs:
             if not getattr(big, 'random_address', None) or big.random_address == DEFAULT_RANDOM_ADDRESS:
                 big.random_address = gen_random_add()
+            if big.id == DEFAULT_BIG_ID:
+                big.id = gen_random_broadcast_id()
 
         # Log the final, fully-updated configuration just before creating the Multicaster
         log.info('Final multicaster config (transport=%s):\n %s', transport, conf.model_dump_json(indent=2))
@@ -468,6 +558,7 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
         auto_started = False
         if any(isinstance(big.audio_source, str) and (big.audio_source.startswith("device:") or big.audio_source.startswith("alsa:") or big.audio_source.startswith("file:")) for big in conf.bigs):
             await mc.start_streaming()
+            _led_on()
             auto_started = True
 
         demo_count = sum(1 for big in conf.bigs if isinstance(big.audio_source, str) and big.audio_source.startswith('file:'))
@@ -492,8 +583,8 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
             'immediate_rendering': getattr(conf, 'immediate_rendering', False),
             'assisted_listening_stream': getattr(conf, 'assisted_listening_stream', False),
             'analog_stereo_mode': getattr(conf.bigs[0], 'analog_stereo_mode', False) if conf.bigs else False,
-            'analog_gain': getattr(conf, 'analog_gain', 50),
-            'software_boost_db': getattr(conf.bigs[0], 'input_gain_db', 0.0) if conf.bigs else 0.0,
+            'analog_gain_db_left': getattr(conf, 'analog_gain_db_left', 0.0),
+            'analog_gain_db_right': getattr(conf, 'analog_gain_db_right', 0.0),
             'stream_password': (conf.bigs[0].code if conf.bigs and getattr(conf.bigs[0], 'code', None) else None),
             'big_ids': [getattr(big, 'id', DEFAULT_BIG_ID) for big in conf.bigs],
             'big_random_addresses': [getattr(big, 'random_address', DEFAULT_RANDOM_ADDRESS) for big in conf.bigs],
@@ -528,6 +619,16 @@ async def initialize2(conf: auracast_config.AuracastConfigGroup):
         multicaster2 = mc
         save_settings(persisted, secondary=True)
 
+@app.post("/set_led_enabled")
+async def set_led_enabled(body: dict):
+    """Enable or disable the blue status LED. Persisted across restarts."""
+    global _LED_ENABLED
+    _LED_ENABLED = bool(body.get("led_enabled", True))
+    _save_led_settings()
+    if not _LED_ENABLED:
+        _led_off()
+    return {"led_enabled": _LED_ENABLED}
+
 @app.post("/stop_audio")
 async def stop_audio():
     """Stops streaming on both multicaster1 and multicaster2 (worker thread)."""
@@ -556,6 +657,28 @@ async def stop_audio():
         log.error("Exception in /stop_audio: %s", traceback.format_exc())
         raise HTTPException(status_code=500, detail=str(e))
 
+@app.post("/adc_gain")
+async def set_adc_gain(payload: dict):
+    """Set ADC gain in dB for left and right channels without restarting the stream.
+
+    Body: {"gain_db_left": float, "gain_db_right": float}
+    """
+    try:
+        gain_db_left = float(payload.get("gain_db_left", 0.0))
+        gain_db_right = float(payload.get("gain_db_right", 0.0))
+        await _set_adc_level(gain_db_left, gain_db_right)
+        # Persist the new values so they survive a restart
+        for load_fn, save_fn in [(load_stream_settings, save_stream_settings), (load_stream_settings2, save_stream_settings2)]:
+            s = load_fn() or {}
+            if s:
+                s['analog_gain_db_left'] = gain_db_left
+                s['analog_gain_db_right'] = gain_db_right
+                save_fn(s)
+        return {"status": "ok", "gain_db_left": gain_db_left, "gain_db_right": gain_db_right}
+    except Exception as e:
+        log.error("Exception in /adc_gain: %s", traceback.format_exc())
+        raise HTTPException(status_code=500, detail=str(e))
+
 @app.post("/stream_lc3")
 async def send_audio(audio_data: dict[str, str]):
     """Sends a block of pre-coded LC3 audio via the worker."""
@@ -591,6 +714,7 @@ async def get_status():
     secondary.update(secondary_persisted)
     status["secondary"] = secondary
     status["secondary_is_streaming"] = bool(secondary.get("is_streaming", False))
+    status["led_enabled"] = _LED_ENABLED
 
     return status
 
@@ -751,7 +875,6 @@ async def _autostart_from_settings():
                         iso_que_len=1,
                         sampling_frequency=rate,
                         octets_per_frame=octets,
-                        input_gain_db=float(settings.get('software_boost_db', 0)),
                     )
                 ]
                 conf = auracast_config.AuracastConfigGroup(
@@ -761,7 +884,8 @@ async def _autostart_from_settings():
                     immediate_rendering=immediate_rendering,
                     assisted_listening_stream=assisted_listening_stream,
                     presentation_delay_us=pres_delay if pres_delay is not None else 40000,
-                    analog_gain=settings.get('analog_gain', 50),
+                    analog_gain_db_left=settings.get('analog_gain_db_left', 0.0),
+                    analog_gain_db_right=settings.get('analog_gain_db_right', 0.0),
                     bigs=bigs,
                 )
                 # Set num_bis for stereo mode if needed
@@ -906,7 +1030,6 @@ async def _autostart_from_settings():
                         iso_que_len=1,
                         sampling_frequency=rate,
                         octets_per_frame=octets,
-                        input_gain_db=float(settings.get('software_boost_db', 0)),
                     )
                 ]
                 conf = auracast_config.AuracastConfigGroup(
@@ -916,7 +1039,8 @@ async def _autostart_from_settings():
                     immediate_rendering=immediate_rendering,
                     assisted_listening_stream=assisted_listening_stream,
                     presentation_delay_us=pres_delay if pres_delay is not None else 40000,
-                    analog_gain=settings.get('analog_gain', 50),
+                    analog_gain_db_left=settings.get('analog_gain_db_left', 0.0),
+                    analog_gain_db_right=settings.get('analog_gain_db_right', 0.0),
                     bigs=bigs,
                 )
                 conf.qos_config = QOS_PRESET_MAP.get(saved_qos_preset, QOS_PRESET_MAP["Fast"])
@@ -938,9 +1062,10 @@ async def _autostart_from_settings():
 async def _startup_autostart_event():
     # Spawn the autostart task without blocking startup
     log.info("[STARTUP] Auracast multicast server startup: initializing settings cache, I2C, and PipeWire cache")
+    _led_off()
     
     # Run install_asoundconf.sh script
-    script_path = os.path.join(os.path.dirname(__file__), '..', 'misc', 'install_asoundconf.sh')
+    script_path = os.path.join(os.path.dirname(__file__), '..', '..', 'misc', 'install_asoundconf.sh')
     try:
         log.info("[STARTUP] Running install_asoundconf.sh script")
         result = subprocess.run(['bash', script_path], capture_output=True, text=True, check=True)
@@ -951,10 +1076,11 @@ async def _startup_autostart_event():
         log.error(f"[STARTUP] Error running install_asoundconf.sh: {str(e)}")
     
     # Hydrate settings cache once to avoid disk I/O during /status
+    _load_led_settings()
     _init_settings_cache_from_disk()
     await _init_i2c_on_startup()
-    # Ensure ADC mixer level is set at startup (default 50%)
-    await _set_adc_level(50)
+    # Ensure ADC mixer level is set at startup (default 0 dB)
+    await _set_adc_level(0.0, 0.0)
     refresh_pw_cache()
     log.info("[STARTUP] Scheduling autostart task")
     asyncio.create_task(_autostart_from_settings())
@@ -1245,26 +1371,12 @@ async def system_update():
             log.error("git checkout failed: %s", stderr.decode())
             raise HTTPException(status_code=500, detail=f"git checkout failed: {stderr.decode()}")
 
-        # 2. Run poetry install (use full path as poetry is in user's ~/.local/bin)
-        poetry_path = os.path.expanduser("~/.local/bin/poetry")
-        proc = await asyncio.create_subprocess_exec(
-            poetry_path, "install",
-            cwd=project_root,
-            stdout=asyncio.subprocess.PIPE,
-            stderr=asyncio.subprocess.PIPE
-        )
-        stdout, stderr = await proc.communicate()
-        if proc.returncode != 0:
-            log.error("poetry install failed: %s", stderr.decode())
-            raise HTTPException(status_code=500, detail=f"poetry install failed: {stderr.decode()}")
-
-        # 3. Restart services via the update script
-        update_script = os.path.join(project_root, 'src', 'service', 'update_and_run_server_and_frontend.sh')
-        proc = await asyncio.create_subprocess_exec(
+        # 2. Hand off remaining work to the (now-updated) system_update.sh script
+        update_script = os.path.join(os.path.dirname(__file__), 'system_update.sh')
+        log.info("Handing off to system_update.sh...")
+        await asyncio.create_subprocess_exec(
             "bash", update_script,
             cwd=project_root,
-            stdout=asyncio.subprocess.PIPE,
-            stderr=asyncio.subprocess.PIPE
         )
         # Don't wait for completion as we'll be restarted
         await asyncio.sleep(0.5)

src/auracast/server/system_update.sh

@@ -0,0 +1,90 @@
+#!/bin/bash
+# system_update.sh - Runs after git checkout in the Python system_update endpoint.
+# Called with the current working directory = project root.
+# All output is also written to /tmp/system_update.log for debugging.
+
+exec > >(tee -a /tmp/system_update.log) 2>&1
+
+SCRIPT_DIR="$(cd "$(dirname "${BASH_SOURCE[0]}")" && pwd)"
+PROJECT_ROOT="$(cd "$SCRIPT_DIR/../../.." && pwd)"
+POETRY="$HOME/.local/bin/poetry"
+OPENOCD_SRC="$HOME/sw_openocd"
+OPENOCD_REPO="ssh://git@gitea.summitwave.work:222/auracaster/sw_openocd.git"
+OPENOCD_BRANCH="change-8818"
+OPENOCD_MARKER="$OPENOCD_SRC/.last_built_commit"
+OPENOCD_DIR="$PROJECT_ROOT/src/openocd"
+
+echo "[system_update] Starting post-checkout update. project_root=$PROJECT_ROOT"
+
+# 1. poetry install
+echo "[system_update] Running poetry install..."
+(cd "$PROJECT_ROOT" && "$POETRY" install)
+if [ $? -ne 0 ]; then
+    echo "[system_update] ERROR: poetry install failed"
+    exit 1
+fi
+
+# 2. Clone/update and build sw_openocd if needed
+if [ ! -d "$OPENOCD_SRC" ]; then
+    echo "[system_update] Installing sw_openocd build dependencies..."
+    sudo apt install -y git build-essential libtool autoconf texinfo \
+        libusb-1.0-0-dev libftdi1-dev libhidapi-dev pkg-config || \
+        echo "[system_update] WARNING: apt install deps had errors, continuing"
+    sudo apt-get install -y pkg-config libjim-dev || \
+        echo "[system_update] WARNING: apt-get install libjim-dev had errors, continuing"
+
+    echo "[system_update] Cloning sw_openocd branch $OPENOCD_BRANCH..."
+    git clone --branch "$OPENOCD_BRANCH" --single-branch "$OPENOCD_REPO" "$OPENOCD_SRC"
+    if [ $? -ne 0 ]; then
+        echo "[system_update] ERROR: git clone sw_openocd failed"
+        exit 1
+    fi
+else
+    echo "[system_update] Updating sw_openocd..."
+    git -C "$OPENOCD_SRC" fetch origin "$OPENOCD_BRANCH"
+    git -C "$OPENOCD_SRC" checkout "$OPENOCD_BRANCH"
+    git -C "$OPENOCD_SRC" pull
+fi
+
+OPENOCD_COMMIT=$(git -C "$OPENOCD_SRC" rev-parse HEAD)
+LAST_BUILT=""
+[ -f "$OPENOCD_MARKER" ] && LAST_BUILT=$(cat "$OPENOCD_MARKER")
+
+if [ "$OPENOCD_COMMIT" != "$LAST_BUILT" ]; then
+    echo "[system_update] Building sw_openocd (commit $OPENOCD_COMMIT)..."
+
+    (cd "$OPENOCD_SRC" && ./bootstrap)
+    if [ $? -ne 0 ]; then echo "[system_update] ERROR: openocd bootstrap failed"; exit 1; fi
+
+    (cd "$OPENOCD_SRC" && ./configure --enable-bcm2835gpio --enable-sysfsgpio)
+    if [ $? -ne 0 ]; then echo "[system_update] ERROR: openocd configure failed"; exit 1; fi
+
+    (cd "$OPENOCD_SRC" && make)
+    if [ $? -ne 0 ]; then echo "[system_update] ERROR: openocd make failed"; exit 1; fi
+
+    (cd "$OPENOCD_SRC" && sudo make install)
+    if [ $? -ne 0 ]; then echo "[system_update] ERROR: openocd make install failed"; exit 1; fi
+
+    echo "$OPENOCD_COMMIT" > "$OPENOCD_MARKER"
+    echo "[system_update] sw_openocd built and installed (commit $OPENOCD_COMMIT)"
+else
+    echo "[system_update] sw_openocd up to date (commit $OPENOCD_COMMIT), skipping build"
+fi
+
+# 3. Flash firmware to both SWD interfaces
+FLASH_SCRIPT="$OPENOCD_DIR/flash.sh"
+HEX_FILE="$OPENOCD_DIR/merged.hex"
+
+for IFACE in swd0 swd1; do
+    echo "[system_update] Flashing $IFACE..."
+    (cd "$OPENOCD_DIR" && bash "$FLASH_SCRIPT" -i "$IFACE" -f "$HEX_FILE")
+    if [ $? -ne 0 ]; then
+        echo "[system_update] ERROR: flash $IFACE failed"
+        exit 1
+    fi
+    echo "[system_update] Flash $IFACE complete"
+done
+
+# 4. Restart services (this will kill this process too)
+echo "[system_update] Restarting services..."
+bash "$PROJECT_ROOT/src/service/update_and_run_server_and_frontend.sh"

src/misc/asound.conf

@@ -6,8 +6,8 @@ pcm.ch1 {
         channels 2
         rate 48000
         format S16_LE
-        period_size 120    
-        buffer_size 240    
+        period_size 240    
+        buffer_size 960    
     }
     bindings.0  0
 }
@@ -21,8 +21,8 @@ pcm.ch2 {
         channels 2
         rate 48000
         format S16_LE
-        period_size 120
-        buffer_size 240
+        period_size 240
+        buffer_size 960
     }    
     bindings.0  1
 }

src/misc/install_asoundconf.sh

@@ -1 +1,2 @@
-sudo cp src/misc/asound.conf /etc/asound.conf
+SCRIPT_DIR="$(cd "$(dirname "$0")" && pwd)"
+sudo cp "$SCRIPT_DIR/asound.conf" /etc/asound.conf

src/openocd/flash.sh

@@ -0,0 +1,48 @@
+#!/bin/bash
+set -e
+
+INTERFACE="swd0"
+HEX_FILE=""
+
+usage() {
+  echo "Usage: $0 -f <hex_file> [-i swd0|swd1]"
+  exit 1
+}
+
+while getopts "f:i:h" opt; do
+  case "$opt" in
+    f) HEX_FILE="$OPTARG" ;;
+    i)
+      if [[ "$OPTARG" == "swd0" || "$OPTARG" == "swd1" ]]; then
+        INTERFACE="$OPTARG"
+      else
+        usage
+      fi
+      ;;
+    h) usage ;;
+    *) usage ;;
+  esac
+done
+
+[[ -n "$HEX_FILE" ]] || usage
+[[ -f "$HEX_FILE" ]] || { echo "HEX file not found: $HEX_FILE"; exit 1; }
+
+sudo openocd \
+  -f ./raspberrypi-${INTERFACE}.cfg \
+  -c "init" \
+  -c "reset init" \
+  -c "flash banks" \
+  -c "flash write_image $HEX_FILE" \
+  -c "verify_image $HEX_FILE" \
+  -c "reset run" \
+  -c "shutdown"
+
+sudo openocd \
+  -f ./raspberrypi-${INTERFACE}.cfg \
+  -c "init" \
+  -c "nrf54l.dap apreg 2 0x000 0x1" \
+  -c "sleep 100" \
+  -c "nrf54l.dap apreg 2 0x000 0x0" \
+  -c "shutdown"
+
+echo "Flashing complete."

src/openocd/raspberrypi-swd0.cfg

@@ -5,4 +5,9 @@ adapter gpio swdio 26
 #adapter gpio trst 26
 #reset_config trst_only
 
+
+source [find target/nordic/nrf54l.cfg]
+
+flash bank $_CHIPNAME.flash nrf54 0x00000000 0 0 0 $_TARGETNAME
+
 adapter speed 1000

src/openocd/raspberrypi-swd1.cfg

@@ -5,4 +5,9 @@ adapter gpio swdio 24
 #adapter gpio trst 27
 #reset_config trst_only
 
+
+source [find target/nordic/nrf54l.cfg]
+
+flash bank $_CHIPNAME.flash nrf54 0x00000000 0 0 0 $_TARGETNAME
+
 adapter speed 1000