Merge branch 'main' into bugfix/1087-UI-reset-refresh

Makes the activation of dante possible.
Fixes issues with local link for DANTE.
First implementation of audiopipeline in its own thread.
Prevents Frontend from interrupting the audio stream.

Openproject:
#1040
#1069
#652
#1041
#1063

Reviewed-on: #31

.gitignore

@@ -51,4 +51,11 @@ 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
+
+
+# Dante license files
+*.lic
+src/dep/dante_package/dante_data/activation/device.lic
+src/dep/dante_package/dante_data/activation/manufacturer.cert

AGENTS.md

@@ -4,8 +4,11 @@
 - this projects uses poetry for package management 
 - if something should be run in a python env use 'poetry run'
 
+# Environment
+- this application normally runs on an embedded linux on a cm4
 
 ## Application
 - this is a bluetooth Auracast transmitter application
+- if you add a new parameter for a stream make sure it is saved to the settings.json so it is persisted
 - it consists of multicast_frontend.py and multicast_server.py mainly which connect to each other via a rest api
-
+- after you implemented something the user will mainly test it and you should call the update_and_run_server_and_frontend.sh script if the server and frontend were already running.

poetry.lock

@@ -270,51 +270,6 @@ optional = false
 python-versions = ">=3.9"
 groups = ["main"]
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     {file = "av-14.4.0.tar.gz", hash = "sha256:3ecbf803a7fdf67229c0edada0830d6bfaea4d10bfb24f0c3f4e607cd1064b42"},
 ]
 
@@ -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

@@ -1,5 +1,11 @@
 from typing import List
-from pydantic import BaseModel
+from pydantic import BaseModel, field_validator
+
+# Discrete TX power levels (dBm) supported by the Nordic SoftDevice Controller
+# for the nRF radio PA. The HCI controller will clamp requested values to the
+# nearest supported step. The maximum is bounded by CONFIG_BT_CTLR_TX_PWR_*
+# in the hci_uart firmware (currently +8 dBm).
+TX_POWER_VALID = [8, 7, 6, 5, 4, 3, 2, 0, -4, -8, -12, -16, -20]
 
 # Define some base to hold the relevant parameters
 class AuracastQoSConfig(BaseModel):
@@ -28,13 +34,24 @@ class AuracastGlobalConfig(BaseModel):
     octets_per_frame: int = 40 #48kbps@24kHz # bitrate = octets_per_frame * 8 / frame len
     frame_duration_us: int = 10000
     presentation_delay_us: int = 40000
-    # TODO:pydantic does not support bytes serialization - use .hex and np.fromhex()
-    manufacturer_data: tuple[int, bytes] | tuple[None, None] = (None, None)
     # LE Audio: Broadcast Audio Immediate Rendering (metadata type 0x09)
     # When true, include a zero-length LTV with type 0x09 in the subgroup metadata
     # so receivers may render earlier than the presentation delay for lower latency.
     immediate_rendering: bool = False
     assisted_listening_stream: bool = False
+    # Bluetooth advertising TX power for this radio in dBm (per advertising set).
+    # Sent through HCI_LE_Set_Extended_Advertising_Parameters; the SDC clamps to
+    # nearest supported hardware step and propagates to primary/secondary adv,
+    # the periodic advertising train and the BIS ISO PDUs.
+    advertising_tx_power: int = 8
+
+    @field_validator('advertising_tx_power')
+    @classmethod
+    def _snap_tx_power(cls, v: int) -> int:
+        # Snap to the nearest supported discrete step in TX_POWER_VALID.
+        if v in TX_POWER_VALID:
+            return v
+        return min(TX_POWER_VALID, key=lambda s: abs(s - v))
 
 # "Audio input. "
 # "'device' -> use the host's default sound input device, "
@@ -62,7 +79,7 @@ class AuracastBigConfigDeu(AuracastBigConfig):
     name: str = 'Hörsaal A'
     language: str ='deu'
     program_info: str = 'Vorlesung DE'
-    audio_source: str = 'file:./testdata/wave_particle_5min_de.wav'
+    audio_source: str = 'file:./testdata/wave_particle_5min_de.lc3'
 
 class AuracastBigConfigEng(AuracastBigConfig):
     id: int = 123
@@ -70,7 +87,7 @@ class AuracastBigConfigEng(AuracastBigConfig):
     name: str  = 'Lecture Hall A'
     language: str ='eng'
     program_info: str = 'Lecture EN'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_en.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_en.lc3'
 
 class AuracastBigConfigFra(AuracastBigConfig):
     id: int = 1234
@@ -79,7 +96,7 @@ class AuracastBigConfigFra(AuracastBigConfig):
     name: str  = 'Auditoire A'
     language: str ='fra'
     program_info: str = 'Auditoire FR'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_fr.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_fr.lc3'
 
 class AuracastBigConfigSpa(AuracastBigConfig):
     id: int =12345
@@ -87,7 +104,7 @@ class AuracastBigConfigSpa(AuracastBigConfig):
     name: str  = 'Auditorio A'
     language: str ='spa'
     program_info: str = 'Auditorio ES'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_es.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_es.lc3'
 
 class AuracastBigConfigIta(AuracastBigConfig):
     id: int =1234567
@@ -95,7 +112,7 @@ class AuracastBigConfigIta(AuracastBigConfig):
     name: str  = 'Aula A'
     language: str ='ita'
     program_info: str = 'Aula IT'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_it.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_it.lc3'
 
 
 class AuracastBigConfigPol(AuracastBigConfig):
@@ -104,11 +121,12 @@ class AuracastBigConfigPol(AuracastBigConfig):
     name: str  = 'Sala Wykładowa'
     language: str ='pol'
     program_info: str = 'Sala Wykładowa PL'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_pl.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_pl.lc3'
 
 
 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
@@ -48,6 +49,7 @@ import bumble.transport
 import bumble.utils
 from bumble.device import Host, AdvertisingChannelMap
 from bumble.audio import io as audio_io
+from bumble.vendor.zephyr.hci import HCI_Write_Tx_Power_Level_Command
 
 from auracast import auracast_config
 from auracast.utils.read_lc3_file import read_lc3_file
@@ -56,7 +58,7 @@ 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 +141,146 @@ 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 = 80 
+            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):
@@ -411,21 +463,6 @@ async def init_broadcast(
             ],
         )
         logger.info('Setup Advertising')
-        advertising_manufacturer_data = (
-        b''
-        if global_config.manufacturer_data == (None, None)
-        else bytes(
-            core.AdvertisingData(
-                [
-                    (
-                        core.AdvertisingData.MANUFACTURER_SPECIFIC_DATA,
-                        struct.pack('<H', global_config.manufacturer_data[0])
-                        + global_config.manufacturer_data[1],
-                    )
-                ]
-            )
-        )
-    )
         bigs[f'big{i}']['broadcast_audio_announcement'] = bap.BroadcastAudioAnnouncement(conf.id)
 
         # Build advertising data types list
@@ -468,13 +505,22 @@ async def init_broadcast(
                 advertising_sid=i,
                 primary_advertising_phy=hci.Phy.LE_1M, # 2m phy config throws error - because for primary advertising channels, 1mbit is only supported
                 secondary_advertising_phy=hci.Phy.LE_1M, # this is the secondary advertising beeing send on non advertising channels (extendend advertising)
-                #advertising_tx_power= # tx power in dbm (max 20)
+                # Pass NO_PREFERENCE (0x7F) here for two reasons:
+                # 1. The Nordic SoftDevice Controller ignores this field for
+                #    advertising sets and always returns the compile-time
+                #    CONFIG_BT_CTLR_TX_PWR_* value. The real TX power is
+                #    applied via the Zephyr VS Write_Tx_Power_Level command
+                #    issued right after create_advertising_set() returns.
+                # 2. Bumble's HCI metadata declares this field as 1-byte
+                #    *unsigned* (a bumble bug — the BT spec defines it as
+                #    signed int8), so negative values would raise
+                #    "bytes must be in range(0, 256)" at serialization.
+                advertising_tx_power=hci.HCI_LE_Set_Extended_Advertising_Parameters_Command.TX_POWER_NO_PREFERENCE,
                 #secondary_advertising_max_skip=10,
             ),
             advertising_data=(
                 bigs[f'big{i}']['broadcast_audio_announcement'].get_advertising_data()
                 + bytes(core.AdvertisingData(advertising_data_types))
-                + advertising_manufacturer_data
             ),
             periodic_advertising_parameters=bumble.device.PeriodicAdvertisingParameters(
                 periodic_advertising_interval_min=80,
@@ -485,6 +531,48 @@ async def init_broadcast(
             auto_start=True,
         )
         bigs[f'big{i}']['advertising_set'] = advertising_set
+        # NOTE: selected_tx_power below reflects the SDC's compile-time max
+        # (LE_Set_Ext_Adv_Params was sent with NO_PREFERENCE). The actual
+        # transmit power is set by the VS Write_Tx_Power_Level call below.
+        logging.debug(
+            'LE_Set_Ext_Adv_Params reports controller fallback TX power: %+d dBm (handle=%d)',
+            getattr(advertising_set, 'selected_tx_power', 0),
+            i,
+        )
+
+        # The Nordic SoftDevice Controller does not honor the per-set
+        # advertising_tx_power passed in HCI_LE_Set_Extended_Advertising_Parameters
+        # (it returns the compile-time CONFIG_BT_CTLR_TX_PWR_* value regardless).
+        # Apply the requested level via the Zephyr Vendor-Specific HCI command
+        # Write_Tx_Power_Level (opcode 0xFC0E), which the SDC honors per
+        # advertising handle. The SDC clamps the value to the nearest supported
+        # hardware step (max bounded by CONFIG_BT_CTLR_TX_PWR_PLUS_8).
+        try:
+            adv_handle = getattr(advertising_set, 'advertising_handle', i)
+            response = await device.send_command(
+                HCI_Write_Tx_Power_Level_Command(
+                    handle_type=HCI_Write_Tx_Power_Level_Command.TX_POWER_HANDLE_TYPE_ADV,
+                    connection_handle=adv_handle,
+                    tx_power_level=global_config.advertising_tx_power,
+                )
+            )
+            rp = getattr(response, 'return_parameters', None)
+            status = getattr(rp, 'status', 0xFF) if rp is not None else 0xFF
+            selected = getattr(rp, 'selected_tx_power_level', None) if rp is not None else None
+            if status == 0 and selected is not None:
+                logging.info(
+                    'Advertising TX power (VS Write_Tx_Power_Level): requested=%+d dBm, controller selected=%+d dBm (handle=%d)',
+                    global_config.advertising_tx_power,
+                    selected,
+                    adv_handle,
+                )
+            else:
+                logging.warning(
+                    'VS Write_Tx_Power_Level failed: status=0x%02X handle=%d requested=%+d dBm',
+                    status, adv_handle, global_config.advertising_tx_power,
+                )
+        except Exception as e:
+            logging.warning('VS Write_Tx_Power_Level not supported by controller: %s', e)
 
         logging.info('Start Periodic Advertising')
         await advertising_set.start_periodic()
@@ -538,7 +626,7 @@ 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}',
@@ -551,6 +639,29 @@ async def init_broadcast(
     return bigs
 
 
+def _lc3_file_byte_gen(filename: str, loop: bool = False):
+    """Stream LC3 frames from disk as individual bytes, with optional looping.
+
+    Yields one byte (int) at a time so it is compatible with the existing
+    ``bytes(itertools.islice(gen, bytes_per_frame))`` consumer without loading
+    the whole file into memory.
+    """
+    while True:
+        with open(filename, 'rb') as f:
+            f.read(18)  # skip 18-byte LC3 header
+            while True:
+                size_b = f.read(2)
+                if len(size_b) < 2:
+                    break
+                frame_size = struct.unpack('=H', size_b)[0]
+                frame = f.read(frame_size)
+                if len(frame) < frame_size:
+                    break
+                yield from frame
+        if not loop:
+            return
+
+
 class Streamer():
     """
     Streamer class that supports multiple input formats. See bumble for streaming from wav or device
@@ -638,6 +749,12 @@ class Streamer():
         except Exception:
             pass
 
+    def get_audio_levels(self) -> list[float]:
+        """Return current RMS audio levels (0.0-1.0) for each BIG."""
+        if not self.bigs:
+            return []
+        return [big.get('_audio_level_rms', 0.0) for big in self.bigs.values()]
+
     async def stream(self):
 
         bigs = self.bigs
@@ -700,13 +817,7 @@ class Streamer():
                 big['precoded'] = True
                 big['lc3_bytes_per_frame'] = global_config.octets_per_frame
                 filename = big_config[i].audio_source.replace('file:', '')
-
-                lc3_bytes = read_lc3_file(filename)
-                lc3_frames = iter(lc3_bytes)
-
-                if big_config[i].loop:
-                    lc3_frames = itertools.cycle(lc3_frames)
-                big['lc3_frames'] = lc3_frames
+                big['lc3_frames'] = _lc3_file_byte_gen(filename, loop=big_config[i].loop)
 
             # use wav files and code them entirely before streaming
             elif big_config[i].precode_wav and big_config[i].audio_source.endswith('.wav'):
@@ -754,7 +865,11 @@ class Streamer():
                     if input_format == 'auto':
                         raise ValueError('input format details required for alsa input')
                     pcm = audio_io.PcmFormat.from_str(input_format)
-                    audio_input = AlsaArecordAudioInput(audio_source[5:], pcm)
+                    device_name = audio_source[5:]
+                    if device_name.startswith('dante_'):
+                        audio_input = PyAlsaAudioInput(device_name, pcm)
+                    else:
+                        audio_input = AlsaArecordAudioInput(device_name, pcm)
                 else:
                     audio_input = await audio_io.create_audio_input(audio_source, input_format)
                 # Store early so stop_streaming can close even if open() fails
@@ -823,6 +938,9 @@ class Streamer():
                     if lc3_frame == b'':  # Not all streams may stop at the same time
                         stream_finished[i] = True
                         continue
+
+                    for q_idx in range(big.get('num_bis', 1)):
+                        await big['iso_queues'][q_idx].write(lc3_frame)
                 else:  # code lc3 on the fly with perf counters
                     # Ensure frames generator exists (so we can aclose() on stop)
                     frames_gen = big.get('frames_gen')
@@ -852,6 +970,11 @@ class Streamer():
                         stream_finished[i] = True
                         continue
 
+                    # 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
+                    big['_audio_level_rms'] = float(rms)
+
                     # Measure LC3 encoding time
                     t1 = time.perf_counter()
                     num_bis = big.get('num_bis', 1)

src/auracast/multicast_control.py

@@ -37,6 +37,12 @@ class Multicaster:
             'is_initialized': self.is_auracast_init,
             'is_streaming': streaming,
         }
+
+    def get_audio_levels(self) -> list[float]:
+        """Return current RMS audio levels (0.0-1.0) for each BIG."""
+        if self.streamer is not None and self.streamer.is_streaming:
+            return self.streamer.get_audio_levels()
+        return []
     
     async def init_broadcast(self):
         self.device_acm = multicast.create_device(self.global_conf)
@@ -137,6 +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/http_to_https_redirect.py

@@ -0,0 +1,42 @@
+"""Minimal HTTP server that redirects all requests to HTTPS (port 443).
+
+Run on port 80 alongside the HTTPS Streamlit frontend so that users who
+type a bare IP address into their browser are automatically forwarded.
+"""
+
+import http.server
+import sys
+
+
+class RedirectHandler(http.server.BaseHTTPRequestHandler):
+    def do_GET(self):
+        host = self.headers.get("Host", "").split(":")[0] or self.server.server_address[0]
+        target = f"https://{host}{self.path}"
+        self.send_response(301)
+        self.send_header("Location", target)
+        self.end_headers()
+
+    # Handle every method the same way
+    do_POST = do_GET
+    do_PUT = do_GET
+    do_DELETE = do_GET
+    do_HEAD = do_GET
+
+    def log_message(self, format, *args):
+        # Keep logging minimal
+        sys.stderr.write(f"[http-redirect] {self.address_string()} -> https  {args[0] if args else ''}\n")
+
+
+def main():
+    port = int(sys.argv[1]) if len(sys.argv) > 1 else 80
+    server = http.server.HTTPServer(("0.0.0.0", port), RedirectHandler)
+    print(f"HTTP->HTTPS redirect server listening on 0.0.0.0:{port}")
+    try:
+        server.serve_forever()
+    except KeyboardInterrupt:
+        pass
+    server.server_close()
+
+
+if __name__ == "__main__":
+    main()

src/auracast/server/start_frontend_https.sh

@@ -33,5 +33,12 @@ echo "Using Avahi domain: $AVAHI_DOMAIN"
 # Path to poetry binary
 POETRY_BIN="/home/caster/.local/bin/poetry"
 
+# Start HTTP->HTTPS redirect server on port 80 (background)
+SCRIPT_DIR="$(cd "$(dirname "$0")" && pwd)"
+python3 "$SCRIPT_DIR/http_to_https_redirect.py" 80 &
+REDIRECT_PID=$!
+echo "HTTP->HTTPS redirect server started (PID $REDIRECT_PID)"
+trap "kill $REDIRECT_PID 2>/dev/null" EXIT
+
 # Start Streamlit HTTPS server (port 443)
 $POETRY_BIN run streamlit run multicast_frontend.py --server.port 443 --server.address 0.0.0.0 --server.enableCORS false --server.enableXsrfProtection false --server.headless true --server.sslCertFile "$CERT" --server.sslKeyFile "$KEY" --browser.gatherUsageStats false

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/auracast/utils/read_lc3_file.py

@@ -21,12 +21,12 @@ def read_lc3_file(filepath):
         logging.info('frame_duration %s', frame_duration)
         logging.info('stream_length %s', stream_length)
 
-        lc3_bytes= b''
+        chunks = []
         while True:
             b = f_lc3.read(2)
             if b == b'':
                 break
             lc3_frame_size = struct.unpack('=H', b)[0]
-            lc3_bytes += f_lc3.read(lc3_frame_size)
+            chunks.append(f_lc3.read(lc3_frame_size))
 
-        return lc3_bytes
+        return b''.join(chunks)

src/dep/dante_package/dante_data/activation/.gitkeep

@@ -0,0 +1 @@
+Placeholder file to have the activation folder available, otherwise dante activation script fails with 'Unable to write'.

src/dep/dante_package/dante_data/capability/dante.json

@@ -1,5 +1,5 @@
 {
-	"trialMode": true,
+	"trialMode": false,
 	"$schema": "./dante.json_schema.json",
 	"platform":
 	{
@@ -16,7 +16,7 @@
 				48000
 			],
 		"samplesPerPeriod" : 16,
-		"periodsPerBuffer" : 300,
+		"periodsPerBuffer" : 150,
 		"networkLatencyMinMs" : 2,
 		"networkLatencyDefaultMs" : 5,
 		"supportedEncodings" :
@@ -24,7 +24,10 @@
 				"PCM16"
 			],
 		"defaultEncoding" : "PCM16",
-		"numDepCores" : 1
+		"numDepCores" : 
+			[
+				3
+			]
 	},
 	"network" :
 	{
@@ -50,31 +53,32 @@
 	"alsaAsrc":
 	{
 		"enableAlsaAsrc": true,
+		"cpuAffinity": 3,
 		"deviceConfigurations": [
 			{
-			"deviceIdentifier": "hw:0,0",
+			"deviceIdentifier": "hw:6,0,0",
 			"direction": "playback",
 			"bitDepth": 16,
 			"numOpenChannels": 6,
 			"alsaChannelRange": "0-5",
 			"danteChannelRange": "0-5",
-			"bufferSize": 4800,
+			"bufferSize": 960,
 			"samplesPerPeriod": 16
 			}
 		]
 	},
 	"product" :
 	{
-		"manfId" : "Audinate",
-		"manfName" : "Audinate Pty Ltd",
-		"modelId" : "OEMDEP",
-		"modelName" : "Linux Dante Embedded Platform",
+		"manfId" : "SummitFC",
+		"manfName" : "Summitwave FlexCo",
+		"modelId" : "TX",
+		"modelName" : "Summitwave TX",
 		"modelVersion" :
 			{
-				"major" : 9,
-				"minor" : 9,
-				"bugfix" : 99
+				"major" : 1,
+				"minor" : 0,
+				"bugfix" : 0
 			},
-		"devicePrefix" : "DEP"
+		"devicePrefix" : "SW-TX"
 	}
 }

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 +1,56 @@
+#!/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"
+
+sudo openocd \
+  -f ./raspberrypi-${INTERFACE}.cfg \
+  -c "init" \
+  -c "nrf54l.dap apreg 2 0x000 0x4" \
+  -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

src/service/10-link-local-mgmt

@@ -0,0 +1,83 @@
+#!/bin/bash
+# NetworkManager dispatcher script: 10-link-local-mgmt
+#
+# Temporarily suppresses IPv4 link-local when a DHCP address is available,
+# using nmcli device modify (active session only, NOT saved to the profile).
+# The persistent profile always keeps ipv4.link-local=enabled so that
+# direct-connect (no DHCP) plug-ins always activate and trigger events.
+#
+# Triggers: up, down, dhcp4-change on ethernet interfaces
+# Install to: /etc/NetworkManager/dispatcher.d/10-link-local-mgmt
+# Permissions: root:root 0755
+
+INTERFACE="$1"
+ACTION="$2"
+CONNECTION_NAME="${CONNECTION_ID:-}"
+
+# Only handle ethernet interfaces
+if [[ ! "$INTERFACE" =~ ^eth ]]; then
+    exit 0
+fi
+
+# If CONNECTION_ID env var is not set, look up the active connection for this interface
+if [ -z "$CONNECTION_NAME" ]; then
+    CONNECTION_NAME=$(nmcli -t -f NAME,DEVICE connection show --active 2>/dev/null \
+        | grep ":${INTERFACE}$" | cut -d: -f1 | head -n1)
+    [ -z "$CONNECTION_NAME" ] && exit 0
+fi
+
+# Update /etc/avahi/hosts to point mDNS hostname at the best available DHCP address
+# across all ethernet interfaces (so Avahi doesn't advertise a link-local address).
+update_avahi() {
+    local hostname
+    hostname=$(hostname)
+    # Find first non-link-local IPv4 across all ethernet interfaces
+    local dhcp_ip
+    dhcp_ip=$(ip -4 addr show 2>/dev/null \
+        | grep -A5 ': eth' \
+        | grep -oP '(?<=inet\s)\d+(\.\d+){3}' \
+        | grep -v '^127\.' \
+        | grep -v '^169\.254\.' \
+        | head -n1)
+
+    if [ -n "$dhcp_ip" ]; then
+        mkdir -p /etc/avahi
+        echo "$dhcp_ip $hostname $hostname.local" > /etc/avahi/hosts
+        logger -t nm-link-local "Avahi: pinned $hostname -> $dhcp_ip"
+    else
+        rm -f /etc/avahi/hosts
+        logger -t nm-link-local "Avahi: removed hosts pin, using all addresses"
+    fi
+    systemctl restart avahi-daemon 2>/dev/null
+}
+
+case "$ACTION" in
+    up|dhcp4-change)
+        DHCP_IP=$(ip -4 addr show "$INTERFACE" 2>/dev/null \
+            | grep -oP '(?<=inet\s)\d+(\.\d+){3}' \
+            | grep -v '^127\.' \
+            | grep -v '^169\.254\.' \
+            | head -n1)
+
+        if [ -n "$DHCP_IP" ]; then
+            logger -t nm-link-local "[$INTERFACE] DHCP $DHCP_IP detected — suppressing link-local (session only)"
+            # Use device modify (not connection modify) so the persistent profile keeps
+            # ipv4.link-local=enabled. This ensures direct-connect plug-ins always activate.
+            # Run in background after a delay — nmcli blocks on NM, which is waiting for
+            # this dispatcher to return, causing a deadlock if called synchronously.
+            (sleep 2 && nmcli device modify "$INTERFACE" ipv4.link-local disabled 2>/dev/null \
+                && logger -t nm-link-local "[$INTERFACE] Link-local suppressed for current session") &
+        else
+            logger -t nm-link-local "[$INTERFACE] No DHCP on $INTERFACE — keeping link-local active"
+        fi
+        update_avahi
+        ;;
+
+    down)
+        # Profile always has ipv4.link-local=enabled so no action needed here.
+        # The suppression from device modify was session-only and is gone when the
+        # connection goes down.
+        logger -t nm-link-local "[$INTERFACE] Down — link-local will be active on next connect"
+        update_avahi
+        ;;
+esac

src/service/auracast-frontend.service

@@ -10,6 +10,8 @@ WorkingDirectory=/home/caster/bumble-auracast/src/auracast/server
 ExecStart=/home/caster/bumble-auracast/src/auracast/server/start_frontend_https.sh
 Restart=on-failure
 Environment=LOG_LEVEL=INFO
+AllowedCPUs=0
+CPUAffinity=0
 
 [Install]
 WantedBy=multi-user.target

src/service/auracast-script.service

@@ -9,6 +9,8 @@ ExecStart=/home/caster/bumble-auracast/.venv/bin/python src/auracast/multicast_s
 Restart=on-failure
 Environment=PYTHONUNBUFFERED=1
 Environment=LOG_LEVEL=INFO
+AllowedCPUs=0
+CPUAffinity=0
 
 [Install]
 WantedBy=default.target

src/service/auracast-server.service

@@ -1,6 +1,7 @@
 [Unit]
 Description=Auracast Backend Server
-After=network.target
+After=network.target dep.service
+Wants=dep.service
 
 [Service]
 Type=simple
@@ -10,8 +11,10 @@ Restart=on-failure
 Environment=PYTHONUNBUFFERED=1
 Environment=LOG_LEVEL=INFO
 CPUSchedulingPolicy=fifo
-CPUSchedulingPriority=99
+CPUSchedulingPriority=10
 LimitRTPRIO=99
+AllowedCPUs=0,1,2
+CPUAffinity=0,1,2
 
 [Install]
 WantedBy=default.target

src/service/dep.service

@@ -0,0 +1,13 @@
+[Unit]
+Description=DEP (Dante Embedded Platform) Container
+After=network.target
+
+[Service]
+Type=oneshot
+RemainAfterExit=yes
+WorkingDirectory=/home/caster/bumble-auracast/src/dep/dante_package
+ExecStart=/bin/bash dep.sh start
+ExecStop=/bin/bash dep.sh stop
+
+[Install]
+WantedBy=multi-user.target

src/service/pipewire-aes67.service

@@ -9,6 +9,8 @@ ExecStartPre=/bin/sh -lc 'for i in $(seq 1 60); do ip route show default >/dev/n
 ExecStart=/usr/bin/pipewire-aes67 -c /home/caster/bumble-auracast/src/service/aes67/pipewire-aes67.conf
 Restart=always
 RestartSec=5s
+AllowedCPUs=0
+CPUAffinity=0
 # Avoid StartLimitHit on quick failures during boot; let RestartSec handle pacing
 StartLimitIntervalSec=0
 

src/service/ptp_aes67.service

@@ -6,6 +6,8 @@ After=network.target
 Type=simple
 ExecStart=/usr/sbin/ptp4l -i eth0 -f /home/caster/bumble-auracast/src/service/aes67/ptp_aes67_1.conf
 Restart=on-failure
+AllowedCPUs=0
+CPUAffinity=0
 StandardOutput=journal
 StandardError=journal
 

src/service/update_and_run_server_and_frontend.sh

@@ -4,6 +4,33 @@ set -e
 # This script installs, enables, and restarts the auracast-server and auracast-frontend services
 # Requires sudo privileges
 
+# Ensure static link local is activated (for direct laptop connection)
+# Enable link-local for all wired ethernet connections
+while IFS=: read -r name type; do
+    if [[ "$type" == *"ethernet"* ]]; then
+        echo "Enabling IPv4 link-local for connection: $name"
+        sudo nmcli connection modify "$name" ipv4.link-local enabled 2>/dev/null || echo "Failed to modify $name"
+        sudo nmcli connection up "$name" 2>/dev/null || echo "Failed to bring up $name"
+    fi
+done < <(nmcli -t -f NAME,TYPE connection show)
+
+
+# Ensure Loopback is loaded with a fixed name and index 
+# Needed for dante
+# TODO image when we create the next image this should be part of it
+echo "options snd-aloop index=6 id=Loopback pcm_substreams=6" | sudo tee /etc/modprobe.d/snd-aloop.conf
+echo snd-aloop | sudo tee /etc/modules-load.d/snd-aloop.conf
+
+
+
+# Install NetworkManager dispatcher script for link-local / Avahi management
+sudo cp /home/caster/bumble-auracast/src/service/10-link-local-mgmt /etc/NetworkManager/dispatcher.d/10-link-local-mgmt
+sudo chown root:root /etc/NetworkManager/dispatcher.d/10-link-local-mgmt
+sudo chmod 755 /etc/NetworkManager/dispatcher.d/10-link-local-mgmt
+
+# Copy system service file for DEP
+sudo cp /home/caster/bumble-auracast/src/service/dep.service /etc/systemd/system/dep.service
+
 # Copy system service file for frontend
 sudo cp /home/caster/bumble-auracast/src/service/auracast-frontend.service /etc/systemd/system/auracast-frontend.service
 
@@ -11,20 +38,25 @@ sudo cp /home/caster/bumble-auracast/src/service/auracast-frontend.service /etc/
 mkdir -p /home/caster/.config/systemd/user
 cp /home/caster/bumble-auracast/src/service/auracast-server.service /home/caster/.config/systemd/user/auracast-server.service
 
-# Reload systemd for frontend
+# Reload systemd for frontend and dep
 sudo systemctl daemon-reload
 # Reload user systemd for server
 systemctl --user daemon-reload
 
+# Enable DEP to start on boot (system)
+sudo systemctl enable dep.service
 # Enable frontend to start on boot (system)
 sudo systemctl enable auracast-frontend.service
 # Enable server to start on boot (user)
 systemctl --user enable auracast-server.service
 
-# Restart both
+# Restart all
+sudo systemctl restart dep.service
+
 sudo systemctl restart auracast-frontend.service
 systemctl --user restart auracast-server.service
 
 #print status
+sudo systemctl status dep.service --no-pager
 sudo systemctl status auracast-frontend.service --no-pager
 systemctl --user status auracast-server.service --no-pager