29 Commits

Author SHA1 Message Date
pober 6375b215cb Autofocus password box on login screen. 2026-05-07 15:39:01 +02:00
pober 25df79eef5 Persists settings on refresh and power cycle. 2026-05-07 15:38:41 +02:00
pober 9c251b7a66 Fixes activation bug by adding the activation folder. 2026-04-28 09:05:39 +02:00
pober c24be9f366 Changed cpu pinning; more headroom for audiopipeline; dep and asrc on core 3 2026-04-28 09:01:58 +02:00
pober c659d632b0 Adds DEP service; gitignore for license; audiopipeline as its own thread with higher prio than http requests. 2026-04-27 15:35:20 +02:00
pober 14827288e7 relaxes asrc to not flip flop bang bang; use pyalsaaudio for dante. 2026-04-27 10:27:54 +02:00
pober 2410b01f15 Fixes loopback startup and makes it persistent with naming. 2026-04-27 10:26:04 +02:00
pober 67c774204a Dante buffers and cpu affinity. 2026-04-24 10:57:43 +02:00
pober c56012134c Fixes that an interface has both a local link and dhcp address - confuses dante. 2026-04-20 15:46:39 +02:00
pober 6d54e72f1d Add new reset mechanism with sleep. 2026-04-10 12:22:18 +02:00
pober df6c85d9ff Add new reset mechanism 2. 2026-04-10 11:58:16 +02:00
pober 8106f61d6a Add new reset mechanism. 2026-04-10 10:57:12 +02:00
pober 0a8dc74d5c Fixes script error in systemupdate. 2026-04-09 15:00:35 +02:00
pober 8475e4d068 New system update logic. 2026-04-09 14:46:17 +02:00
pober 3f01ef5968 Adds openocd with nrf support build to the server update function. Adds 2bad8ad2cd889d8c8d255b8e0dc0e7a187b98c9a hci_uart_beacon commit build hex file to project. (#26)
Co-authored-by: Pbopbo <p.obernesser@freenet.de>
Reviewed-on: #26
2026-04-09 12:04:18 +00:00
pober 67992e65ec Updates poetry lock. 2026-04-09 11:59:30 +02:00
pober 0b12323921 fix/gain-4dbU (#25)
Co-authored-by: Pbopbo <p.obernesser@freenet.de>
Reviewed-on: #25
2026-04-09 09:54:14 +00:00
Pbopbo 6e633d2880 Merge branch 'wip_alsaaudio' TODO poetry lock 2026-04-09 11:51:37 +02:00
pstruebi 7bdf6f8417 feature/blue_led (#23)
Co-authored-by: pstruebi <office@summitwave.eu>
Co-authored-by: pober <paul.obernesser@summitwave.eu>
Co-authored-by: Pbopbo <p.obernesser@freenet.de>
Reviewed-on: #23
Co-authored-by: pstruebi <struebin.patrick@gmail.com>
Co-committed-by: pstruebi <struebin.patrick@gmail.com>
2026-04-07 14:34:11 +00:00
Pbopbo 291d75b137 stereo seems to work, NEEDS RADIO FIRMWARE WITH 2 TX BUFFERS. 2026-04-07 14:36:15 +02:00
Pbopbo a126613739 First working version of two monos at the same time. 2026-04-02 18:56:17 +02:00
pober 036b5f80dd Updates poetry lock. 2026-04-02 18:10:23 +02:00
Pbopbo e818765b4f Adds sw_pyalsaaudio repo so our custom function works. 2026-04-02 17:37:38 +02:00
Pbopbo 3d59a6dabf ASRC: Adds NONBLOCK read from ALSA buffer; controls the amount of frames in the ALSA buffer; Adds resampling to get rid of audio glitches; no latency buildup anymore. 2026-04-01 14:00:26 +02:00
Pbopbo cf69ad2957 134ms constant delay, no build up, seems to be no glitches, bang bang control. 2026-03-30 14:45:25 +02:00
Pbopbo cdfecaf5eb delay method wip save to test no thread method. 2026-03-24 13:14:56 +01:00
pstruebi 4036fee1f5 Randomize Broadcast ID per stream instead of using static values 2026-03-24 12:09:16 +01:00
Pbopbo 1687a2b790 Latency lowered. 2026-03-18 17:37:34 +01:00
Pbopbo a605195646 First good audio with alsaaudio. 2026-03-18 16:55:55 +01:00
25 changed files with 14367 additions and 399 deletions
+8 -1
View File
@@ -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
Generated
+33 -46
View File
@@ -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"},
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{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"
+3 -1
View File
@@ -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]
+2 -2
View File
@@ -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)
+130 -90
View File
@@ -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 = 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):
@@ -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}',
@@ -760,7 +811,11 @@ 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
@@ -811,13 +866,6 @@ 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 +913,6 @@ 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
+4
View File
@@ -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(
+364 -136
View File
@@ -71,6 +71,10 @@ if not is_pw_disabled():
with st.form("signin_form"):
pw = st.text_input("Password", type="password")
submitted = st.form_submit_button("Sign in")
st.components.v1.html(
"<script>setTimeout(()=>window.parent.document.querySelector('input[type=\"password\"]')?.focus(),100)</script>",
height=0
)
if submitted:
if verify_password(pw, pw_rec):
st.session_state['frontend_authenticated'] = True
@@ -198,35 +202,64 @@ 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,
@@ -404,13 +437,43 @@ else:
disabled=is_streaming
)
# Use analog-specific defaults (not from saved settings which may have Dante values)
default_name = "Analog_Radio_1"
default_program_info = "Analog Radio Broadcast"
default_lang = "deu"
quality_options = list(QUALITY_MAP.keys())
default_quality = "Medium (24kHz)" if "Medium (24kHz)" in quality_options else quality_options[0]
# Use saved settings if audio_mode matches, otherwise use analog-specific defaults
saved_audio_mode = saved_settings.get('audio_mode')
if saved_audio_mode == 'Analog':
default_name = saved_settings.get('channel_names', ["Analog_Radio_1"])[0]
raw_program_info = saved_settings.get('program_info', default_name)
if isinstance(raw_program_info, list) and raw_program_info:
default_program_info = raw_program_info[0]
else:
default_program_info = raw_program_info
default_lang = saved_settings.get('languages', ["deu"])[0]
# Map saved sampling rate to quality label
saved_rate = saved_settings.get('auracast_sampling_rate_hz')
if saved_rate == 48000:
default_quality = "High (48kHz)"
elif saved_rate == 32000:
default_quality = "Good (32kHz)"
elif saved_rate == 24000:
default_quality = "Medium (24kHz)"
elif saved_rate == 16000:
default_quality = "Fair (16kHz)"
else:
default_quality = "Medium (24kHz)"
saved_pwd = saved_settings.get('stream_password', '')
else:
# Use analog-specific defaults when switching from another mode
default_name = "Analog_Radio_1"
default_program_info = "Analog Radio Broadcast"
default_lang = "deu"
default_quality = "Medium (24kHz)" if "Medium (24kHz)" in quality_options else quality_options[0]
saved_pwd = ''
if default_quality not in quality_options:
default_quality = quality_options[0]
quality1 = st.selectbox(
"Stream Quality (Radio 1)",
quality_options,
@@ -421,7 +484,7 @@ else:
stream_passwort1 = st.text_input(
"Stream Passwort (Radio 1)",
value="",
value=saved_pwd,
type="password",
disabled=is_streaming,
help="Optional: Set a broadcast code for Radio 1."
@@ -528,7 +591,10 @@ else:
input_device1 = None
else:
# Mono mode: show all available channels
saved_input_device = saved_settings.get('input_device')
default_r1_idx = 0
if saved_input_device in analog_names:
default_r1_idx = analog_names.index(saved_input_device)
input_device1 = st.selectbox(
"Input Device (Radio 1)",
analog_names,
@@ -577,22 +643,53 @@ else:
)
if radio2_enabled and not stereo_enabled:
# Use analog-specific defaults for Radio 2
default_name_r2 = "Analog_Radio_2"
default_program_info_r2 = "Analog Radio Broadcast"
default_lang_r2 = "deu"
# Use saved settings if audio_mode matches, otherwise use analog-specific defaults for Radio 2
secondary_settings = saved_settings.get('secondary', {})
saved_audio_mode = saved_settings.get('audio_mode')
if saved_audio_mode == 'Analog' and secondary_settings:
default_name_r2 = secondary_settings.get('channel_names', ["Analog_Radio_2"])[0] if isinstance(secondary_settings.get('channel_names'), list) else secondary_settings.get('channel_names', "Analog_Radio_2")
raw_program_info_r2 = secondary_settings.get('program_info', default_name_r2)
if isinstance(raw_program_info_r2, list) and raw_program_info_r2:
default_program_info_r2 = raw_program_info_r2[0]
else:
default_program_info_r2 = raw_program_info_r2
default_lang_r2 = secondary_settings.get('languages', ["deu"])[0] if isinstance(secondary_settings.get('languages'), list) else secondary_settings.get('languages', 'deu')
# Map saved sampling rate to quality label
saved_rate_r2 = secondary_settings.get('auracast_sampling_rate_hz')
if saved_rate_r2 == 48000:
default_quality_r2 = "High (48kHz)"
elif saved_rate_r2 == 32000:
default_quality_r2 = "Good (32kHz)"
elif saved_rate_r2 == 24000:
default_quality_r2 = "Medium (24kHz)"
elif saved_rate_r2 == 16000:
default_quality_r2 = "Fair (16kHz)"
else:
default_quality_r2 = "Medium (24kHz)"
saved_pwd_r2 = secondary_settings.get('stream_password', '')
else:
# Use analog-specific defaults when switching from another mode
default_name_r2 = "Analog_Radio_2"
default_program_info_r2 = "Analog Radio Broadcast"
default_lang_r2 = "deu"
default_quality_r2 = "Medium (24kHz)" if "Medium (24kHz)" in quality_options else quality_options[0]
saved_pwd_r2 = ''
if default_quality_r2 not in quality_options:
default_quality_r2 = quality_options[0]
quality2 = st.selectbox(
"Stream Quality (Radio 2)",
quality_options,
index=quality_options.index(default_quality),
index=quality_options.index(default_quality_r2),
disabled=is_streaming,
help="Select the audio sampling rate for Radio 2."
)
stream_passwort2 = st.text_input(
"Stream Passwort (Radio 2)",
value="",
value=saved_pwd_r2,
type="password",
disabled=is_streaming,
help="Optional: Set a broadcast code for Radio 2."
@@ -653,7 +750,11 @@ else:
if not is_streaming:
if analog_names:
secondary_settings = saved_settings.get('secondary', {})
saved_input_device2 = secondary_settings.get('input_device')
default_r2_idx = 1 if len(analog_names) > 1 else 0
if saved_input_device2 in analog_names:
default_r2_idx = analog_names.index(saved_input_device2)
input_device2 = st.selectbox(
"Input Device (Radio 2)",
analog_names,
@@ -684,8 +785,8 @@ 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 +802,8 @@ 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":
@@ -745,10 +846,15 @@ else:
)
# Dante stereo mode toggle
saved_r1_config = saved_settings.get('dante_radio1', {})
saved_audio_mode = saved_settings.get('audio_mode')
dante_stereo_enabled = False
if saved_audio_mode == 'Network - Dante':
# Check if any input device starts with dante_stereo_ to detect stereo mode
input_device = saved_settings.get('input_device', '')
dante_stereo_enabled = input_device.startswith('dante_stereo_')
dante_stereo_enabled = st.checkbox(
"🎧 Stereo Mode",
value=bool(saved_r1_config.get('dante_stereo_mode', False)),
value=dante_stereo_enabled,
help="Enable stereo streaming for Dante inputs. Select left and right channels from ASRC channels 1-6. Radio 2 and multi-stream configurations will be disabled in stereo mode.",
disabled=is_streaming
)
@@ -757,13 +863,23 @@ else:
dante_left_channel = None
dante_right_channel = None
if dante_stereo_enabled:
dante_channel_options = ["dante_asrc_ch1", "dante_asrc_ch2", "dante_asrc_ch3",
dante_channel_options = ["dante_asrc_ch1", "dante_asrc_ch2", "dante_asrc_ch3",
"dante_asrc_ch4", "dante_asrc_ch5", "dante_asrc_ch6"]
dante_channel_labels = ["CH1", "CH2", "CH3", "CH4", "CH5", "CH6"]
# Parse saved stereo device name to extract left and right channels
input_device = saved_settings.get('input_device', '')
saved_left = 'dante_asrc_ch1'
saved_right = 'dante_asrc_ch2'
if input_device.startswith('dante_stereo_'):
# Format: dante_stereo_<left>_<right>
parts = input_device.split('_')
if len(parts) >= 4:
saved_left = f"dante_asrc_ch{parts[2]}"
saved_right = f"dante_asrc_ch{parts[3]}"
col_left, col_right = st.columns(2)
with col_left:
saved_left = saved_r1_config.get('dante_stereo_left', 'dante_asrc_ch1')
left_idx = dante_channel_options.index(saved_left) if saved_left in dante_channel_options else 0
dante_left_channel = st.selectbox(
"Left Channel",
@@ -774,7 +890,6 @@ else:
help="Select the Dante ASRC channel for the left stereo channel"
)
with col_right:
saved_right = saved_r1_config.get('dante_stereo_right', 'dante_asrc_ch2')
right_idx = dante_channel_options.index(saved_right) if saved_right in dante_channel_options else 1
dante_right_channel = st.selectbox(
"Right Channel",
@@ -784,7 +899,7 @@ else:
disabled=is_streaming,
help="Select the Dante ASRC channel for the right stereo channel"
)
if dante_left_channel == dante_right_channel:
st.warning("⚠️ Left and right channels are the same. Select different channels for true stereo.")
else:
@@ -792,7 +907,22 @@ else:
# Stream count dropdown for Radio 1 (disabled in stereo mode - forced to 1 stream at 48kHz)
r1_stream_options = list(dante_stream_options.keys())
saved_r1_streams = saved_r1_config.get('stream_config', '1x48')
# Infer stream configuration from saved sampling rate
saved_rate = saved_settings.get('auracast_sampling_rate_hz')
saved_r1_streams = '1 × 48kHz' # default
if saved_rate:
if saved_rate == 48000:
channel_names = saved_settings.get('channel_names', [])
if len(channel_names) == 2:
saved_r1_streams = '2 × 24kHz'
elif len(channel_names) == 3:
saved_r1_streams = '3 × 16kHz'
else:
saved_r1_streams = '1 × 48kHz'
elif saved_rate == 24000:
saved_r1_streams = '2 × 24kHz'
elif saved_rate == 16000:
saved_r1_streams = '3 × 16kHz'
default_r1_idx = r1_stream_options.index(saved_r1_streams) if saved_r1_streams in r1_stream_options else 0
if dante_stereo_enabled:
@@ -822,15 +952,25 @@ else:
r1_available_qualities = []
for quality in ["High (48kHz)", "Good (32kHz)", "Medium (24kHz)", "Fair (16kHz)"]:
# Check if this quality is equal to or lower than the max
if (r1_max_quality == "High (48kHz)" or
if (r1_max_quality == "High (48kHz)" or
(r1_max_quality == "Medium (24kHz)" and quality in ["Medium (24kHz)", "Fair (16kHz)"]) or
(r1_max_quality == "Fair (16kHz)" and quality == "Fair (16kHz)")):
r1_available_qualities.append(quality)
saved_r1_quality = saved_r1_config.get('radio_quality', r1_max_quality)
# Map saved sampling rate to quality label
saved_r1_quality = r1_max_quality
saved_rate = saved_settings.get('auracast_sampling_rate_hz')
if saved_rate == 48000:
saved_r1_quality = "High (48kHz)"
elif saved_rate == 32000:
saved_r1_quality = "Good (32kHz)"
elif saved_rate == 24000:
saved_r1_quality = "Medium (24kHz)"
elif saved_rate == 16000:
saved_r1_quality = "Fair (16kHz)"
if saved_r1_quality not in r1_available_qualities:
saved_r1_quality = r1_max_quality
r1_radio_quality = st.selectbox(
"Stream Quality (Radio 1)",
r1_available_qualities,
@@ -838,29 +978,29 @@ else:
disabled=is_streaming,
help=f"Select stream quality for Radio 1. Maximum quality based on configuration: {r1_max_quality}"
)
# Radio-level settings for Radio 1
# First row: Assistive listening, immediate rendering, presentation delay, QoS
col_r1_flags1, col_r1_flags2, col_r1_pdelay, col_r1_qos = st.columns([1, 1, 0.7, 0.6], gap="small")
with col_r1_flags1:
r1_assisted_listening = st.checkbox(
"Assistive (R1)",
value=bool(saved_r1_config.get('assisted_listening', False)),
value=bool(saved_settings.get('assisted_listening_stream', False)),
disabled=is_streaming,
help="Assistive listening stream"
)
with col_r1_flags2:
r1_immediate_rendering = st.checkbox(
"Immediate (R1)",
value=bool(saved_r1_config.get('immediate_rendering', False)),
value=bool(saved_settings.get('immediate_rendering', False)),
disabled=is_streaming,
help="Ignore presentation delay"
)
with col_r1_pdelay:
default_pdelay = int(saved_r1_config.get('presentation_delay_us', 40000) or 40000)
default_pdelay = int(saved_settings.get('presentation_delay_us', 40000) or 40000)
default_pdelay_ms = max(10, min(200, default_pdelay // 1000))
r1_presentation_delay_ms = st.number_input(
"Delay (ms, R1)",
@@ -868,10 +1008,10 @@ else:
disabled=is_streaming,
help="Presentation delay for Radio 1"
)
with col_r1_qos:
qos_options = list(QOS_PRESET_MAP.keys())
saved_qos = saved_r1_config.get('qos_preset', 'Fast')
saved_qos = saved_settings.get('qos_preset', 'Fast')
default_qos_idx = qos_options.index(saved_qos) if saved_qos in qos_options else 0
r1_qos_preset = st.selectbox(
"QoS (R1)", options=qos_options, index=default_qos_idx,
@@ -889,24 +1029,31 @@ else:
if dante_stereo_enabled:
# Stereo mode: single stream with combined L+R channels
with st.expander("Stereo Stream - Radio 1", expanded=True):
saved_streams = saved_r1_config.get('streams', [])
saved_stream = saved_streams[0] if saved_streams else {}
# Read from flat settings structure
channel_names = saved_settings.get('channel_names', [])
program_infos = saved_settings.get('program_info', [])
languages = saved_settings.get('languages', [])
saved_name = channel_names[0] if channel_names else 'Dante_Stereo'
saved_program_info = program_infos[0] if program_infos else saved_name
saved_language = languages[0] if languages else 'eng'
saved_password = saved_settings.get('stream_password', '')
# First row: Channel name and password
col_name, col_pwd = st.columns([2, 1])
with col_name:
stream_name = st.text_input(
"Channel Name",
value=saved_stream.get('name', 'Dante_Stereo'),
value=saved_name,
disabled=is_streaming,
key="r1_stereo_name"
)
with col_pwd:
stream_password = st.text_input(
"Stream Password",
value=saved_stream.get('stream_password', ''),
value=saved_password,
type="password",
disabled=is_streaming,
key="r1_stereo_password",
@@ -915,19 +1062,19 @@ else:
# Second row: Program info and language
col_prog, col_lang_code = st.columns([2, 1])
with col_prog:
program_info = st.text_input(
"Program Info",
value=saved_stream.get('program_info', 'Dante Stereo Broadcast'),
value=saved_program_info,
disabled=is_streaming,
key="r1_stereo_program"
)
with col_lang_code:
language = st.text_input(
"Language",
value=saved_stream.get('language', 'eng'),
value=saved_language,
disabled=is_streaming,
key="r1_stereo_lang",
help="ISO 639-3 language code"
@@ -961,47 +1108,58 @@ else:
})
else:
# Normal mono mode: multiple streams with individual channels
# Read from flat settings structure
channel_names = saved_settings.get('channel_names', [])
program_infos = saved_settings.get('program_info', [])
languages = saved_settings.get('languages', [])
input_devices = saved_settings.get('input_devices', [])
stream_passwords = saved_settings.get('stream_passwords', []) if 'stream_passwords' in saved_settings else []
for i in range(r1_num_streams):
with st.expander(f"Stream {i+1} - Radio 1", expanded=True):
saved_streams = saved_r1_config.get('streams', [])
saved_stream = saved_streams[i] if i < len(saved_streams) else {}
# Get saved values from flat structure
saved_name = channel_names[i] if i < len(channel_names) else f'Dante_R1_S{i+1}'
saved_program_info = program_infos[i] if i < len(program_infos) else f'Dante Radio 1 Stream {i+1}'
saved_language = languages[i] if i < len(languages) else 'eng'
saved_password = stream_passwords[i] if i < len(stream_passwords) else ''
saved_input_device = input_devices[i] if i < len(input_devices) else None
# First row: Channel name and language
col_name, col_lang = st.columns([2, 1])
with col_name:
stream_name = st.text_input(
f"Channel Name",
value=saved_stream.get('name', f'Dante_R1_S{i+1}'),
value=saved_name,
disabled=is_streaming,
key=f"r1_stream_{i}_name"
)
with col_lang:
stream_password = st.text_input(
f"Stream Password",
value=saved_stream.get('stream_password', ''),
value=saved_password,
type="password",
disabled=is_streaming,
key=f"r1_stream_{i}_password",
help="Optional: Set a broadcast code for this stream"
)
# Second row: Program info and language
col_prog, col_lang_code = st.columns([2, 1])
with col_prog:
program_info = st.text_input(
f"Program Info",
value=saved_stream.get('program_info', f'Dante Radio 1 Stream {i+1}'),
value=saved_program_info,
disabled=is_streaming,
key=f"r1_stream_{i}_program"
)
with col_lang_code:
language = st.text_input(
f"Language",
value=saved_stream.get('language', 'eng'),
value=saved_language,
disabled=is_streaming,
key=f"r1_stream_{i}_lang",
help="ISO 639-3 language code"
@@ -1013,10 +1171,10 @@ else:
with col_device:
# Session state key for persisting the selection
device_session_key = f"r1_stream_{i}_device_saved"
if not is_streaming and input_options:
# Get default from session state first, then from saved settings
default_input_name = st.session_state.get(device_session_key, saved_stream.get('input_device'))
default_input_name = st.session_state.get(device_session_key, saved_input_device)
default_input_label = None
for label, name in option_name_map.items():
if name == default_input_name:
@@ -1024,7 +1182,7 @@ else:
break
if default_input_label not in input_options and input_options:
default_input_label = input_options[0]
selected_option = st.selectbox(
f"Input Device",
input_options,
@@ -1037,7 +1195,7 @@ else:
st.session_state[device_session_key] = input_device
else:
# When streaming, get the device from session state
current_device = st.session_state.get(device_session_key, saved_stream.get('input_device', 'No device'))
current_device = st.session_state.get(device_session_key, saved_input_device or 'No device')
# Convert internal name to display label
display_label = current_device
@@ -1066,26 +1224,45 @@ else:
# --- Radio 2 Section ---
with st.container(border=True):
st.subheader("Radio 2")
# Disable Radio 2 in stereo mode
saved_r2_config = saved_settings.get('dante_radio2', {})
secondary_settings = saved_settings.get('secondary', {})
if dante_stereo_enabled:
st.info("🎧 Radio 2 is automatically disabled in stereo mode")
radio2_enabled = False
else:
# Enable/disable checkbox for Radio 2
# Use saved settings or streaming state to determine default
radio2_enabled_default = secondary_is_streaming
# Check if secondary radio has saved settings (indicates it was enabled)
if secondary_settings.get('auracast_sampling_rate_hz') or secondary_settings.get('channel_names'):
radio2_enabled_default = True
radio2_enabled = st.checkbox(
"Enable Radio 2",
value=radio2_enabled_default,
disabled=is_streaming,
help="Activate a second Dante radio with its own quality and timing settings."
)
if radio2_enabled:
# Stream count dropdown for Radio 2
r2_stream_options = r1_stream_options
saved_r2_streams = saved_r2_config.get('stream_config', '1x48')
# Infer stream configuration from saved secondary sampling rate
saved_rate2 = secondary_settings.get('auracast_sampling_rate_hz')
saved_r2_streams = '1 × 48kHz' # default
if saved_rate2:
if saved_rate2 == 48000:
channel_names2 = secondary_settings.get('channel_names', [])
if len(channel_names2) == 2:
saved_r2_streams = '2 × 24kHz'
elif len(channel_names2) == 3:
saved_r2_streams = '3 × 16kHz'
else:
saved_r2_streams = '1 × 48kHz'
elif saved_rate2 == 24000:
saved_r2_streams = '2 × 24kHz'
elif saved_rate2 == 16000:
saved_r2_streams = '3 × 16kHz'
default_r2_idx = r2_stream_options.index(saved_r2_streams) if saved_r2_streams in r2_stream_options else 0
r2_stream_config = st.selectbox(
@@ -1107,11 +1284,20 @@ else:
(r2_max_quality == "Medium (24kHz)" and quality in ["Medium (24kHz)", "Fair (16kHz)"]) or
(r2_max_quality == "Fair (16kHz)" and quality == "Fair (16kHz)")):
r2_available_qualities.append(quality)
saved_r2_quality = saved_r2_config.get('radio_quality', r2_max_quality)
# Map saved secondary sampling rate to quality label
saved_r2_quality = r2_max_quality
if saved_rate2 == 48000:
saved_r2_quality = "High (48kHz)"
elif saved_rate2 == 32000:
saved_r2_quality = "Good (32kHz)"
elif saved_rate2 == 24000:
saved_r2_quality = "Medium (24kHz)"
elif saved_rate2 == 16000:
saved_r2_quality = "Fair (16kHz)"
if saved_r2_quality not in r2_available_qualities:
saved_r2_quality = r2_max_quality
r2_radio_quality = st.selectbox(
"Stream Quality (Radio 2)",
r2_available_qualities,
@@ -1119,29 +1305,28 @@ else:
disabled=is_streaming,
help=f"Select stream quality for Radio 2. Maximum quality based on configuration: {r2_max_quality}"
)
# Radio-level settings for Radio 2
# First row: Assistive listening, immediate rendering, presentation delay, QoS
col_r2_flags1, col_r2_flags2, col_r2_pdelay, col_r2_qos = st.columns([1, 1, 0.7, 0.6], gap="small")
with col_r2_flags1:
r2_assisted_listening = st.checkbox(
"Assistive (R2)",
value=bool(saved_r2_config.get('assisted_listening', False)),
value=bool(secondary_settings.get('assisted_listening_stream', False)),
disabled=is_streaming,
help="Assistive listening stream"
)
with col_r2_flags2:
r2_immediate_rendering = st.checkbox(
"Immediate (R2)",
value=bool(saved_r2_config.get('immediate_rendering', False)),
value=bool(secondary_settings.get('immediate_rendering', False)),
disabled=is_streaming,
help="Ignore presentation delay"
)
with col_r2_pdelay:
default_pdelay = int(saved_r2_config.get('presentation_delay_us', 40000) or 40000)
default_pdelay = int(secondary_settings.get('presentation_delay_us', 40000) or 40000)
default_pdelay_ms = max(10, min(200, default_pdelay // 1000))
r2_presentation_delay_ms = st.number_input(
"Delay (ms, R2)",
@@ -1149,13 +1334,13 @@ else:
disabled=is_streaming,
help="Presentation delay for Radio 2"
)
with col_r2_qos:
qos_options = list(QOS_PRESET_MAP.keys())
saved_qos = saved_r2_config.get('qos_preset', 'Fast')
default_qos_idx = qos_options.index(saved_qos) if saved_qos in qos_options else 0
saved_qos = secondary_settings.get('qos_preset', 'Fast')
default_qos_idx2 = qos_options.index(saved_qos) if saved_qos in qos_options else 0
r2_qos_preset = st.selectbox(
"QoS (R2)", options=qos_options, index=default_qos_idx,
"QoS (R2)", options=qos_options, index=default_qos_idx2,
disabled=is_streaming,
help="Quality of Service preset for Radio 2"
)
@@ -1163,48 +1348,59 @@ else:
# Per-stream configuration for Radio 2
st.write("**Stream Configuration (Radio 2)**")
r2_streams = []
# Read from flat secondary settings structure
channel_names2 = secondary_settings.get('channel_names', [])
program_infos2 = secondary_settings.get('program_info', [])
languages2 = secondary_settings.get('languages', [])
input_devices2 = secondary_settings.get('input_devices', [])
stream_passwords2 = secondary_settings.get('stream_passwords', []) if 'stream_passwords' in secondary_settings else []
for i in range(r2_num_streams):
with st.expander(f"Stream {i+1} - Radio 2", expanded=True):
saved_streams = saved_r2_config.get('streams', [])
saved_stream = saved_streams[i] if i < len(saved_streams) else {}
# Get saved values from flat secondary structure
saved_name2 = channel_names2[i] if i < len(channel_names2) else f'Dante_R2_S{i+1}'
saved_program_info2 = program_infos2[i] if i < len(program_infos2) else f'Dante Radio 2 Stream {i+1}'
saved_language2 = languages2[i] if i < len(languages2) else 'eng'
saved_password2 = stream_passwords2[i] if i < len(stream_passwords2) else ''
saved_input_device2 = input_devices2[i] if i < len(input_devices2) else None
# First row: Channel name and password
col_name, col_pwd = st.columns([2, 1])
with col_name:
stream_name = st.text_input(
f"Channel Name",
value=saved_stream.get('name', f'Dante_R2_S{i+1}'),
value=saved_name2,
disabled=is_streaming,
key=f"r2_stream_{i}_name"
)
with col_pwd:
stream_password = st.text_input(
f"Stream Password",
value=saved_stream.get('stream_password', ''),
value=saved_password2,
type="password",
disabled=is_streaming,
key=f"r2_stream_{i}_password",
help="Optional: Set a broadcast code for this stream"
)
# Second row: Program info and language
col_prog, col_lang = st.columns([2, 1])
with col_prog:
program_info = st.text_input(
f"Program Info",
value=saved_stream.get('program_info', f'Dante Radio 2 Stream {i+1}'),
value=saved_program_info2,
disabled=is_streaming,
key=f"r2_stream_{i}_program"
)
with col_lang:
language = st.text_input(
f"Language",
value=saved_stream.get('language', 'eng'),
value=saved_language2,
disabled=is_streaming,
key=f"r2_stream_{i}_lang",
help="ISO 639-3 language code"
@@ -1216,10 +1412,10 @@ else:
with col_device:
# Session state key for persisting the selection
device_session_key = f"r2_stream_{i}_device_saved"
if not is_streaming and input_options:
# Get default from session state first, then from saved settings
default_input_name = st.session_state.get(device_session_key, saved_stream.get('input_device'))
default_input_name = st.session_state.get(device_session_key, saved_input_device2)
default_input_label = None
for label, name in option_name_map.items():
if name == default_input_name:
@@ -1227,7 +1423,7 @@ else:
break
if default_input_label not in input_options and input_options:
default_input_label = input_options[0]
selected_option = st.selectbox(
f"Input Device",
input_options,
@@ -1240,7 +1436,7 @@ else:
st.session_state[device_session_key] = input_device
else:
# When streaming, get the device from session state
current_device = st.session_state.get(device_session_key, saved_stream.get('input_device', 'No device'))
current_device = st.session_state.get(device_session_key, saved_input_device2 or 'No device')
# Convert internal name to display label
display_label = current_device
@@ -1325,8 +1521,30 @@ else:
if audio_mode in ("USB", "Network"):
# USB/Network: single set of controls shared with the single channel
# Use saved settings if audio_mode matches, otherwise use defaults
saved_audio_mode = saved_settings.get('audio_mode')
if saved_audio_mode in ("USB", "Network"):
# Map saved sampling rate to quality label
saved_rate = saved_settings.get('auracast_sampling_rate_hz')
if saved_rate == 48000:
default_quality = "High (48kHz)"
elif saved_rate == 32000:
default_quality = "Good (32kHz)"
elif saved_rate == 24000:
default_quality = "Medium (24kHz)"
elif saved_rate == 16000:
default_quality = "Fair (16kHz)"
else:
default_quality = "Medium (24kHz)"
saved_pwd = saved_settings.get('stream_password', '')
else:
# Use defaults when switching from another mode
default_quality = "Medium (24kHz)" if "Medium (24kHz)" in quality_options else quality_options[0]
saved_pwd = ''
quality_options = list(QUALITY_MAP.keys())
default_quality = "Medium (24kHz)" if "Medium (24kHz)" in quality_options else quality_options[0]
if default_quality not in quality_options:
default_quality = quality_options[0]
quality = st.selectbox(
"Stream Quality (Sampling Rate)",
quality_options,
@@ -1337,7 +1555,7 @@ else:
stream_passwort = st.text_input(
"Stream Passwort",
value="",
value=saved_pwd,
type="password",
disabled=is_streaming,
help="Optional: Set a broadcast code to protect your stream. Leave empty for an open (uncoded) broadcast."
@@ -1585,11 +1803,11 @@ 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),
code=((cfg['stream_passwort'] or '').strip() or None),
name=cfg['name'],
program_info=cfg['program_info'],
language=cfg['language'],
@@ -1599,7 +1817,6 @@ 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 +1860,6 @@ 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 +1867,6 @@ 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 +1969,21 @@ 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:
+401 -84
View File
@@ -3,12 +3,14 @@ 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
import asyncio
import random
import subprocess
import threading
from dotenv import load_dotenv
from fastapi import FastAPI, HTTPException
@@ -26,6 +28,57 @@ 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 +186,10 @@ 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
@@ -152,6 +209,28 @@ multicaster1: multicast_control.Multicaster | None = None
multicaster2: multicast_control.Multicaster | None = None
_stream_lock = asyncio.Lock() # serialize initialize/stop_audio on API side
# BLE / audio event loop set in __main__ before uvicorn starts.
# All coroutines that touch Bumble objects or the audio pipeline MUST run
# on this loop. HTTP handlers call _on_ble_loop() to cross into it.
_ble_loop: asyncio.AbstractEventLoop | None = None
async def _on_ble_loop(coro):
"""Submit *coro* to the BLE event loop and await the result.
Called from uvicorn's event loop. Bridges HTTP handler coroutines into
the isolated BLE loop so that serial I/O (serial_asyncio / HCI) and the
audio pipeline are never preempted by HTTP accept/read/write callbacks.
asyncio.run_coroutine_threadsafe() schedules the coroutine on _ble_loop
(thread-safe), returning a concurrent.futures.Future.
asyncio.wrap_future() adapts that into an asyncio.Future so the caller
can simply `await` it inside uvicorn's loop.
"""
assert _ble_loop is not None, "BLE loop not yet initialised"
future = asyncio.run_coroutine_threadsafe(coro, _ble_loop)
return await asyncio.wrap_future(future)
async def _init_i2c_on_startup() -> None:
# Ensure i2c-dev kernel module is loaded (required for /dev/i2c-* access)
@@ -238,16 +317,18 @@ 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 +337,46 @@ 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 +396,7 @@ async def _stop_all() -> bool:
was_running = True
finally:
multicaster2 = None
_led_off()
return was_running
async def _status_primary() -> dict:
@@ -332,6 +445,11 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
first_source = conf.bigs[0].audio_source if conf.bigs else ''
input_device_name = None
audio_mode_persist = 'Demo'
# Capture original per-BIG device names before transformation
original_input_devices = [
big.audio_source.split(':', 1)[1] if (isinstance(big.audio_source, str) and big.audio_source.startswith('device:')) else None
for big in conf.bigs
]
if any(isinstance(b.audio_source, str) and b.audio_source.startswith('device:') for b in conf.bigs):
if isinstance(first_source, str) and first_source.startswith('device:'):
input_device_name = first_source.split(':', 1)[1] if ':' in first_source else None
@@ -343,8 +461,9 @@ 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 +506,15 @@ 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 +570,12 @@ 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 +586,7 @@ 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:'))
@@ -483,6 +602,7 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
'languages': [big.language for big in conf.bigs],
'audio_mode': audio_mode_persist,
'input_device': input_device_name,
'input_devices': original_input_devices,
'program_info': [getattr(big, 'program_info', None) for big in conf.bigs],
'gain': [getattr(big, 'input_gain', 1.0) for big in conf.bigs],
'auracast_sampling_rate_hz': conf.auracast_sampling_rate_hz,
@@ -492,8 +612,8 @@ 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],
@@ -511,7 +631,10 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
@app.post("/init")
async def initialize(conf: auracast_config.AuracastConfigGroup):
"""Initializes the primary broadcaster on the streamer thread."""
"""Initializes the primary broadcaster on the BLE loop."""
return await _on_ble_loop(_initialize_impl(conf))
async def _initialize_impl(conf: auracast_config.AuracastConfigGroup):
async with _stream_lock:
global multicaster1, global_config_group
mc, persisted = await init_radio(TRANSPORT1, conf, multicaster1)
@@ -521,16 +644,33 @@ async def initialize(conf: auracast_config.AuracastConfigGroup):
@app.post("/init2")
async def initialize2(conf: auracast_config.AuracastConfigGroup):
"""Initializes the secondary broadcaster on the streamer thread."""
"""Initializes the secondary broadcaster on the BLE loop."""
return await _on_ble_loop(_initialize2_impl(conf))
async def _initialize2_impl(conf: auracast_config.AuracastConfigGroup):
async with _stream_lock:
global multicaster2
mc, persisted = await init_radio(TRANSPORT2, conf, multicaster2)
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)."""
"""Stops streaming on both multicasters via the BLE loop."""
return await _on_ble_loop(_stop_audio_impl())
async def _stop_audio_impl():
"""Runs on BLE loop: stops all streamers and persists is_streaming=False."""
try:
was_running = await _stop_all()
@@ -556,11 +696,33 @@ 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."""
"""Sends a block of pre-coded LC3 audio via the BLE loop."""
try:
await _stream_lc3(audio_data, list(global_config_group.bigs))
await _on_ble_loop(_stream_lc3(audio_data, list(global_config_group.bigs)))
return {"status": "audio_sent"}
except Exception as e:
raise HTTPException(status_code=500, detail=str(e))
@@ -591,6 +753,7 @@ 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 +914,6 @@ 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 +923,8 @@ 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 +1069,6 @@ 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 +1078,8 @@ 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"])
@@ -934,13 +1097,27 @@ async def _autostart_from_settings():
await do_primary()
await do_secondary()
async def _ble_startup():
"""I2C init, ADC level reset, and autostart task scheduling on the BLE loop.
Bridged from _startup_autostart_event() so that these async subprocess
calls and the long-lived autostart coroutine all run on _ble_loop, never
on uvicorn's HTTP loop.
"""
await _init_i2c_on_startup()
await _set_adc_level(0.0, 0.0)
log.info("[STARTUP] Scheduling autostart task on BLE loop")
asyncio.create_task(_autostart_from_settings())
@app.on_event("startup")
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)
@@ -951,13 +1128,13 @@ 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)
refresh_pw_cache()
log.info("[STARTUP] Scheduling autostart task")
asyncio.create_task(_autostart_from_settings())
# I2C init, ADC setup and the autostart task must run on the BLE loop so
# they share the same event loop as the Bumble HCI transport.
log.info("[STARTUP] Bridging I2C init and autostart to BLE loop")
asyncio.run_coroutine_threadsafe(_ble_startup(), _ble_loop)
@app.get("/audio_inputs_pw_usb")
async def audio_inputs_pw_usb():
@@ -1028,6 +1205,9 @@ async def refresh_audio_devices():
@app.post("/shutdown")
async def shutdown():
"""Stops broadcasting and releases all audio/Bluetooth resources."""
return await _on_ble_loop(_shutdown_impl())
async def _shutdown_impl():
try:
await _stop_all()
return {"status": "stopped"}
@@ -1040,6 +1220,9 @@ async def system_reboot():
Requires the service user to have passwordless sudo permissions to run 'reboot'.
"""
return await _on_ble_loop(_system_reboot_impl())
async def _system_reboot_impl():
try:
# Best-effort: stop any active streaming cleanly WITHOUT persisting state
try:
@@ -1063,47 +1246,27 @@ async def system_reboot():
@app.post("/restart_dep")
async def restart_dep():
"""Restart DEP by running dep.sh stop then dep.sh start in the dep directory.
"""Restart DEP via systemctl restart dep.service.
Requires the service user to have passwordless sudo permissions to run dep.sh.
Requires the service user to have passwordless sudo permissions for systemctl.
"""
try:
# Get the dep directory path (dep.sh is in dante_package subdirectory)
dep_dir = os.path.join(os.path.dirname(__file__), '..', '..', 'dep', 'dante_package')
# Run dep.sh stop first
log.info("Stopping DEP...")
stop_process = await asyncio.create_subprocess_exec(
"sudo", "bash", "dep.sh", "stop",
cwd=dep_dir,
log.info("Restarting DEP via systemctl...")
proc = await asyncio.create_subprocess_exec(
"sudo", "systemctl", "restart", "dep.service",
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
stop_stdout, stop_stderr = await stop_process.communicate()
if stop_process.returncode != 0:
error_msg = stop_stderr.decode() if stop_stderr else "Unknown error"
log.error(f"Failed to stop DEP: {error_msg}")
raise HTTPException(status_code=500, detail=f"Failed to stop DEP: {error_msg}")
# Run dep.sh start after stop succeeds
log.info("Starting DEP...")
start_process = await asyncio.create_subprocess_exec(
"sudo", "bash", "dep.sh", "start",
cwd=dep_dir,
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
start_stdout, start_stderr = await start_process.communicate()
if start_process.returncode == 0:
stdout, stderr = await proc.communicate()
if proc.returncode == 0:
log.info("DEP restarted successfully")
return {"status": "success", "message": "DEP restarted successfully"}
else:
error_msg = start_stderr.decode() if start_stderr else "Unknown error"
log.error(f"Failed to start DEP: {error_msg}")
raise HTTPException(status_code=500, detail=f"Failed to start DEP: {error_msg}")
error_msg = stderr.decode() if stderr else "Unknown error"
log.error(f"Failed to restart DEP: {error_msg}")
raise HTTPException(status_code=500, detail=f"Failed to restart DEP: {error_msg}")
except HTTPException:
raise
except Exception as e:
@@ -1196,6 +1359,9 @@ async def check_update():
@app.post("/system_update")
async def system_update():
"""Update application: git pull main branch (latest tag), poetry install, restart services."""
return await _on_ble_loop(_system_update_impl())
async def _system_update_impl():
try:
# Best-effort: stop any active streaming cleanly
try:
@@ -1245,26 +1411,12 @@ 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)
@@ -1677,5 +1829,170 @@ if __name__ == '__main__':
level=os.environ.get('LOG_LEVEL', log.INFO),
format='%(module)s.py:%(lineno)d %(levelname)s: %(message)s'
)
# ── GIL switch interval ─────────────────────────────────────────────────
# CPython releases the GIL every sys.getswitchinterval() seconds (default
# 5 ms). The audio pipeline fires every 10 ms, so a 5 ms granularity
# means up to half a frame period can be wasted waiting for the GIL.
# Reducing to 1 ms gives the BLE thread much tighter access.
import sys
sys.setswitchinterval(0.001)
log.info("GIL switch interval set to 1 ms")
# ── BLE / audio event loop ──────────────────────────────────────────────
# Bumble (serial_asyncio / HCI) and the audio pipeline run exclusively on
# this loop. Uvicorn's HTTP accept/read/write callbacks run on a separate
# asyncio loop in the main thread, so they can never stall BLE advertising
# or audio encoding.
#
# Route handlers that touch Bumble objects call _on_ble_loop(), which uses
# asyncio.run_coroutine_threadsafe() + asyncio.wrap_future() to submit the
# coroutine to _ble_loop and await the result back in uvicorn's loop.
# Hot-path read-only endpoints (/status, /audio_level*) access
# multicaster state directly Python's GIL makes attribute reads safe.
def _pthread_sched_lib():
"""Return a ctypes handle with correctly typed pthread scheduling symbols.
Uses RTLD_DEFAULT (ctypes.CDLL(None)) to resolve symbols from all
currently loaded shared libraries. This handles both:
- glibc < 2.34: pthread_self/pthread_setschedparam live in libpthread.so.0
- glibc >= 2.34: pthreads merged into libc.so.6
using find_library("c") would miss libpthread on older glibc and cause
a NULL function pointer → SEGV when called.
Explicit restype/argtypes are mandatory: pthread_t is c_ulong (64-bit
on ARM64/x86-64) but ctypes defaults to c_int (32-bit), truncating
the thread handle and causing a SEGV inside pthread_setschedparam.
"""
import ctypes
SCHED_FIFO = 1
SCHED_OTHER = 0
class SchedParam(ctypes.Structure):
_fields_ = [("sched_priority", ctypes.c_int)]
lib = ctypes.CDLL(None, use_errno=True) # RTLD_DEFAULT
lib.pthread_self.restype = ctypes.c_ulong
lib.pthread_self.argtypes = []
lib.pthread_getschedparam.restype = ctypes.c_int
lib.pthread_getschedparam.argtypes = [
ctypes.c_ulong,
ctypes.POINTER(ctypes.c_int),
ctypes.POINTER(SchedParam),
]
lib.pthread_setschedparam.restype = ctypes.c_int
lib.pthread_setschedparam.argtypes = [
ctypes.c_ulong,
ctypes.c_int,
ctypes.POINTER(SchedParam),
]
return lib, SchedParam, SCHED_FIFO, SCHED_OTHER
def _configure_ble_thread_scheduling():
"""Confirm or establish SCHED_FIFO for the BLE/audio thread.
When launched via the systemd unit (CPUSchedulingPolicy=fifo), new
threads inherit the process RT policy automatically just log and
return. When run directly (development), attempt to elevate to
SCHED_FIFO/30 (requires CAP_SYS_NICE), falling back gracefully.
"""
import ctypes
try:
lib, SchedParam, SCHED_FIFO, _ = _pthread_sched_lib()
tid = lib.pthread_self()
policy = ctypes.c_int(-1)
param = SchedParam(0)
lib.pthread_getschedparam(tid, ctypes.byref(policy), ctypes.byref(param))
if policy.value == SCHED_FIFO:
log.info("[BLE-LOOP] Already SCHED_FIFO priority=%d (inherited from systemd)",
param.sched_priority)
return
param.sched_priority = 30
ret = lib.pthread_setschedparam(tid, SCHED_FIFO, ctypes.byref(param))
if ret == 0:
log.info("[BLE-LOOP] SCHED_FIFO priority=30 set")
else:
err = ctypes.get_errno()
log.warning("[BLE-LOOP] SCHED_FIFO failed (errno=%d: %s) "
"use systemd CPUSchedulingPolicy=fifo or grant CAP_SYS_NICE",
err, os.strerror(err))
try:
os.setpriority(os.PRIO_PROCESS, 0,
os.getpriority(os.PRIO_PROCESS, 0) - 5)
except PermissionError:
pass
except Exception as exc:
log.warning("[BLE-LOOP] Scheduling setup error: %s", exc)
def _configure_http_thread_scheduling():
"""Demote the HTTP (uvicorn) thread to SCHED_OTHER + nice=+10.
When systemd sets CPUSchedulingPolicy=fifo, every thread in the
process including uvicorn's main loop inherits SCHED_FIFO.
We demote the HTTP thread back to SCHED_OTHER so the BLE thread
always wins CPU arbitration when both are runnable.
Lowering scheduling policy never requires special privileges.
"""
import ctypes
try:
lib, SchedParam, SCHED_FIFO, SCHED_OTHER = _pthread_sched_lib()
tid = lib.pthread_self()
policy = ctypes.c_int(-1)
param = SchedParam(0)
lib.pthread_getschedparam(tid, ctypes.byref(policy), ctypes.byref(param))
if policy.value == SCHED_FIFO:
param.sched_priority = 0
ret = lib.pthread_setschedparam(tid, SCHED_OTHER, ctypes.byref(param))
if ret == 0:
log.info("[HTTP] Demoted SCHED_FIFO → SCHED_OTHER")
else:
err = ctypes.get_errno()
log.warning("[HTTP] Could not demote from SCHED_FIFO (errno=%d)", err)
else:
log.info("[HTTP] Already SCHED_OTHER, no demotion needed")
except Exception as exc:
log.warning("[HTTP] Scheduling demotion error: %s", exc)
try:
os.nice(10)
log.info("[HTTP] nice=+10 (lower priority)")
except Exception as exc:
log.debug("[HTTP] os.nice: %s", exc)
_ble_loop_ready = threading.Event()
def _run_ble_loop():
# Confirm or establish RT scheduling before entering the event loop.
_configure_ble_thread_scheduling()
async def _ble_runner():
global _ble_loop
_ble_loop = asyncio.get_running_loop()
_ble_loop_ready.set()
# Keep the loop alive; it is stopped when the process exits because
# this is a daemon thread.
await asyncio.Event().wait()
asyncio.run(_ble_runner())
_ble_thread = threading.Thread(target=_run_ble_loop, name="ble-loop", daemon=True)
_ble_thread.start()
if not _ble_loop_ready.wait(timeout=5):
log.error("BLE event loop failed to start within 5 s aborting")
raise RuntimeError("BLE event loop startup timeout")
log.info("BLE event loop started on thread '%s'", _ble_thread.name)
# ── HTTP / uvicorn event loop (main thread) ─────────────────────────────
# Demote the HTTP thread from SCHED_FIFO (if set by systemd) to
# SCHED_OTHER + nice=+10 so the BLE thread always preempts it.
_configure_http_thread_scheduling()
# Bind to localhost only for security: prevents network access, only frontend on same machine can connect
uvicorn.run(app, host="127.0.0.1", port=5000, access_log=False)
+90
View File
@@ -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"
@@ -0,0 +1 @@
Placeholder file to have the activation folder available, otherwise dante activation script fails with 'Unable to write'.
@@ -1,5 +1,5 @@
{
"trialMode": true,
"trialMode": false,
"$schema": "./dante.json_schema.json",
"platform":
{
@@ -16,7 +16,7 @@
48000
],
"samplesPerPeriod" : 16,
"periodsPerBuffer" : 300,
"periodsPerBuffer" : 150,
"networkLatencyMinMs" : 2,
"networkLatencyDefaultMs" : 5,
"supportedEncodings" :
@@ -24,7 +24,10 @@
"PCM16"
],
"defaultEncoding" : "PCM16",
"numDepCores" : 1
"numDepCores" :
[
3
]
},
"network" :
{
@@ -50,31 +53,32 @@
"alsaAsrc":
{
"enableAlsaAsrc": true,
"cpuAffinity": 3,
"deviceConfigurations": [
{
"deviceIdentifier": "hw:0,0",
"deviceIdentifier": "hw:6,0,0",
"direction": "playback",
"bitDepth": 16,
"numOpenChannels": 6,
"alsaChannelRange": "0-5",
"danteChannelRange": "0-5",
"bufferSize": 4800,
"bufferSize": 960,
"samplesPerPeriod": 16
}
]
},
"product" :
{
"manfId" : "Audinate",
"manfName" : "Audinate Pty Ltd",
"modelId" : "OEMDEP",
"modelName" : "Linux Dante Embedded Platform",
"manfId" : "SummitFC",
"manfName" : "Summitwave FlexCo",
"modelId" : "TX",
"modelName" : "Summitwave TX",
"modelVersion" :
{
"major" : 9,
"minor" : 9,
"bugfix" : 99
"major" : 1,
"minor" : 0,
"bugfix" : 0
},
"devicePrefix" : "DEP"
"devicePrefix" : "SW-TX"
}
}
+4 -4
View File
@@ -6,8 +6,8 @@ pcm.ch1 {
channels 2
rate 48000
format S16_LE
period_size 120
buffer_size 240
period_size 240
buffer_size 960
}
bindings.0 0
}
@@ -21,8 +21,8 @@ pcm.ch2 {
channels 2
rate 48000
format S16_LE
period_size 120
buffer_size 240
period_size 240
buffer_size 960
}
bindings.0 1
}
+2 -1
View File
@@ -1 +1,2 @@
sudo cp src/misc/asound.conf /etc/asound.conf
SCRIPT_DIR="$(cd "$(dirname "$0")" && pwd)"
sudo cp "$SCRIPT_DIR/asound.conf" /etc/asound.conf
+56
View File
@@ -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."
+13111
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File diff suppressed because it is too large Load Diff
+5
View File
@@ -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
+5
View File
@@ -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
+83
View File
@@ -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
+2
View File
@@ -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
+2
View File
@@ -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
+5 -2
View File
@@ -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
+13
View File
@@ -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
+2
View File
@@ -9,6 +9,8 @@ ExecStartPre=/bin/sh -lc 'for i in $(seq 1 60); do ip route show default >/dev/n
ExecStart=/usr/bin/pipewire-aes67 -c /home/caster/bumble-auracast/src/service/aes67/pipewire-aes67.conf
Restart=always
RestartSec=5s
AllowedCPUs=0
CPUAffinity=0
# Avoid StartLimitHit on quick failures during boot; let RestartSec handle pacing
StartLimitIntervalSec=0
+2
View File
@@ -6,6 +6,8 @@ After=network.target
Type=simple
ExecStart=/usr/sbin/ptp4l -i eth0 -f /home/caster/bumble-auracast/src/service/aes67/ptp_aes67_1.conf
Restart=on-failure
AllowedCPUs=0
CPUAffinity=0
StandardOutput=journal
StandardError=journal
+22 -19
View File
@@ -14,24 +14,22 @@ while IFS=: read -r name type; do
fi
done < <(nmcli -t -f NAME,TYPE connection show)
# Configure Avahi to prefer DHCP address over static fallback for mDNS
# Get the DHCP-assigned IP (first non-localhost, non-192.168.42.10 IP)
DHCP_IP=$(ip -4 addr show eth0 2>/dev/null | grep -oP '(?<=inet\s)\d+(\.\d+){3}' | grep -v '^127\.' | grep -v '^169\.254\.' | head -n1)
HOSTNAME=$(hostname)
if [ -n "$DHCP_IP" ]; then
echo "DHCP address detected: $DHCP_IP, configuring Avahi to prefer it for mDNS."
# Add entry to /etc/avahi/hosts to explicitly map hostname to DHCP IP
sudo mkdir -p /etc/avahi
echo "$DHCP_IP $HOSTNAME $HOSTNAME.local" | sudo tee /etc/avahi/hosts > /dev/null
# Restart avahi to apply the hosts file
sudo systemctl restart avahi-daemon
else
echo "No DHCP address detected, mDNS will use link local"
# Remove hosts file to let Avahi advertise all IPs
sudo rm -f /etc/avahi/hosts
sudo systemctl restart avahi-daemon
fi
# 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
@@ -40,20 +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