34 Commits

Author SHA1 Message Date
4d8418f670 Merge branch 'main' into feature/1khz_testtone 2026-04-10 07:55:36 +00:00
0a8dc74d5c Fixes script error in systemupdate. 2026-04-09 15:00:35 +02:00
8475e4d068 New system update logic. 2026-04-09 14:46:17 +02:00
03d54eaddf Replace in-memory LC3 file loading with streaming byte generator to reduce memory usage
Refactors LC3 file handling to stream frames byte-by-byte from disk instead of loading the entire file into memory. Adds _lc3_file_byte_gen generator function that skips the 18-byte LC3 header, reads frame size headers, and yields individual bytes with optional looping. Removes read_lc3_file usage and itertools.cycle approach in favor of the new streaming generator.
2026-04-09 14:30:13 +02:00
5f7fd1c0ff Fix LC3 file reading to return individual frames instead of concatenated stream
Modifies read_lc3_file to collect LC3 frames in a list before joining, and updates multicast streamer to write each LC3 frame to all BIS queues in the BIG. Previously the entire LC3 file was concatenated into a single byte string without frame boundaries.
2026-04-09 14:05:46 +02:00
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
f82de89ce3 Replace WAV test files with LC3 format, remove unused announcement files
Updates all audio source references in config and demo mode from WAV to LC3 format. Removes obsolete WAV test files including wave_particle samples in multiple languages (de, en, es, fr, it, pl) at various sample rates, test tones, and unused announcement files. Refactors demo content handling to use kwargs for stream name and program info configuration.
2026-04-09 13:52:56 +02:00
e35b8aa2f9 Set stream name and program info for test tone broadcasts, update test tone WAV files 2026-04-09 13:47:48 +02:00
pober
67992e65ec Updates poetry lock. 2026-04-09 11:59:30 +02:00
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
85532b034c Add demo content selector for 1 kHz test tone option
Adds a selectbox in Demo mode UI to choose between program material and a 1 kHz test tone. Includes test tone WAV files at 16/24/48 kHz sample rates. The server detects and persists the demo content type based on the selected audio source files.
2026-04-08 09:28:28 +02:00
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
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
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
pober
e1d717ed5c Adds DHCP/static IP toggle for both ports in the UI. 2026-03-03 15:50:19 +01:00
pober
540d8503ac Some corrections for Activates link local for both ports, removes fallback IP. 2026-03-03 15:35:13 +01:00
pober
c82f375539 Activates link local for both ports, removes fallback IP. 2026-03-03 15:02:55 +01:00
70bde5295f Fixes mDNS issue; when DHCP IP is present use this for mDNS and not the static fallback IP. 2026-02-16 16:25:59 +01:00
f5f93b4b8e analog_input_gain (#21)
- add input boost slider
- add level meter

Reviewed-on: https://gitea.pstruebi.xyz/auracaster/bumble-auracast/pulls/21
2026-02-12 17:09:46 +01:00
3322b9edf4 add 192.168.42.10 as default ip with update script 2026-02-12 17:08:23 +01:00
d6230e7522 add software gain boost parameter for input signal amplification 2026-02-12 13:30:07 +01:00
f2382470d8 add network information display showing hostname and IP address 2026-02-10 16:51:22 +01:00
7c2f0bf0cb add HTTP to HTTPS redirect server on port 80 2026-02-10 16:37:34 +01:00
184e9c84af impelement a gain slider 2026-01-20 18:00:37 +01:00
6852c74cd0 add a delete recordings button 2026-01-20 17:45:23 +01:00
81 changed files with 14261 additions and 187 deletions

3
.gitignore vendored
View File

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

View File

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

79
poetry.lock generated
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@@ -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"},
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]
[[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"

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]

View File

@@ -62,7 +62,7 @@ class AuracastBigConfigDeu(AuracastBigConfig):
name: str = 'Hörsaal A'
language: str ='deu'
program_info: str = 'Vorlesung DE'
audio_source: str = 'file:./testdata/wave_particle_5min_de.wav'
audio_source: str = 'file:./testdata/wave_particle_5min_de.lc3'
class AuracastBigConfigEng(AuracastBigConfig):
id: int = 123
@@ -70,7 +70,7 @@ class AuracastBigConfigEng(AuracastBigConfig):
name: str = 'Lecture Hall A'
language: str ='eng'
program_info: str = 'Lecture EN'
audio_source: str = 'file:./testdata/wave_particle_5min_en.wav'
audio_source: str = 'file:./testdata/wave_particle_5min_en.lc3'
class AuracastBigConfigFra(AuracastBigConfig):
id: int = 1234
@@ -79,7 +79,7 @@ class AuracastBigConfigFra(AuracastBigConfig):
name: str = 'Auditoire A'
language: str ='fra'
program_info: str = 'Auditoire FR'
audio_source: str = 'file:./testdata/wave_particle_5min_fr.wav'
audio_source: str = 'file:./testdata/wave_particle_5min_fr.lc3'
class AuracastBigConfigSpa(AuracastBigConfig):
id: int =12345
@@ -87,7 +87,7 @@ class AuracastBigConfigSpa(AuracastBigConfig):
name: str = 'Auditorio A'
language: str ='spa'
program_info: str = 'Auditorio ES'
audio_source: str = 'file:./testdata/wave_particle_5min_es.wav'
audio_source: str = 'file:./testdata/wave_particle_5min_es.lc3'
class AuracastBigConfigIta(AuracastBigConfig):
id: int =1234567
@@ -95,7 +95,7 @@ class AuracastBigConfigIta(AuracastBigConfig):
name: str = 'Aula A'
language: str ='ita'
program_info: str = 'Aula IT'
audio_source: str = 'file:./testdata/wave_particle_5min_it.wav'
audio_source: str = 'file:./testdata/wave_particle_5min_it.lc3'
class AuracastBigConfigPol(AuracastBigConfig):
@@ -104,10 +104,12 @@ class AuracastBigConfigPol(AuracastBigConfig):
name: str = 'Sala Wykładowa'
language: str ='pol'
program_info: str = 'Sala Wykładowa PL'
audio_source: str = 'file:./testdata/wave_particle_5min_pl.wav'
audio_source: str = 'file:./testdata/wave_particle_5min_pl.lc3'
class AuracastConfigGroup(AuracastGlobalConfig):
bigs: List[AuracastBigConfig] = [
AuracastBigConfigDeu(),
]
analog_gain_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)

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 = 40
if avail < SETPOINT - TOLERANCE:
self._bang_bang = -1
elif avail > SETPOINT + TOLERANCE:
self._bang_bang = 1
else:
self._bang_bang = 0
logging.debug("PyALSA: read length=%d, data length=%d, avail=%d, bang_bang=%d", length, len(data), avail, self._bang_bang)
#if self.counter % 50 == 0:
frame_size = frame_size + 1 # consume samples a little faster to avoid latency akkumulation
if length > 0:
if self._first_read:
expected_mono = self._periodsize * 2
expected_stereo = self._periodsize * 2 * 2
# self._hw_channels = 2 if len(data) == expected_stereo else 1
self._hw_channels = self._actual_channels
logging.info("PyALSA first read: bytes=%d detected_hw_channels=%d requested_channels=%d",
len(data), self._hw_channels, self._actual_channels)
self._first_read = False
if self._hw_channels == 2 and self._actual_channels == 1:
pcm_stereo = np.frombuffer(data, dtype=np.int16)
pcm_mono = pcm_stereo[::2]
data = pcm_mono.tobytes()
actual_samples = len(data) // (2 * self._actual_channels)
ratio = frame_size / actual_samples
pcm_f32 = np.frombuffer(data, dtype=np.int16).astype(np.float32) / 32768.0
if self._actual_channels > 1:
pcm_f32 = pcm_f32.reshape(-1, self._actual_channels)
resampled = self._resampler.process(pcm_f32, ratio, end_of_input=False)
if self._actual_channels > 1:
resampled = resampled.reshape(-1)
self._resampler_buffer = np.concatenate([self._resampler_buffer, resampled])
else:
logging.warning("PyALSA: No data read from ALSA")
self._resampler_buffer = np.concatenate([
self._resampler_buffer,
np.zeros(frame_size * self._actual_channels, dtype=np.float32),
])
except alsaaudio.ALSAAudioError as e:
logging.error("PyALSA: ALSA read error: %s", e)
self._resampler_buffer = np.concatenate([
self._resampler_buffer,
np.zeros(frame_size * self._actual_channels, dtype=np.float32),
])
except Exception as e:
logging.error("PyALSA: Unexpected error in _read: %s", e, exc_info=True)
self._resampler_buffer = np.concatenate([
self._resampler_buffer,
np.zeros(frame_size * self._actual_channels, dtype=np.float32),
])
pcm_buffer, overflowed = self._stream.read(frame_size)
if overflowed:
logging.warning("SoundDeviceAudioInput: overflowed")
needed = frame_size * self._actual_channels
if len(self._resampler_buffer) < needed:
pad = np.zeros(needed - len(self._resampler_buffer), dtype=np.float32)
self._resampler_buffer = np.concatenate([self._resampler_buffer, pad])
logging.debug("PyALSA: padded buffer with %d samples", needed - len(self._resampler_buffer))
n_available = self._stream.read_available
output = self._resampler_buffer[:needed]
self._resampler_buffer = self._resampler_buffer[needed:]
# adapt = n_available > 20
# if adapt:
# pcm_extra, overflowed = self._stream.read(3)
# logging.info('consuming extra samples, available was %d', n_available)
# if overflowed:
# logging.warning("SoundDeviceAudioInput: overflowed")
# out = bytes(pcm_buffer) + bytes(pcm_extra)
# else:
out = bytes(pcm_buffer)
logging.debug("PyALSA: resampler_buffer remaining=%d", len(self._resampler_buffer))
return np.clip(output * 32767.0, -32768, 32767).astype(np.int16).tobytes()
self.max_avail = max(self.max_avail, n_available)
#Diagnostics
#with open(self.logfile_name, "a", encoding="utf-8") as f:
# f.write(f"{n_available}, {adapt}, {round(self._runavg, 2)}, {overflowed}\n")
def _close(self) -> None:
if self._pcm:
self._pcm.close()
self._pcm = None
if self.counter % 500 == 0:
logging.info(
"read available=%d, max=%d, latency:%d",
n_available, self.max_avail, self._stream.latency
)
self.max_avail = 0
self.counter += 1
return out
audio_io.SoundDeviceAudioInput = ModSoundDeviceAudioInput
audio_io.SoundDeviceAudioInput = PyAlsaAudioInput
# modified from bumble
class ModWaveAudioInput(audio_io.ThreadedAudioInput):
@@ -538,7 +589,7 @@ async def init_broadcast(
def on_flow():
data_packet_queue = iso_queue.data_packet_queue
print(
logging.info(
f'\rPACKETS: pending={data_packet_queue.pending}, '
f'queued={data_packet_queue.queued}, '
f'completed={data_packet_queue.completed}',
@@ -551,6 +602,29 @@ async def init_broadcast(
return bigs
def _lc3_file_byte_gen(filename: str, loop: bool = False):
"""Stream LC3 frames from disk as individual bytes, with optional looping.
Yields one byte (int) at a time so it is compatible with the existing
``bytes(itertools.islice(gen, bytes_per_frame))`` consumer without loading
the whole file into memory.
"""
while True:
with open(filename, 'rb') as f:
f.read(18) # skip 18-byte LC3 header
while True:
size_b = f.read(2)
if len(size_b) < 2:
break
frame_size = struct.unpack('=H', size_b)[0]
frame = f.read(frame_size)
if len(frame) < frame_size:
break
yield from frame
if not loop:
return
class Streamer():
"""
Streamer class that supports multiple input formats. See bumble for streaming from wav or device
@@ -638,6 +712,12 @@ class Streamer():
except Exception:
pass
def get_audio_levels(self) -> list[float]:
"""Return current RMS audio levels (0.0-1.0) for each BIG."""
if not self.bigs:
return []
return [big.get('_audio_level_rms', 0.0) for big in self.bigs.values()]
async def stream(self):
bigs = self.bigs
@@ -700,13 +780,7 @@ class Streamer():
big['precoded'] = True
big['lc3_bytes_per_frame'] = global_config.octets_per_frame
filename = big_config[i].audio_source.replace('file:', '')
lc3_bytes = read_lc3_file(filename)
lc3_frames = iter(lc3_bytes)
if big_config[i].loop:
lc3_frames = itertools.cycle(lc3_frames)
big['lc3_frames'] = lc3_frames
big['lc3_frames'] = _lc3_file_byte_gen(filename, loop=big_config[i].loop)
# use wav files and code them entirely before streaming
elif big_config[i].precode_wav and big_config[i].audio_source.endswith('.wav'):
@@ -823,6 +897,9 @@ class Streamer():
if lc3_frame == b'': # Not all streams may stop at the same time
stream_finished[i] = True
continue
for q_idx in range(big.get('num_bis', 1)):
await big['iso_queues'][q_idx].write(lc3_frame)
else: # code lc3 on the fly with perf counters
# Ensure frames generator exists (so we can aclose() on stop)
frames_gen = big.get('frames_gen')
@@ -852,6 +929,11 @@ class Streamer():
stream_finished[i] = True
continue
# Compute RMS audio level (normalized 0.0-1.0) for level monitoring
pcm_samples = np.frombuffer(pcm_frame, dtype=np.int16).astype(np.float32)
rms = np.sqrt(np.mean(pcm_samples ** 2)) / 32768.0 if len(pcm_samples) > 0 else 0.0
big['_audio_level_rms'] = float(rms)
# Measure LC3 encoding time
t1 = time.perf_counter()
num_bis = big.get('num_bis', 1)

View File

@@ -37,6 +37,12 @@ class Multicaster:
'is_initialized': self.is_auracast_init,
'is_streaming': streaming,
}
def get_audio_levels(self) -> list[float]:
"""Return current RMS audio levels (0.0-1.0) for each BIG."""
if self.streamer is not None and self.streamer.is_streaming:
return self.streamer.get_audio_levels()
return []
async def init_broadcast(self):
self.device_acm = multicast.create_device(self.global_conf)
@@ -137,6 +143,10 @@ async def main():
level=os.environ.get('LOG_LEVEL', logging.DEBUG),
format='%(module)s.py:%(lineno)d %(levelname)s: %(message)s'
)
# Enable debug logging for bumble
# logging.getLogger('bumble').setLevel(logging.DEBUG)
os.chdir(os.path.dirname(__file__))
global_conf = auracast_config.AuracastGlobalConfig(

View File

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

View File

@@ -1,6 +1,7 @@
# frontend/app.py
import os
import time
import math
import logging as log
from PIL import Image
@@ -196,6 +197,116 @@ if audio_mode == "Demo":
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_db_left = 0 # default (dB)
analog_gain_db_right = 0 # default (dB)
if audio_mode == "Analog":
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,
help="Show real-time audio level meters for active radios. Only works while streaming.")
if show_level_monitor and is_streaming:
@st.fragment(run_every=0.2)
def _audio_level_fragment():
cols = st.columns(2)
# Radio 1
with cols[0]:
try:
r = requests.get(f"{BACKEND_URL}/audio_level", timeout=0.2)
levels = r.json().get("levels", []) if r.ok else []
except Exception:
levels = []
if levels:
rms = max(levels)
db = max(-60.0, 20.0 * (math.log10(rms) if rms > 0 else -3.0))
pct = int(max(0, min(100, (db + 60) * 100 / 60)))
st.markdown(
f"**Radio 1**"
f'<div style="background:#333;border-radius:4px;height:18px;width:100%;margin-top:4px;">'
f'<div style="background:#2ecc71;height:100%;width:{pct}%;border-radius:4px;transition:width 0.15s;"></div>'
f'</div>',
unsafe_allow_html=True,
)
else:
st.markdown("**Radio 1** &nbsp; --")
# Radio 2
with cols[1]:
try:
r2 = requests.get(f"{BACKEND_URL}/audio_level2", timeout=0.2)
levels2 = r2.json().get("levels", []) if r2.ok else []
except Exception:
levels2 = []
if levels2:
rms2 = max(levels2)
db2 = max(-60.0, 20.0 * (math.log10(rms2) if rms2 > 0 else -3.0))
pct2 = int(max(0, min(100, (db2 + 60) * 100 / 60)))
st.markdown(
f"**Radio 2**"
f'<div style="background:#333;border-radius:4px;height:18px;width:100%;margin-top:4px;">'
f'<div style="background:#2ecc71;height:100%;width:{pct2}%;border-radius:4px;transition:width 0.15s;"></div>'
f'</div>',
unsafe_allow_html=True,
)
else:
st.markdown("**Radio 2** &nbsp; --")
_audio_level_fragment()
# Placeholder for validation errors (will be filled in later)
validation_error_placeholder = st.empty()
@@ -240,6 +351,17 @@ if audio_mode == "Demo":
disabled=is_streaming,
help="Select the demo stream configuration."
)
demo_content_options = ["Program material", "1 kHz test tone"]
saved_demo_content = saved_settings.get('demo_content', 'Program material')
if saved_demo_content not in demo_content_options:
saved_demo_content = 'Program material'
demo_content = st.selectbox(
"Demo Content",
demo_content_options,
index=demo_content_options.index(saved_demo_content),
disabled=is_streaming,
help="Select whether demo streams use program audio files or a continuous 1 kHz test tone."
)
# Stream password and flags (same as USB/AES67)
saved_pwd = saved_settings.get('stream_password', '') or ''
stream_passwort = st.text_input(
@@ -602,6 +724,8 @@ else:
'immediate_rendering': immediate_rendering2,
'presentation_delay_ms': presentation_delay_ms2,
'qos_preset': qos_preset2,
'analog_gain_db_left': analog_gain_db_left,
'analog_gain_db_right': analog_gain_db_right,
}
radio1_cfg = {
@@ -616,7 +740,9 @@ else:
'immediate_rendering': immediate_rendering1,
'presentation_delay_ms': presentation_delay_ms1,
'qos_preset': qos_preset1,
'stereo_mode': stereo_enabled, # Add stereo mode setting
'stereo_mode': stereo_enabled,
'analog_gain_db_left': analog_gain_db_left,
'analog_gain_db_right': analog_gain_db_right,
}
if audio_mode == "Network - Dante":
@@ -1422,12 +1548,22 @@ if start_stream:
bigs1 = []
for i in range(demo_cfg['streams']):
cfg_cls, lang = lang_cfgs[i % len(lang_cfgs)]
if demo_content == "1 kHz test tone":
source_file = f'../testdata/test_tone_1k_{int(q["rate"]/1000)}kHz_mono.lc3'
big_kwargs = {
'name': 'test tone',
'program_info': '1khz',
}
else:
source_file = f'../testdata/wave_particle_5min_{lang}_{int(q["rate"]/1000)}kHz_mono.lc3'
big_kwargs = {}
bigs1.append(cfg_cls(
code=(stream_passwort.strip() or None),
audio_source=f'file:../testdata/wave_particle_5min_{lang}_{int(q["rate"]/1000)}kHz_mono.wav',
audio_source=f'file:{source_file}',
iso_que_len=32,
sampling_frequency=q['rate'],
octets_per_frame=q['octets'],
**big_kwargs,
))
max_per_mc = {48000: 1, 24000: 2, 16000: 3}
@@ -1499,9 +1635,10 @@ if start_stream:
immediate_rendering=bool(cfg['immediate_rendering']),
presentation_delay_us=int(cfg['presentation_delay_ms'] * 1000),
qos_config=QOS_PRESET_MAP[cfg['qos_preset']],
analog_gain_db_left=cfg.get('analog_gain_db_left', 0.0),
analog_gain_db_right=cfg.get('analog_gain_db_right', 0.0),
bigs=[
auracast_config.AuracastBigConfig(
id=cfg.get('id', 123456),
code=(cfg['stream_passwort'].strip() or None),
name=cfg['name'],
program_info=cfg['program_info'],
@@ -1555,8 +1692,6 @@ 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_')
@@ -1564,7 +1699,6 @@ 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'],
@@ -1667,7 +1801,21 @@ 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:
@@ -1682,6 +1830,123 @@ with st.expander("System control", expanded=False):
except Exception as e:
st.warning(f"Could not read temperatures: {e}")
st.subheader("Network Information")
try:
import subprocess, socket
device_hostname = socket.gethostname()
st.write(f"Hostname: **{device_hostname}**")
network_info_resp = requests.get(f"{BACKEND_URL}/network_info", timeout=5)
if network_info_resp.status_code == 200:
network_data = network_info_resp.json()
interfaces = network_data.get("interfaces", {})
port_mapping = network_data.get("port_mapping", {})
for port_name in ["port1", "port2"]:
if port_name not in port_mapping:
continue
interface_name = port_mapping[port_name]
interface_data = interfaces.get(interface_name, {})
port_label = "Port 1" if port_name == "port1" else "Port 2"
st.markdown(f"### {port_label}")
ip_address = interface_data.get("ip_address", "N/A")
is_dhcp = interface_data.get("is_dhcp", True)
st.write(f"Interface: **{interface_name}**")
st.write(f"IP Address: **{ip_address}**")
col1, col2 = st.columns([1, 3])
with col1:
toggle_key = f"{port_name}_dhcp_toggle"
current_mode = "DHCP" if is_dhcp else "Static IP"
new_mode = st.radio(
"Mode",
options=["DHCP", "Static IP"],
index=0 if is_dhcp else 1,
key=toggle_key,
horizontal=True
)
with col2:
if new_mode == "Static IP":
ip_input_key = f"{port_name}_ip_input"
default_ip = ip_address if ip_address != "N/A" and not is_dhcp else ""
new_ip = st.text_input(
"Static IP Address",
value=default_ip,
key=ip_input_key,
placeholder="192.168.1.100"
)
if st.button(f"Apply", key=f"{port_name}_apply_btn"):
if not new_ip:
st.error("Please enter an IP address")
else:
import re
ip_pattern = re.compile(r'^(\d{1,3}\.){3}\d{1,3}$')
if not ip_pattern.match(new_ip):
st.error("Invalid IP address format")
else:
octets = new_ip.split('.')
if not all(0 <= int(octet) <= 255 for octet in octets):
st.error("IP address octets must be between 0 and 255")
else:
try:
config_payload = {
"interface": interface_name,
"is_dhcp": False,
"ip_address": new_ip,
"netmask": "24"
}
config_resp = requests.post(
f"{BACKEND_URL}/set_network_config",
json=config_payload,
timeout=10
)
if config_resp.status_code == 200:
st.success(f"Static IP {new_ip} applied to {interface_name}")
time.sleep(2)
st.rerun()
else:
st.error(f"Failed to apply configuration: {config_resp.text}")
except Exception as e:
st.error(f"Error applying configuration: {e}")
else:
if new_mode != current_mode:
if st.button(f"Apply DHCP", key=f"{port_name}_dhcp_apply_btn"):
try:
config_payload = {
"interface": interface_name,
"is_dhcp": True
}
config_resp = requests.post(
f"{BACKEND_URL}/set_network_config",
json=config_payload,
timeout=10
)
if config_resp.status_code == 200:
st.success(f"DHCP enabled for {interface_name}")
time.sleep(2)
st.rerun()
else:
st.error(f"Failed to apply configuration: {config_resp.text}")
except Exception as e:
st.error(f"Error applying configuration: {e}")
st.markdown("---")
else:
result = subprocess.run(["hostname", "-I"], capture_output=True, text=True, timeout=2)
ips = [ip for ip in result.stdout.strip().split() if not ip.startswith('127.') and ':' not in ip]
if ips:
st.write(f"IP Address: **{ips[0]}**")
else:
st.warning("No valid IP address found.")
except Exception as e:
st.warning(f"Could not determine network info: {e}")
st.subheader("CA Certificate")
st.caption("Download the CA certificate to trust this device's HTTPS connection.")
try:
@@ -1842,7 +2107,7 @@ with st.expander("Record", expanded=False):
selected_device_name = None
# Recording controls
col_record, col_download = st.columns([1, 1])
col_record, col_download, col_delete = st.columns([1, 1, 1])
with col_record:
if st.button("Start Recording (5s)", disabled=not selected_device_name):
@@ -1879,6 +2144,22 @@ with st.expander("Record", expanded=False):
st.warning(f"Could not fetch recording: {e}")
else:
st.button("Download Last Recording", disabled=True, help="No recording available yet")
with col_delete:
if st.button("Delete Recordings", type="secondary"):
try:
r = requests.delete(f"{BACKEND_URL}/delete_recordings", timeout=10)
if r.ok:
result = r.json()
if result.get('success'):
st.success(f"Deleted {result.get('deleted_count', 0)} recording(s)")
st.session_state['last_recording'] = None
else:
st.error("Failed to delete recordings")
else:
st.error(f"Failed to delete recordings: {r.status_code}")
except Exception as e:
st.error(f"Error deleting recordings: {e}")
log.basicConfig(
level=os.environ.get('LOG_LEVEL', log.DEBUG),

View File

@@ -3,6 +3,7 @@ TODO: in the future the multicaster objects should run in their own threads or e
"""
import os
import re
import logging as log
import json
from datetime import datetime
@@ -26,6 +27,57 @@ from auracast.utils.sounddevice_utils import (
)
load_dotenv()
# Blue LED on GPIO pin 12 (BCM) turns on while transmitting
LED_PIN = 12
try:
import RPi.GPIO as _GPIO
_GPIO.setmode(_GPIO.BCM)
_GPIO.setup(LED_PIN, _GPIO.OUT)
_GPIO_AVAILABLE = True
except Exception:
_GPIO_AVAILABLE = False
_GPIO = None # type: ignore
_LED_ENABLED: bool = True # toggled via /set_led_enabled
_LED_SETTINGS_FILE = os.path.join(os.path.dirname(__file__), 'led_settings.json')
def _load_led_settings() -> None:
global _LED_ENABLED
try:
if os.path.exists(_LED_SETTINGS_FILE):
with open(_LED_SETTINGS_FILE, 'r', encoding='utf-8') as f:
data = json.load(f)
_LED_ENABLED = bool(data.get('led_enabled', True))
except Exception:
_LED_ENABLED = True
def _save_led_settings() -> None:
try:
os.makedirs(os.path.dirname(_LED_SETTINGS_FILE), exist_ok=True)
with open(_LED_SETTINGS_FILE, 'w', encoding='utf-8') as f:
json.dump({'led_enabled': _LED_ENABLED}, f)
except Exception:
pass
def _led_on():
if _GPIO_AVAILABLE and _LED_ENABLED:
try:
_GPIO.output(LED_PIN, _GPIO.LOW)
except Exception:
pass
def _led_off():
if _GPIO_AVAILABLE:
try:
_GPIO.output(LED_PIN, _GPIO.HIGH)
except Exception:
pass
# Configure bumble debug logging
# log.getLogger('bumble').setLevel(log.DEBUG)
# make sure pipewire sets latency
# Primary and secondary persisted settings files
STREAM_SETTINGS_FILE1 = os.path.join(os.path.dirname(__file__), 'stream_settings.json')
@@ -133,6 +185,10 @@ def save_settings(persisted: dict, secondary: bool = False) -> None:
def gen_random_add() -> str:
return ':'.join(['%02X' % random.randint(0, 255) for _ in range(6)])
def gen_random_broadcast_id() -> int:
"""Generate a random 24-bit Broadcast ID (1..0xFFFFFF)."""
return random.randint(1, 0xFFFFFF)
app = FastAPI()
# Allow CORS for frontend on localhost
@@ -238,12 +294,18 @@ async def _init_i2c_on_startup() -> None:
log.warning("Exception running i2cget (%s): %s", " ".join(read_cmd), e, exc_info=True)
async def _set_adc_level_on_startup() -> None:
"""Ensure ADC mixer level is set at startup.
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 2 set 'ADC' x%
Runs: amixer -c 2 sset ADC {gain_db_left}dB,{gain_db_right}dB
Args:
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
"""
cmd = ["amixer", "-c", "2", "set", "ADC", "80%"]
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,
@@ -252,15 +314,46 @@ async def _set_adc_level_on_startup() -> 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:
@@ -280,6 +373,7 @@ async def _stop_all() -> bool:
was_running = True
finally:
multicaster2 = None
_led_off()
return was_running
async def _status_primary() -> dict:
@@ -338,6 +432,10 @@ 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_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:
@@ -380,18 +478,15 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
if is_stereo and sel == 'ch1':
# Stereo mode: use ALSA directly to capture both channels from hardware
# ch1=left (channel 0), ch2=right (channel 1)
big.audio_source = 'alsa:hw:CARD=i2s,DEV=0'
big.audio_source = 'device:hw:2'
big.input_format = f"int16le,{hardware_capture_rate},2"
log.info("Configured analog stereo input: using ALSA hw:CARD=i2s,DEV=0 with ch1=left, ch2=right")
elif is_stereo and sel == 'ch2':
# Skip ch2 in stereo mode as it's already captured as part of stereo pair
continue
else:
# Mono mode: individual channel capture
device_index = resolve_input_device_index(sel)
if device_index is None:
raise HTTPException(status_code=400, detail=f"Audio device '{sel}' not found.")
big.audio_source = f'device:{device_index}'
# Mono mode: use dsnoop virtual device directly (ch1=left, ch2=right)
big.audio_source = f'device:{sel}'
big.input_format = f"int16le,{hardware_capture_rate},1"
continue
@@ -447,10 +542,12 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
conf.qos_config.max_transport_latency_ms = int(conf.qos_config.number_of_retransmissions) * 10 + 3
# Only generate a new random_address if the BIG is still at the model default.
# Generate fresh random_address and broadcast ID for any BIG still at model defaults.
for big in conf.bigs:
if not getattr(big, 'random_address', None) or big.random_address == DEFAULT_RANDOM_ADDRESS:
big.random_address = gen_random_add()
if big.id == DEFAULT_BIG_ID:
big.id = gen_random_broadcast_id()
# Log the final, fully-updated configuration just before creating the Multicaster
log.info('Final multicaster config (transport=%s):\n %s', transport, conf.model_dump_json(indent=2))
@@ -461,11 +558,19 @@ 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:'))
demo_rate = int(conf.auracast_sampling_rate_hz or 0)
demo_type = None
demo_sources = [
str(b.audio_source)
for b in conf.bigs
if isinstance(b.audio_source, str) and b.audio_source.startswith('file:')
]
is_demo_tone = bool(demo_sources) and all('test_tone_1k_' in src for src in demo_sources)
demo_content = '1 kHz test tone' if is_demo_tone else 'Program material'
if demo_count > 0 and demo_rate > 0:
if demo_rate in (48000, 24000, 16000):
demo_type = f"{demo_count} × {demo_rate//1000}kHz"
@@ -485,13 +590,16 @@ 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_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],
'demo_total_streams': demo_count,
'demo_stream_type': demo_type,
'demo_content': demo_content,
'is_streaming': auto_started,
'demo_sources': [str(b.audio_source) for b in conf.bigs if isinstance(b.audio_source, str) and b.audio_source.startswith('file:')],
'demo_sources': demo_sources,
}
return mc, persisted
except HTTPException:
@@ -519,6 +627,16 @@ async def initialize2(conf: auracast_config.AuracastConfigGroup):
multicaster2 = mc
save_settings(persisted, secondary=True)
@app.post("/set_led_enabled")
async def set_led_enabled(body: dict):
"""Enable or disable the blue status LED. Persisted across restarts."""
global _LED_ENABLED
_LED_ENABLED = bool(body.get("led_enabled", True))
_save_led_settings()
if not _LED_ENABLED:
_led_off()
return {"led_enabled": _LED_ENABLED}
@app.post("/stop_audio")
async def stop_audio():
"""Stops streaming on both multicaster1 and multicaster2 (worker thread)."""
@@ -547,6 +665,28 @@ async def stop_audio():
log.error("Exception in /stop_audio: %s", traceback.format_exc())
raise HTTPException(status_code=500, detail=str(e))
@app.post("/adc_gain")
async def set_adc_gain(payload: dict):
"""Set ADC gain in dB for left and right channels without restarting the stream.
Body: {"gain_db_left": float, "gain_db_right": float}
"""
try:
gain_db_left = float(payload.get("gain_db_left", 0.0))
gain_db_right = float(payload.get("gain_db_right", 0.0))
await _set_adc_level(gain_db_left, gain_db_right)
# Persist the new values so they survive a restart
for load_fn, save_fn in [(load_stream_settings, save_stream_settings), (load_stream_settings2, save_stream_settings2)]:
s = load_fn() or {}
if s:
s['analog_gain_db_left'] = gain_db_left
s['analog_gain_db_right'] = gain_db_right
save_fn(s)
return {"status": "ok", "gain_db_left": gain_db_left, "gain_db_right": gain_db_right}
except Exception as e:
log.error("Exception in /adc_gain: %s", traceback.format_exc())
raise HTTPException(status_code=500, detail=str(e))
@app.post("/stream_lc3")
async def send_audio(audio_data: dict[str, str]):
"""Sends a block of pre-coded LC3 audio via the worker."""
@@ -582,9 +722,24 @@ 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
@app.get("/audio_level")
async def get_audio_level():
"""Return current RMS audio levels for primary radio (lightweight, for polling)."""
if multicaster1 is None:
return {"levels": []}
return {"levels": multicaster1.get_audio_levels()}
@app.get("/audio_level2")
async def get_audio_level2():
"""Return current RMS audio levels for secondary radio (lightweight, for polling)."""
if multicaster2 is None:
return {"levels": []}
return {"levels": multicaster2.get_audio_levels()}
async def _autostart_from_settings():
settings1 = load_stream_settings() or {}
settings2 = load_stream_settings2() or {}
@@ -737,6 +892,8 @@ 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_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
@@ -890,6 +1047,8 @@ 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_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"])
@@ -911,9 +1070,10 @@ async def _autostart_from_settings():
async def _startup_autostart_event():
# Spawn the autostart task without blocking startup
log.info("[STARTUP] Auracast multicast server startup: initializing settings cache, I2C, and PipeWire cache")
_led_off()
# Run install_asoundconf.sh script
script_path = os.path.join(os.path.dirname(__file__), '..', 'misc', 'install_asoundconf.sh')
script_path = os.path.join(os.path.dirname(__file__), '..', '..', 'misc', 'install_asoundconf.sh')
try:
log.info("[STARTUP] Running install_asoundconf.sh script")
result = subprocess.run(['bash', script_path], capture_output=True, text=True, check=True)
@@ -924,10 +1084,11 @@ async def _startup_autostart_event():
log.error(f"[STARTUP] Error running install_asoundconf.sh: {str(e)}")
# Hydrate settings cache once to avoid disk I/O during /status
_load_led_settings()
_init_settings_cache_from_disk()
await _init_i2c_on_startup()
# Ensure ADC mixer level is set at startup
await _set_adc_level_on_startup()
# Ensure ADC mixer level is set at startup (default 0 dB)
await _set_adc_level(0.0, 0.0)
refresh_pw_cache()
log.info("[STARTUP] Scheduling autostart task")
asyncio.create_task(_autostart_from_settings())
@@ -1218,26 +1379,12 @@ 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)
@@ -1413,6 +1560,235 @@ async def download_recording(filename: str):
raise HTTPException(status_code=500, detail=str(e))
@app.delete("/delete_recordings")
async def delete_recordings():
"""Delete all recordings in the recordings folder."""
try:
deleted_count = 0
for filename in os.listdir(RECORDINGS_DIR):
filepath = os.path.join(RECORDINGS_DIR, filename)
if os.path.isfile(filepath):
try:
os.remove(filepath)
deleted_count += 1
log.info("Deleted recording: %s", filename)
except Exception as e:
log.warning("Failed to delete %s: %s", filename, e)
log.info("Deleted %d recordings", deleted_count)
return {"success": True, "deleted_count": deleted_count}
except Exception as e:
log.error("Exception in /delete_recordings: %s", e, exc_info=True)
raise HTTPException(status_code=500, detail=str(e))
@app.get("/network_info")
async def get_network_info():
"""Get network information for all ethernet interfaces."""
try:
interfaces = {}
hardcoded_devices = ["eth0", "eth1"]
proc = await asyncio.create_subprocess_exec(
"nmcli", "-t", "-f", "NAME,DEVICE", "connection", "show",
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
stdout, stderr = await proc.communicate()
device_to_connection = {}
if proc.returncode == 0:
for line in stdout.decode().strip().split('\n'):
if not line:
continue
parts = line.split(':')
if len(parts) >= 2:
connection_name = parts[0]
device_name = parts[1]
if device_name in hardcoded_devices:
device_to_connection[device_name] = connection_name
for device in hardcoded_devices:
ip_address = None
proc = await asyncio.create_subprocess_exec(
"nmcli", "-t", "-f", "IP4.ADDRESS", "device", "show", device,
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
stdout, stderr = await proc.communicate()
if proc.returncode == 0:
for line in stdout.decode().strip().split('\n'):
if line.startswith('IP4.ADDRESS'):
ip_parts = line.split(':')
if len(ip_parts) >= 2:
full_ip = ip_parts[1]
ip_address = full_ip.split('/')[0]
if not ip_address.startswith('169.254.'):
break
method = "auto"
connection_name = device_to_connection.get(device)
if connection_name:
proc = await asyncio.create_subprocess_exec(
"nmcli", "-t", "-f", "ipv4.method", "connection", "show", connection_name,
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
stdout, stderr = await proc.communicate()
if proc.returncode == 0:
for line in stdout.decode().strip().split('\n'):
if line.startswith('ipv4.method:'):
method = line.split(':')[1]
break
is_dhcp = method == "auto"
interfaces[device] = {
"ip_address": ip_address or "N/A",
"is_dhcp": is_dhcp,
"method": method,
"connection_name": connection_name
}
port_mapping = {
"port1": "eth0",
"port2": "eth1"
}
return {
"interfaces": interfaces,
"port_mapping": port_mapping
}
except HTTPException:
raise
except Exception as e:
log.error("Exception in /network_info: %s", e, exc_info=True)
raise HTTPException(status_code=500, detail=str(e))
@app.post("/set_network_config")
async def set_network_config(config: dict):
"""Set network configuration (DHCP or Static IP) for a specific interface.
Expected payload:
{
"interface": "eth0",
"is_dhcp": true/false,
"ip_address": "192.168.1.100" (required if is_dhcp is false),
"netmask": "24" (optional, defaults to 24)
}
"""
try:
interface = config.get("interface")
is_dhcp = config.get("is_dhcp", True)
ip_address = config.get("ip_address")
netmask = config.get("netmask", "24")
if not interface:
raise HTTPException(status_code=400, detail="Interface name is required")
proc = await asyncio.create_subprocess_exec(
"nmcli", "-t", "-f", "NAME,DEVICE", "connection", "show",
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
stdout, stderr = await proc.communicate()
if proc.returncode != 0:
raise HTTPException(status_code=500, detail="Failed to get network connections")
connection_name = None
for line in stdout.decode().strip().split('\n'):
if not line:
continue
parts = line.split(':')
if len(parts) >= 2 and parts[1] == interface:
connection_name = parts[0]
break
if not connection_name:
log.info(f"No connection found for {interface}, creating new connection")
proc = await asyncio.create_subprocess_exec(
"sudo", "nmcli", "con", "add", "type", "ethernet",
"ifname", interface, "con-name", f"Wired connection {interface}",
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
stdout, stderr = await proc.communicate()
if proc.returncode != 0:
raise HTTPException(status_code=500, detail=f"Failed to create connection for {interface}: {stderr.decode()}")
connection_name = f"Wired connection {interface}"
if is_dhcp:
proc = await asyncio.create_subprocess_exec(
"sudo", "nmcli", "con", "modify", connection_name, "ipv4.method", "auto",
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
await proc.communicate()
if proc.returncode != 0:
raise HTTPException(status_code=500, detail="Failed to set DHCP mode")
proc = await asyncio.create_subprocess_exec(
"sudo", "nmcli", "con", "modify", connection_name, "ipv4.addresses", "",
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
await proc.communicate()
else:
if not ip_address:
raise HTTPException(status_code=400, detail="IP address is required for static configuration")
import re
ip_pattern = re.compile(r'^(\d{1,3}\.){3}\d{1,3}$')
if not ip_pattern.match(ip_address):
raise HTTPException(status_code=400, detail="Invalid IP address format")
octets = ip_address.split('.')
if not all(0 <= int(octet) <= 255 for octet in octets):
raise HTTPException(status_code=400, detail="IP address octets must be between 0 and 255")
proc = await asyncio.create_subprocess_exec(
"sudo", "nmcli", "con", "modify", connection_name,
"ipv4.method", "manual",
"ipv4.addresses", f"{ip_address}/{netmask}",
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
await proc.communicate()
if proc.returncode != 0:
raise HTTPException(status_code=500, detail="Failed to set static IP")
proc = await asyncio.create_subprocess_exec(
"sudo", "nmcli", "con", "up", connection_name,
stdout=asyncio.subprocess.PIPE,
stderr=asyncio.subprocess.PIPE
)
stdout, stderr = await proc.communicate()
if proc.returncode != 0:
log.info("Connection activation returned non-zero (may be expected if no cable): %s", stderr.decode())
return {"status": "success", "message": f"Network configuration updated for {interface}"}
except HTTPException:
raise
except Exception as e:
log.error("Exception in /set_network_config: %s", e, exc_info=True)
raise HTTPException(status_code=500, detail=str(e))
if __name__ == '__main__':
import os
os.chdir(os.path.dirname(__file__))

View File

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

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"

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View File

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

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
}

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

40
src/openocd/flash.sh Normal file
View File

@@ -0,0 +1,40 @@
#!/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"
echo "Flashing complete."

13111
src/openocd/merged.hex Normal file

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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

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

29
src/service/update_and_run_server_and_frontend.sh Normal file → Executable file
View File

@@ -4,6 +4,35 @@ set -e
# This script installs, enables, and restarts the auracast-server and auracast-frontend services
# Requires sudo privileges
# Ensure static link local is activated (for direct laptop connection)
# Enable link-local for all wired ethernet connections
while IFS=: read -r name type; do
if [[ "$type" == *"ethernet"* ]]; then
echo "Enabling IPv4 link-local for connection: $name"
sudo nmcli connection modify "$name" ipv4.link-local enabled 2>/dev/null || echo "Failed to modify $name"
sudo nmcli connection up "$name" 2>/dev/null || echo "Failed to bring up $name"
fi
done < <(nmcli -t -f NAME,TYPE connection show)
# Configure Avahi to prefer DHCP address over static fallback for mDNS
# Get the DHCP-assigned IP (first non-localhost, non-192.168.42.10 IP)
DHCP_IP=$(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
# Copy system service file for frontend
sudo cp /home/caster/bumble-auracast/src/service/auracast-frontend.service /etc/systemd/system/auracast-frontend.service