auracaster/bumble-auracast
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make everything controllable by just one button

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This commit is contained in:
pstruebi
2025-06-15 13:44:09 +02:00
parent 501b880931
commit d54d18987a
3 changed files with 110 additions and 54 deletions
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1
.gitignore vendored
View File

@@ -39,3 +39,4 @@ __pycache__/
wg_config/wg_confs/
records/
src/auracast/server/stream_settings.json

111
src/auracast/server/multicast_frontend.py
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@@ -8,26 +8,44 @@ from auracast import auracast_config
PTIME = 40
BACKEND_URL = "http://localhost:5000"
# Try loading persisted settings from backend
saved_settings = {}
try:
resp = requests.get(f"{BACKEND_URL}/status", timeout=1)
if resp.status_code == 200:
saved_settings = resp.json()
except Exception:
saved_settings = {}
st.title("🎙️ Auracast Audio Mode Control")
# Audio mode selection
# Audio mode selection with persisted default
options = ["Webapp", "USB"]
saved_audio_mode = saved_settings.get("audio_mode", "Webapp")
if saved_audio_mode not in options:
saved_audio_mode = "Webapp"
audio_mode = st.selectbox(
"Audio Mode",
["Webapp", "Network", "Cloud Announcements"]
options,
index=options.index(saved_audio_mode)
)
if audio_mode in ["Webapp", "Network"]:
# Stream quality selection
quality = st.selectbox("Stream Quality", ["High (48kHz)", "Mid (24kHz)", "Fair (16kHz)"])
if audio_mode in ["Webapp", "USB"]:
# Stream quality selection (temporarily disabled)
# quality = st.selectbox("Stream Quality", ["High (48kHz)", "Mid (24kHz)", "Fair (16kHz)"])
quality_map = {
"High (48kHz)": {"rate": 48000, "octets": 120},
"Mid (24kHz)": {"rate": 24000, "octets": 60},
"Fair (16kHz)": {"rate": 16000, "octets": 40},
"High (48kHz)": {"rate": 48000, "octets": 120},
"Mid (24kHz)": {"rate": 24000, "octets": 60},
"Fair (16kHz)": {"rate": 16000, "octets": 40},
}
stream_name = st.text_input("Channel Name", value="Broadcast0")
language = st.text_input("Language (ISO 639-3)", value="deu")
start_stream = st.button("Setup Auracast")
# Default to high quality while UI is hidden
quality = "High (48kHz)"
default_name = saved_settings.get('channel_names', ["Broadcast0"])[0]
default_lang = saved_settings.get('languages', ["deu"])[0]
stream_name = st.text_input("Channel Name", value=default_name)
language = st.text_input("Language (ISO 639-3)", value=default_lang)
start_stream = st.button("Start Auracast")
if start_stream:
# Prepare config using the model (do NOT send qos_config, only relevant fields)
@@ -40,7 +58,11 @@ if audio_mode in ["Webapp", "Network"]:
name=stream_name,
program_info=f"{stream_name} {quality}",
language=language,
audio_source="webrtc" if audio_mode=="Webapp" else "network",
audio_source=(
"webrtc" if audio_mode == "Webapp" else (
"usb" if audio_mode == "USB" else "network"
)
),
input_format="auto",
iso_que_len=1, # TODO: this should be way less to decrease delay
sampling_frequency=q['rate'],
@@ -57,45 +79,42 @@ if audio_mode in ["Webapp", "Network"]:
except Exception as e:
st.error(f"Error: {e}")
if audio_mode == "Webapp":
st.markdown("Click start and speak; watch your backend logs to see incoming RTP.")
if audio_mode == "Webapp" and start_stream:
st.markdown("Starting microphone; allow access if prompted and speak.")
component = f"""
<button id='go'>Start microphone</button>
<script>
const go = document.getElementById('go');
go.onclick = async () => {{
go.disabled = true;
const pc = new RTCPeerConnection(); // No STUN needed for localhost
const stream = await navigator.mediaDevices.getUserMedia({{audio:true}});
stream.getTracks().forEach(t => pc.addTrack(t, stream));
// --- WebRTC offer/answer exchange ---
const offer = await pc.createOffer()
// Patch SDP offer to include a=ptime using global PTIME
let sdp = offer.sdp;
const ptime_line = 'a=ptime:{PTIME}';
const maxptime_line = 'a=maxptime:{PTIME}';
if (sdp.includes('a=sendrecv')) {{
sdp = sdp.replace('a=sendrecv', 'a=sendrecv\\n' + ptime_line + '\\n' + maxptime_line);
}} else {{
sdp += '\\n' + ptime_line + '\\n' + maxptime_line;
}}
const patched_offer = new RTCSessionDescription({{sdp, type: offer.type}})
await pc.setLocalDescription(patched_offer)
// Send offer to backend
const response = await fetch(
"{BACKEND_URL}/offer",
{{
method: 'POST',
headers: {{'Content-Type':'application/json'}},
body: JSON.stringify({{sdp: pc.localDescription.sdp, type: pc.localDescription.type}})
(async () => {{
const pc = new RTCPeerConnection(); // No STUN needed for localhost
const stream = await navigator.mediaDevices.getUserMedia({{audio:true}});
stream.getTracks().forEach(t => pc.addTrack(t, stream));
// --- WebRTC offer/answer exchange ---
const offer = await pc.createOffer();
// Patch SDP offer to include a=ptime using global PTIME
let sdp = offer.sdp;
const ptime_line = 'a=ptime:{PTIME}';
const maxptime_line = 'a=maxptime:{PTIME}';
if (sdp.includes('a=sendrecv')) {{
sdp = sdp.replace('a=sendrecv', 'a=sendrecv\\n' + ptime_line + '\\n' + maxptime_line);
}} else {{
sdp += '\\n' + ptime_line + '\\n' + maxptime_line;
}}
)
const answer = await response.json()
await pc.setRemoteDescription(new RTCSessionDescription({{sdp: answer.sdp, type: answer.type}}))
}};
const patched_offer = new RTCSessionDescription({{sdp, type: offer.type}});
await pc.setLocalDescription(patched_offer);
// Send offer to backend
const response = await fetch(
"{BACKEND_URL}/offer",
{{
method: 'POST',
headers: {{'Content-Type':'application/json'}},
body: JSON.stringify({{sdp: pc.localDescription.sdp, type: pc.localDescription.type}})
}}
);
const answer = await response.json();
await pc.setRemoteDescription(new RTCSessionDescription({{sdp: answer.sdp, type: answer.type}}));
}})();
</script>
"""
st.components.v1.html(component, height=80)
st.components.v1.html(component, height=0)
else:
st.header("Advertised Streams (Cloud Announcements)")
st.info("This feature requires backend support to list advertised streams.")

52
src/auracast/server/multicast_server.py
View File

@@ -2,6 +2,8 @@ import glob
import os
import logging as log
import uuid
import json
from datetime import datetime
import numpy as np
from pydantic import BaseModel
@@ -15,6 +17,27 @@ from typing import List, Set
import traceback
from auracast.utils.webrtc_audio_input import WebRTCAudioInput
# Path to persist stream settings
STREAM_SETTINGS_FILE = os.path.join(os.path.dirname(__file__), 'stream_settings.json')
def load_stream_settings() -> dict:
"""Load persisted stream settings if available."""
if os.path.exists(STREAM_SETTINGS_FILE):
try:
with open(STREAM_SETTINGS_FILE, 'r', encoding='utf-8') as f:
return json.load(f)
except Exception:
return {}
return {}
def save_stream_settings(settings: dict):
"""Save stream settings to disk."""
try:
with open(STREAM_SETTINGS_FILE, 'w', encoding='utf-8') as f:
json.dump(settings, f, indent=2)
except Exception as e:
log.error('Unable to persist stream settings: %s', e)
app = FastAPI()
# Allow CORS for frontend on localhost
@@ -52,6 +75,20 @@ async def initialize(conf: auracast_config.AuracastConfigGroup):
HTTPException(status_code=500, detail='No suitable transport found.')
# initialize the streams dict
# persist stream settings for later retrieval
# Derive audio_mode from first BIG audio_source
first_source = conf.bigs[0].audio_source if conf.bigs else ''
audio_mode_persist = (
'Webapp' if first_source == 'webrtc' else
'USB' if first_source == 'usb' else
'Network'
)
save_stream_settings({
'channel_names': [big.name for big in conf.bigs],
'languages': [big.language for big in conf.bigs],
'audio_mode': audio_mode_persist,
'timestamp': datetime.utcnow().isoformat()
})
global_config_group = conf
log.info(
'Initializing multicaster with config:\n %s', conf.model_dump_json(indent=2)
@@ -96,14 +133,13 @@ async def stop_audio():
@app.get("/status")
async def get_status():
"""Gets the current status of the multicaster."""
if multicaster:
return multicaster.get_status()
else:
return {
'is_initialized': False,
'is_streaming': False,
}
"""Gets the current status of the multicaster together with persisted stream info."""
status = multicaster.get_status() if multicaster else {
'is_initialized': False,
'is_streaming': False,
}
status.update(load_stream_settings())
return status
PTIME = 160 # TODO: seems to have no effect at all