improve selection of audio source

README.md

@@ -138,5 +138,27 @@ sudo systemctl status auracast-frontend
 
 If you want to run the services as a specific user, edit the `User=` line in the service files accordingly.
 
+# install port audio so it can see pipewire devices on raspian
+
+sudo apt update
+sudo apt install --no-install-recommends \
+        git build-essential cmake pkg-config \
+        libasound2-dev libpulse-dev libjack-jackd2-dev
+git clone https://github.com/PortAudio/portaudio.git
+cd portaudio            # branch = master (19.7‑devel)
+rm -rf build
+cmake -S . -B build -G"Unix Makefiles" \
+      -DBUILD_SHARED_LIBS=ON \
+      -DPA_USE_ALSA=ON \
+      -DPA_USE_OSS=ON  \
+      -DPA_USE_PULSEAUDIO=ON \
+      -DPA_USE_JACK=OFF
+cmake --build build -j$(nproc)
+sudo apt remove -y libportaudio2 portaudio19-dev libportaudiocpp0
+sudo cmake --install build          # installs to /usr/local/lib
+sudo ldconfig                        # refresh linker cache
+
+
 # Known issues:
-- When running on a laptop there might be issues switching between usb and browser audio input since they use the same audio device
+- When running on a laptop there might be issues switching between usb and browser audio input since they use the same audio device
+

poetry.lock

@@ -1476,27 +1476,43 @@ files = [
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 ]
 
@@ -2448,17 +2464,17 @@ files = [
 
 [[package]]
 name = "sounddevice"
-version = "0.5.1"
+version = "0.5.2"
 description = "Play and Record Sound with Python"
 optional = false
 python-versions = ">=3.7"
 groups = ["main"]
 files = [
-    {file = "sounddevice-0.5.1-py3-none-any.whl", hash = "sha256:e2017f182888c3f3c280d9fbac92e5dbddac024a7e3442f6e6116bd79dab8a9c"},
+    {file = "sounddevice-0.5.2-py3-none-any.whl", hash = "sha256:82375859fac2e73295a4ab3fc60bd4782743157adc339561c1f1142af472f505"},
-    {file = "sounddevice-0.5.1-py3-none-macosx_10_6_x86_64.macosx_10_6_universal2.whl", hash = "sha256:d16cb23d92322526a86a9490c427bf8d49e273d9ccc0bd096feecd229cde6031"},
+    {file = "sounddevice-0.5.2-py3-none-macosx_10_6_x86_64.macosx_10_6_universal2.whl", hash = "sha256:943f27e66037d41435bdd0293454072cdf657b594c9cde63cd01ee3daaac7ab3"},
-    {file = "sounddevice-0.5.1-py3-none-win32.whl", hash = "sha256:d84cc6231526e7a08e89beff229c37f762baefe5e0cc2747cbe8e3a565470055"},
+    {file = "sounddevice-0.5.2-py3-none-win32.whl", hash = "sha256:3a113ce614a2c557f14737cb20123ae6298c91fc9301eb014ada0cba6d248c5f"},
-    {file = "sounddevice-0.5.1-py3-none-win_amd64.whl", hash = "sha256:4313b63f2076552b23ac3e0abd3bcfc0c1c6a696fc356759a13bd113c9df90f1"},
+    {file = "sounddevice-0.5.2-py3-none-win_amd64.whl", hash = "sha256:e18944b767d2dac3771a7771bdd7ff7d3acd7d334e72c4bedab17d1aed5dbc22"},
-    {file = "sounddevice-0.5.1.tar.gz", hash = "sha256:09ca991daeda8ce4be9ac91e15a9a81c8f81efa6b695a348c9171ea0c16cb041"},
+    {file = "sounddevice-0.5.2.tar.gz", hash = "sha256:c634d51bd4e922d6f0fa5e1a975cc897c947f61d31da9f79ba7ea34dff448b49"},
 ]
 
 [package.dependencies]
@@ -2903,4 +2919,4 @@ test = ["pytest", "pytest-asyncio"]
 [metadata]
 lock-version = "2.1"
 python-versions = ">=3.11"
-content-hash = "ad107b21cc3f0f5e78c4bb2dccd12cf320d37b2d3f262618796d848b05d06086"
+content-hash = "ee5d8e347947d5b3651aa1b9bcfb8b952e706d0a4a7d28bc0a9d3e5526d82c16"

pyproject.toml

@@ -6,16 +6,15 @@ requires-python = ">=3.11"
 dependencies = [
     "bumble @ git+ssh://git@ssh.pstruebi.xyz:222/auracaster/bumble_mirror.git@12bcdb7770c0d57a094bc0a96cd52e701f97fece",
     "lc3 @ git+ssh://git@ssh.pstruebi.xyz:222/auracaster/liblc3.git@7558637303106c7ea971e7bb8cedf379d3e08bcc",
-    "sounddevice",
     "aioconsole",
     "fastapi==0.115.11",
     "uvicorn==0.34.0",
     "aiohttp==3.9.3",
-    "sounddevice (>=0.5.1,<0.6.0)",
     "aioconsole (>=0.8.1,<0.9.0)",
     "numpy (>=2.2.6,<3.0.0)",
     "streamlit (>=1.45.1,<2.0.0)",
-    "aiortc (>=1.13.0,<2.0.0)"
+    "aiortc (>=1.13.0,<2.0.0)",
+    "sounddevice (>=0.5.2,<0.6.0)"
 ]
 
 [project.optional-dependencies]

src/auracast/server/multicast_frontend.py

@@ -203,8 +203,21 @@ else:
     # Input device selection for USB mode
     if audio_mode == "USB/Network":
         resp = requests.get(f"{BACKEND_URL}/audio_inputs")
-        input_options = [f"{d['id']}:{d['name']}" for d in resp.json().get('inputs', [])]
+        device_list = resp.json().get('inputs', [])
+        # Display "name [id]" but use name as value
+        input_options = [f"{d['name']} [{d['id']}]" for d in device_list]
+        option_name_map = {f"{d['name']} [{d['id']}]": d['name'] for d in device_list}
+        device_names = [d['name'] for d in device_list]
 
+        # Determine default input by name
+        default_input_name = saved_settings.get('input_device')
+        if default_input_name not in device_names and device_names:
+            default_input_name = device_names[0]
+        default_input_label = None
+        for label, name in option_name_map.items():
+            if name == default_input_name:
+                default_input_label = label
+                break
         if not input_options:
             st.warning("No hardware audio input devices found. Plug in a USB input device and click Refresh.")
             if st.button("Refresh"):
@@ -215,11 +228,13 @@ else:
                 st.rerun()
             input_device = None
         else:
-            if default_input not in input_options:
-                default_input = input_options[0]
             col1, col2 = st.columns([3, 1], vertical_alignment="bottom")
             with col1:
-                selected_option = st.selectbox("Input Device", input_options, index=input_options.index(default_input))
+                selected_option = st.selectbox(
+                    "Input Device",
+                    input_options,
+                    index=input_options.index(default_input_label) if default_input_label in input_options else 0
+                )
             with col2:
                 if st.button("Refresh"):
                     try:
@@ -227,8 +242,8 @@ else:
                     except Exception as e:
                         st.error(f"Failed to refresh devices: {e}")
                     st.rerun()
-            # We send only the numeric/card identifier (before :) or 'default'
+            # Send only the device name to backend
-            input_device = selected_option.split(":", 1)[0] if ":" in selected_option else selected_option
+            input_device = option_name_map[selected_option] if selected_option in option_name_map else None
     else:
         input_device = None
     

src/auracast/server/multicast_server.py

@@ -27,6 +27,13 @@ class Offer(BaseModel):
     sdp: str
     type: str
 
+def get_device_index_by_name(name: str):
+    """Return the device index for a given device name, or None if not found."""
+    for d in AUDIO_INPUT_DEVICES_CACHE:
+        if d["name"] == name:
+            return d["id"]
+    return None
+
 
 # Path to persist stream settings
 STREAM_SETTINGS_FILE = os.path.join(os.path.dirname(__file__), 'stream_settings.json')
@@ -50,6 +57,9 @@ def save_stream_settings(settings: dict):
         log.error('Unable to persist stream settings: %s', e)
 
 
+# TODO: select this dynamically from pw-cli ls Node determine id
+os.environ["PIPEWIRE_NODE"] = "88" 
+
 app = FastAPI()
 
 # Allow CORS for frontend on localhost
@@ -83,26 +93,36 @@ async def initialize(conf: auracast_config.AuracastConfigGroup):
                     conf.transport = f'serial:{device},115200,rtscts'
                     break
         if conf.transport == 'auto':
-            HTTPException(status_code=500, detail='No suitable transport found.')
+            raise HTTPException(status_code=500, detail='No suitable transport found.')
         # Derive audio_mode and input_device from first BIG audio_source
         first_source = conf.bigs[0].audio_source if conf.bigs else ''
         if first_source.startswith('device:'):
             audio_mode_persist = 'USB'
-            input_device = first_source.split(':', 1)[1] if ':' in first_source else 'default'
+            input_device_name = first_source.split(':', 1)[1] if ':' in first_source else None
+            # Map device name to current index for use with sounddevice
+            device_index = get_device_index_by_name(input_device_name) if input_device_name else None
+            # Patch config to use index for sounddevice (but persist name)
+            if device_index is not None:
+                for big in conf.bigs:
+                    if big.audio_source.startswith('device:'):
+                        big.audio_source = f'device:{device_index}'
+            else:
+                log.error(f"Device name '{input_device_name}' not found in current device list.")
+                raise HTTPException(status_code=400, detail=f"Audio device '{input_device_name}' not found.")
         elif first_source == 'webrtc':
             audio_mode_persist = 'Webapp'
-            input_device = None
+            input_device_name = None
         elif first_source.startswith('file:'):
             audio_mode_persist = 'Demo'
-            input_device = None
+            input_device_name = None
         else:
             audio_mode_persist = 'Network'
-            input_device = None
+            input_device_name = None
         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,
-            'input_device': input_device,
+            'input_device': input_device_name,
             'program_info': [getattr(big, 'program_info', None) for big in conf.bigs],
             'gain': [getattr(big, 'input_gain', 1.0) for big in conf.bigs],
             'timestamp': datetime.utcnow().isoformat()
@@ -137,12 +157,17 @@ async def initialize2(conf: auracast_config.AuracastConfigGroup):
                     conf.transport = f'serial:{device},115200,rtscts'
                     break
         if conf.transport == 'auto':
-            HTTPException(status_code=500, detail='No suitable transport found.')
+            raise HTTPException(status_code=500, detail='No suitable transport found.')
-        if multicaster2 is not None:
+        # Patch device name to index for sounddevice
-            try:
+        for big in conf.bigs:
-                await multicaster2.shutdown()
+            if big.audio_source.startswith('device:'):
-            except Exception:
+                device_name = big.audio_source.split(':', 1)[1]
-                log.warning("Failed to shutdown previous multicaster2", exc_info=True)
+                device_index = get_device_index_by_name(device_name)
+                if device_index is not None:
+                    big.audio_source = f'device:{device_index}'
+                else:
+                    log.error(f"Device name '{device_name}' not found in current device list.")
+                    raise HTTPException(status_code=400, detail=f"Audio device '{device_name}' not found.")
         log.info('Initializing multicaster2 with config:\n %s', conf.model_dump_json(indent=2))
         multicaster2 = multicast_control.Multicaster(conf, conf.bigs)
         await multicaster2.init_broadcast()
@@ -220,13 +245,11 @@ async def scan_audio_devices():
             sd._terminate()
             sd._initialize()
 
-        # TODO: select this dynamically from pw-cli ls Node determine id
-        #os.environ["PIPEWIRE_NODE"] = "76" 
         devs = sd.query_devices()
         inputs = [
             {"id": idx, "name": d["name"]}
             for idx, d in enumerate(devs)
-        #    if d.get("max_input_channels", 0) > 0 and ("(hw:" in d["name"].lower() or "usb" in d["name"].lower())
+            if d.get("max_input_channels", 0) > 0
         ]
         log.info('Found %d audio input devices: %s', len(inputs), inputs)
         AUDIO_INPUT_DEVICES_CACHE = inputs
@@ -243,8 +266,9 @@ async def startup_event():
 
 @app.get("/audio_inputs")
 async def list_audio_inputs():
-    """Return available hardware audio input devices from cache."""
+    """Return available hardware audio input devices from cache (by name, for selection)."""
-    return {"inputs": AUDIO_INPUT_DEVICES_CACHE}
+    # Only expose name and id for frontend
+    return {"inputs": [{"name": d["name"], "id": d["id"]} for d in AUDIO_INPUT_DEVICES_CACHE]}
 
 
 @app.post("/refresh_audio_inputs")

src/scripts/list_pw_nodes.py

@@ -0,0 +1,4 @@
+import sounddevice as sd, pprint
+print(sd._libname)                       # → /usr/local/lib/libportaudio.so.2
+print(sd.get_portaudio_version())        # 19.7.0‑devel (or newer)
+pprint.pprint(sd.query_devices())        # every PipeWire sink/source appears

src/service/aes67/pipewire-aes67.conf

@@ -87,7 +87,7 @@ context.modules = [
                             media.class = "Audio/Source"
                             device.api = aes67
                             # You can adjust the latency buffering here. Use integer values only
-                            sess.latency.msec = 50
+                            sess.latency.msec = 5
                             node.group = pipewire.ptp0
                         }
                     }
@@ -105,49 +105,4 @@ context.modules = [
             ]
         }
     },
-    { name = libpipewire-module-rtp-sink
-        args = {
-            ### Please select the interface here
-            local.ifname = eth0
-            ### If you want to create multiple output streams, please copy the whole
-            ### module-rtp-sink block, but change this multicast IP to another unused
-            ### one keeping 239.69.x.x range unless you know you need another one
-            destination.ip = 239.69.150.243
-            destination.port = 5004
-            net.mtu = 1280
-            net.ttl = 32
-            net.loop = false
-            # These should typically be equal
-            # You can customize packet length, but 1 ms should work for every device
-            # Consult receiver documentation to ensure it supports the value you set
-            sess.min-ptime = 1
-            sess.max-ptime = 1
-            ### Please change this, especially if you create multiple sinks
-            sess.name = "PipeWire RTP stream"
-            sess.media = "audio"
-            # This property is used if you aren't using ptp4l 4
-            sess.ts-refclk = "ptp=traceable"
-            sess.ts-offset = 0
-            # You can adjust the latency buffering here. Use integer values only
-            sess.latency.msec = 3
-            audio.format = "S24BE"
-            audio.rate = 48000
-            audio.channels = 2
-            # These channel names will be visible both to applications and AES67 receivers
-            node.channel-names = ["CH1", "CH2"]
-
-            stream.props = {
-                ### Please change the sink name, this is necessary when you create multiple sinks
-                node.name = "rtp-sink"
-                media.class = "Audio/Sink"
-                node.virtual = false
-                device.api = aes67
-                sess.sap.announce = true
-                node.always-process = true
-                node.group = pipewire.ptp0
-                rtp.ntp = 0
-                rtp.fetch-ts-refclk = true
-            }
-        }
-    },
 ]