STT experiments, implements both vosx and whisper; whisper is too slow (cpu heavy) vosx has low acc; bigger vosx needs too much storage.

Fixes the bug that local link loses connection after a few minutes.

Openproject:
#1025
#608

Reviewed-on: #34

pyproject.toml

@@ -19,7 +19,9 @@ dependencies = [
     "smbus2 (>=0.5.0,<0.6.0)",
     "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"
+    "pyalsaaudio @ git+ssh://git@gitea.summitwave.work:222/auracaster/sw_pyalsaaudio.git@b3d11582e03df6929b2e7acbaa1306afc7b8a6bc",
+    "vosk (>=0.3.45)",
+    "faster-whisper (>=1.0.0)"
 ]
 
 [project.optional-dependencies]

src/auracast/auracast_config.py

@@ -1,5 +1,11 @@
 from typing import List
-from pydantic import BaseModel
+from pydantic import BaseModel, field_validator
+
+# Discrete TX power levels (dBm) supported by the Nordic SoftDevice Controller
+# for the nRF radio PA. The HCI controller will clamp requested values to the
+# nearest supported step. The maximum is bounded by CONFIG_BT_CTLR_TX_PWR_*
+# in the hci_uart firmware (currently +8 dBm).
+TX_POWER_VALID = [8, 7, 6, 5, 4, 3, 2, 0, -4, -8, -12, -16, -20]
 
 # Define some base to hold the relevant parameters
 class AuracastQoSConfig(BaseModel):
@@ -28,13 +34,24 @@ class AuracastGlobalConfig(BaseModel):
     octets_per_frame: int = 40 #48kbps@24kHz # bitrate = octets_per_frame * 8 / frame len
     frame_duration_us: int = 10000
     presentation_delay_us: int = 40000
-    # TODO:pydantic does not support bytes serialization - use .hex and np.fromhex()
-    manufacturer_data: tuple[int, bytes] | tuple[None, None] = (None, None)
     # LE Audio: Broadcast Audio Immediate Rendering (metadata type 0x09)
     # When true, include a zero-length LTV with type 0x09 in the subgroup metadata
     # so receivers may render earlier than the presentation delay for lower latency.
     immediate_rendering: bool = False
     assisted_listening_stream: bool = False
+    # Bluetooth advertising TX power for this radio in dBm (per advertising set).
+    # Sent through HCI_LE_Set_Extended_Advertising_Parameters; the SDC clamps to
+    # nearest supported hardware step and propagates to primary/secondary adv,
+    # the periodic advertising train and the BIS ISO PDUs.
+    advertising_tx_power: int = 8
+
+    @field_validator('advertising_tx_power')
+    @classmethod
+    def _snap_tx_power(cls, v: int) -> int:
+        # Snap to the nearest supported discrete step in TX_POWER_VALID.
+        if v in TX_POWER_VALID:
+            return v
+        return min(TX_POWER_VALID, key=lambda s: abs(s - v))
 
 # "Audio input. "
 # "'device' -> use the host's default sound input device, "
@@ -62,7 +79,7 @@ class AuracastBigConfigDeu(AuracastBigConfig):
     name: str = 'Hörsaal A'
     language: str ='deu'
     program_info: str = 'Vorlesung DE'
-    audio_source: str = 'file:./testdata/wave_particle_5min_de.wav'
+    audio_source: str = 'file:./testdata/wave_particle_5min_de.lc3'
 
 class AuracastBigConfigEng(AuracastBigConfig):
     id: int = 123
@@ -70,7 +87,7 @@ class AuracastBigConfigEng(AuracastBigConfig):
     name: str  = 'Lecture Hall A'
     language: str ='eng'
     program_info: str = 'Lecture EN'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_en.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_en.lc3'
 
 class AuracastBigConfigFra(AuracastBigConfig):
     id: int = 1234
@@ -79,7 +96,7 @@ class AuracastBigConfigFra(AuracastBigConfig):
     name: str  = 'Auditoire A'
     language: str ='fra'
     program_info: str = 'Auditoire FR'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_fr.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_fr.lc3'
 
 class AuracastBigConfigSpa(AuracastBigConfig):
     id: int =12345
@@ -87,7 +104,7 @@ class AuracastBigConfigSpa(AuracastBigConfig):
     name: str  = 'Auditorio A'
     language: str ='spa'
     program_info: str = 'Auditorio ES'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_es.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_es.lc3'
 
 class AuracastBigConfigIta(AuracastBigConfig):
     id: int =1234567
@@ -95,7 +112,7 @@ class AuracastBigConfigIta(AuracastBigConfig):
     name: str  = 'Aula A'
     language: str ='ita'
     program_info: str = 'Aula IT'
-    audio_source: str  = 'file:./testdata/wave_particle_5min_it.wav'
+    audio_source: str  = 'file:./testdata/wave_particle_5min_it.lc3'
 
 
 class AuracastBigConfigPol(AuracastBigConfig):
@@ -104,7 +121,7 @@ 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):

src/auracast/dcp_parser.py

@@ -0,0 +1,72 @@
+"""DCP XML subtitle file parser (Interop and SMPTE 428-7 formats).
+
+Timecode format: HH:MM:SS:FF  (frame-based, default 24 fps)
+                 HH:MM:SS.mmm (millisecond decimal, also accepted)
+"""
+from __future__ import annotations
+
+import re
+import xml.etree.ElementTree as ET
+from dataclasses import dataclass
+from typing import List
+
+
+@dataclass
+class Subtitle:
+    time_in: float   # seconds (float)
+    time_out: float  # seconds (float)
+    text: str
+
+
+def _parse_timecode(tc: str, fps: int = 24) -> float:
+    """Parse a DCP timecode string to float seconds."""
+    # HH:MM:SS:FF
+    m = re.match(r'^(\d+):(\d+):(\d+):(\d+)$', tc.strip())
+    if m:
+        h, mi, s, f = int(m.group(1)), int(m.group(2)), int(m.group(3)), int(m.group(4))
+        return h * 3600 + mi * 60 + s + f / fps
+
+    # HH:MM:SS.mmm
+    m = re.match(r'^(\d+):(\d+):(\d+)\.(\d+)$', tc.strip())
+    if m:
+        h, mi, s = int(m.group(1)), int(m.group(2)), int(m.group(3))
+        frac = float('0.' + m.group(4))
+        return h * 3600 + mi * 60 + s + frac
+
+    raise ValueError(f"Unrecognized DCP timecode: {tc!r}")
+
+
+def parse_dcp_xml(path: str, fps: int = 24) -> List[Subtitle]:
+    """Parse a DCP XML subtitle file and return a time-sorted list of Subtitles."""
+    tree = ET.parse(path)
+    root = tree.getroot()
+
+    # Strip namespace so element lookups work regardless of schema version
+    ns_match = re.match(r'\{(.+?)\}', root.tag)
+    ns = ns_match.group(0) if ns_match else ''
+
+    subtitles: List[Subtitle] = []
+
+    for subtitle_el in root.iter(f'{ns}Subtitle'):
+        time_in_str = subtitle_el.get('TimeIn', '')
+        time_out_str = subtitle_el.get('TimeOut', '')
+        if not time_in_str or not time_out_str:
+            continue
+
+        parts: List[str] = []
+        for text_el in subtitle_el.iter(f'{ns}Text'):
+            t = (text_el.text or '').strip()
+            if t:
+                parts.append(t)
+
+        text = ' '.join(parts)
+        if not text:
+            continue
+
+        subtitles.append(Subtitle(
+            time_in=_parse_timecode(time_in_str, fps),
+            time_out=_parse_timecode(time_out_str, fps),
+            text=text,
+        ))
+
+    return sorted(subtitles, key=lambda s: s.time_in)

src/auracast/faster_whisper_textcast.py

@@ -0,0 +1,259 @@
+"""faster-whisper speech-to-text → TextCast streamer.
+
+Captures mono audio from an analog ALSA/sounddevice input, runs
+faster-whisper offline ASR in a background thread (chunked, every
+CHUNK_S seconds), and broadcasts recognised text over the TextCast BLE
+broadcast using the same SDU framing as text_multicast.py.
+
+Usage (CLI):
+    poetry run python -m auracast.faster_whisper_textcast \\
+        --model tiny.en \\
+        --device ch1 \\
+        --transport serial:/dev/ttyAMA3,1000000,rtscts
+"""
+from __future__ import annotations
+
+import asyncio
+import logging
+import os
+import queue
+import threading
+import time
+from typing import Optional
+
+import numpy as np
+import samplerate
+import sounddevice as sd
+
+from auracast import auracast_config, multicast
+from auracast.text_multicast import (
+    SDU_SIZE,
+    SDU_INTERVAL_US,
+    _make_text_frame,
+    _make_idle_frame,
+)
+
+log = logging.getLogger('faster_whisper_textcast')
+
+CAPTURE_SAMPLE_RATE = 48_000
+WHISPER_SAMPLE_RATE = 16_000
+BLOCK_FRAMES_48K = 4800       # 100 ms capture blocks
+CHUNK_S = 3.0                  # transcribe every N seconds of audio
+CAPTION_HOLD_S = 4.0           # keep caption visible after last transcription
+SILENCE_RMS = 0.003            # skip transcription if chunk is below this RMS
+BROADCAST_NAME = 'LiveCaption'
+
+VALID_MODELS = ['tiny.en', 'base.en', 'small.en', 'tiny', 'base', 'small']
+
+
+def _tail_to_fit(text: str, max_bytes: int) -> str:
+    """Return the tail of *text* that fits in *max_bytes* UTF-8 bytes."""
+    encoded = text.encode('utf-8')
+    if len(encoded) <= max_bytes:
+        return text
+    tail = encoded[-max_bytes:].decode('utf-8', errors='ignore')
+    sp = tail.find(' ')
+    return tail[sp + 1:] if sp != -1 else tail
+
+
+def _resolve_device(device: str) -> Optional[int]:
+    """Return sounddevice index for a name or numeric string, or None for default."""
+    if not device:
+        return None
+    if device.isdigit():
+        return int(device)
+    for i, d in enumerate(sd.query_devices()):
+        if d['name'] == device and d['max_input_channels'] > 0:
+            return i
+    log.warning("Device '%s' not found in sounddevice list – using default input", device)
+    return None
+
+
+async def _iso_write_loop(bigs: dict, shared: dict, lock: threading.Lock) -> None:
+    """ISO SDU write loop – runs at ~10 ms per iteration."""
+    iso_queue = bigs['big0']['iso_queue']
+    last_sent: str = ''
+
+    while True:
+        now = time.monotonic()
+        with lock:
+            text: str = shared.get('text', '')
+            expiry: float = shared.get('expiry', 0.0)
+
+        if text and now < expiry:
+            display_text = _tail_to_fit(text, SDU_SIZE - 2)
+            if display_text != last_sent:
+                log.info("Caption: %s", display_text)
+                last_sent = display_text
+            frame = _make_text_frame(display_text)
+        else:
+            if last_sent:
+                log.info("Caption cleared")
+                last_sent = ''
+                with lock:
+                    shared['text'] = ''
+            frame = _make_idle_frame()
+
+        await iso_queue.write(frame)
+
+
+def _whisper_thread(
+    model_size: str,
+    device: str,
+    shared: dict,
+    lock: threading.Lock,
+    stop_event: threading.Event,
+) -> None:
+    """Blocking audio capture + faster-whisper transcription loop."""
+    try:
+        from faster_whisper import WhisperModel  # type: ignore
+    except ImportError:
+        log.error("faster-whisper is not installed. Run: poetry add faster-whisper")
+        return
+
+    log.info("Loading faster-whisper model '%s' (int8, CPU) …", model_size)
+    model = WhisperModel(model_size, device="cpu", compute_type="int8")
+    log.info("Model '%s' loaded.", model_size)
+
+    audio_q: queue.Queue = queue.Queue()
+    resampler = samplerate.Resampler('sinc_fastest', channels=1)
+    ratio = WHISPER_SAMPLE_RATE / CAPTURE_SAMPLE_RATE
+    chunk_frames = int(CHUNK_S * WHISPER_SAMPLE_RATE)
+    audio_buffer = np.zeros(0, dtype=np.float32)
+
+    dev_idx = _resolve_device(device)
+
+    def _cb(indata: np.ndarray, frames: int, time_info, status) -> None:
+        if status:
+            log.warning("Audio status: %s", status)
+        if stop_event.is_set():
+            raise sd.CallbackStop()
+        mono = indata[:, 0].astype(np.float32)
+        downsampled = resampler.process(mono, ratio, end_of_input=False)
+        audio_q.put(downsampled.copy())
+
+    try:
+        with sd.InputStream(
+            samplerate=CAPTURE_SAMPLE_RATE,
+            blocksize=BLOCK_FRAMES_48K,
+            device=dev_idx,
+            dtype='float32',
+            channels=1,
+            callback=_cb,
+        ):
+            log.info("WhisperCast listening on device '%s' (idx=%s) …", device, dev_idx)
+            while not stop_event.is_set():
+                try:
+                    chunk = audio_q.get(timeout=0.2)
+                    audio_buffer = np.concatenate([audio_buffer, chunk])
+                except queue.Empty:
+                    continue
+
+                if len(audio_buffer) < chunk_frames:
+                    continue
+
+                pcm = audio_buffer[:chunk_frames].copy()
+                audio_buffer = audio_buffer[chunk_frames:]
+
+                rms = float(np.sqrt(np.mean(pcm ** 2)))
+                if rms < SILENCE_RMS:
+                    continue
+
+                t0 = time.monotonic()
+                segments, _ = model.transcribe(
+                    pcm,
+                    beam_size=1,
+                    language="en",
+                    vad_filter=True,
+                    vad_parameters={"min_silence_duration_ms": 300},
+                )
+                text = ' '.join(s.text.strip() for s in segments).strip()
+                elapsed = time.monotonic() - t0
+
+                if text:
+                    log.info("Transcribed (%.2fs): %s", elapsed, text)
+                    with lock:
+                        shared['text'] = text
+                        shared['expiry'] = time.monotonic() + CAPTION_HOLD_S
+                else:
+                    log.debug("Silent chunk skipped (rms=%.4f, took=%.2fs)", rms, elapsed)
+
+    except Exception as exc:
+        log.error("WhisperCast thread error: %s", exc, exc_info=True)
+
+
+async def broadcast_whisper(
+    transport: str,
+    model_size: str = 'tiny.en',
+    device: str = 'ch1',
+) -> None:
+    """Start a faster-whisper → TextCast broadcast. Runs until cancelled."""
+    if model_size not in VALID_MODELS:
+        raise ValueError(f"Unknown model '{model_size}'. Valid: {VALID_MODELS}")
+
+    config = auracast_config.AuracastConfigGroup(
+        bigs=[
+            auracast_config.AuracastBigConfig(
+                name=BROADCAST_NAME,
+                program_info='Live Captions',
+                language='eng',
+                audio_source='file:dummy',
+                iso_que_len=4,
+            ),
+        ],
+        auracast_sampling_rate_hz=16000,
+        octets_per_frame=SDU_SIZE,
+        frame_duration_us=SDU_INTERVAL_US,
+        presentation_delay_us=40_000,
+        qos_config=auracast_config.AuracastQosRobust(),
+        transport=transport,
+    )
+
+    shared: dict = {'text': '', 'expiry': 0.0}
+    lock = threading.Lock()
+    stop_event = threading.Event()
+
+    async with multicast.create_device(config) as ble_device:
+        bigs = await multicast.init_broadcast(ble_device, config, config.bigs)
+
+        t = threading.Thread(
+            target=_whisper_thread,
+            args=(model_size, device, shared, lock, stop_event),
+            daemon=True,
+        )
+        t.start()
+        log.info("WhisperCast started (device=%s, model=%s)", device, model_size)
+
+        try:
+            await _iso_write_loop(bigs, shared, lock)
+        except asyncio.CancelledError:
+            log.info("WhisperCast cancelled – shutting down")
+            stop_event.set()
+            t.join(timeout=5.0)
+            raise
+
+
+def main() -> None:
+    global CHUNK_S
+    import argparse
+    parser = argparse.ArgumentParser(description='faster-whisper → Auracast TextCast')
+    parser.add_argument(
+        '--model', default='tiny.en', choices=VALID_MODELS,
+        help='Whisper model size (default: tiny.en)',
+    )
+    parser.add_argument('--device', default='ch1',
+                        help='sounddevice input name or index (default: ch1)')
+    parser.add_argument(
+        '--transport',
+        default=os.environ.get('AURACAST_TRANSPORT', 'serial:/dev/ttyAMA3,1000000,rtscts'),
+        help='Bumble HCI transport string',
+    )
+    parser.add_argument('--chunk', type=float, default=CHUNK_S,
+                        help=f'Seconds per transcription chunk (default: {CHUNK_S})')
+    args = parser.parse_args()
+    CHUNK_S = args.chunk
+    multicast.run_async(broadcast_whisper(args.transport, args.model, args.device))
+
+
+if __name__ == '__main__':
+    main()

src/auracast/multicast.py

@@ -49,6 +49,7 @@ import bumble.transport
 import bumble.utils
 from bumble.device import Host, AdvertisingChannelMap
 from bumble.audio import io as audio_io
+from bumble.vendor.zephyr.hci import HCI_Write_Tx_Power_Level_Command
 
 from auracast import auracast_config
 from auracast.utils.read_lc3_file import read_lc3_file
@@ -462,21 +463,6 @@ async def init_broadcast(
             ],
         )
         logger.info('Setup Advertising')
-        advertising_manufacturer_data = (
-        b''
-        if global_config.manufacturer_data == (None, None)
-        else bytes(
-            core.AdvertisingData(
-                [
-                    (
-                        core.AdvertisingData.MANUFACTURER_SPECIFIC_DATA,
-                        struct.pack('<H', global_config.manufacturer_data[0])
-                        + global_config.manufacturer_data[1],
-                    )
-                ]
-            )
-        )
-    )
         bigs[f'big{i}']['broadcast_audio_announcement'] = bap.BroadcastAudioAnnouncement(conf.id)
 
         # Build advertising data types list
@@ -519,13 +505,22 @@ async def init_broadcast(
                 advertising_sid=i,
                 primary_advertising_phy=hci.Phy.LE_1M, # 2m phy config throws error - because for primary advertising channels, 1mbit is only supported
                 secondary_advertising_phy=hci.Phy.LE_1M, # this is the secondary advertising beeing send on non advertising channels (extendend advertising)
-                #advertising_tx_power= # tx power in dbm (max 20)
+                # Pass NO_PREFERENCE (0x7F) here for two reasons:
+                # 1. The Nordic SoftDevice Controller ignores this field for
+                #    advertising sets and always returns the compile-time
+                #    CONFIG_BT_CTLR_TX_PWR_* value. The real TX power is
+                #    applied via the Zephyr VS Write_Tx_Power_Level command
+                #    issued right after create_advertising_set() returns.
+                # 2. Bumble's HCI metadata declares this field as 1-byte
+                #    *unsigned* (a bumble bug — the BT spec defines it as
+                #    signed int8), so negative values would raise
+                #    "bytes must be in range(0, 256)" at serialization.
+                advertising_tx_power=hci.HCI_LE_Set_Extended_Advertising_Parameters_Command.TX_POWER_NO_PREFERENCE,
                 #secondary_advertising_max_skip=10,
             ),
             advertising_data=(
                 bigs[f'big{i}']['broadcast_audio_announcement'].get_advertising_data()
                 + bytes(core.AdvertisingData(advertising_data_types))
-                + advertising_manufacturer_data
             ),
             periodic_advertising_parameters=bumble.device.PeriodicAdvertisingParameters(
                 periodic_advertising_interval_min=80,
@@ -536,6 +531,48 @@ async def init_broadcast(
             auto_start=True,
         )
         bigs[f'big{i}']['advertising_set'] = advertising_set
+        # NOTE: selected_tx_power below reflects the SDC's compile-time max
+        # (LE_Set_Ext_Adv_Params was sent with NO_PREFERENCE). The actual
+        # transmit power is set by the VS Write_Tx_Power_Level call below.
+        logging.debug(
+            'LE_Set_Ext_Adv_Params reports controller fallback TX power: %+d dBm (handle=%d)',
+            getattr(advertising_set, 'selected_tx_power', 0),
+            i,
+        )
+
+        # The Nordic SoftDevice Controller does not honor the per-set
+        # advertising_tx_power passed in HCI_LE_Set_Extended_Advertising_Parameters
+        # (it returns the compile-time CONFIG_BT_CTLR_TX_PWR_* value regardless).
+        # Apply the requested level via the Zephyr Vendor-Specific HCI command
+        # Write_Tx_Power_Level (opcode 0xFC0E), which the SDC honors per
+        # advertising handle. The SDC clamps the value to the nearest supported
+        # hardware step (max bounded by CONFIG_BT_CTLR_TX_PWR_PLUS_8).
+        try:
+            adv_handle = getattr(advertising_set, 'advertising_handle', i)
+            response = await device.send_command(
+                HCI_Write_Tx_Power_Level_Command(
+                    handle_type=HCI_Write_Tx_Power_Level_Command.TX_POWER_HANDLE_TYPE_ADV,
+                    connection_handle=adv_handle,
+                    tx_power_level=global_config.advertising_tx_power,
+                )
+            )
+            rp = getattr(response, 'return_parameters', None)
+            status = getattr(rp, 'status', 0xFF) if rp is not None else 0xFF
+            selected = getattr(rp, 'selected_tx_power_level', None) if rp is not None else None
+            if status == 0 and selected is not None:
+                logging.info(
+                    'Advertising TX power (VS Write_Tx_Power_Level): requested=%+d dBm, controller selected=%+d dBm (handle=%d)',
+                    global_config.advertising_tx_power,
+                    selected,
+                    adv_handle,
+                )
+            else:
+                logging.warning(
+                    'VS Write_Tx_Power_Level failed: status=0x%02X handle=%d requested=%+d dBm',
+                    status, adv_handle, global_config.advertising_tx_power,
+                )
+        except Exception as e:
+            logging.warning('VS Write_Tx_Power_Level not supported by controller: %s', e)
 
         logging.info('Start Periodic Advertising')
         await advertising_set.start_periodic()
@@ -602,6 +639,29 @@ async def init_broadcast(
     return bigs
 
 
+def _lc3_file_byte_gen(filename: str, loop: bool = False):
+    """Stream LC3 frames from disk as individual bytes, with optional looping.
+
+    Yields one byte (int) at a time so it is compatible with the existing
+    ``bytes(itertools.islice(gen, bytes_per_frame))`` consumer without loading
+    the whole file into memory.
+    """
+    while True:
+        with open(filename, 'rb') as f:
+            f.read(18)  # skip 18-byte LC3 header
+            while True:
+                size_b = f.read(2)
+                if len(size_b) < 2:
+                    break
+                frame_size = struct.unpack('=H', size_b)[0]
+                frame = f.read(frame_size)
+                if len(frame) < frame_size:
+                    break
+                yield from frame
+        if not loop:
+            return
+
+
 class Streamer():
     """
     Streamer class that supports multiple input formats. See bumble for streaming from wav or device
@@ -757,13 +817,7 @@ class Streamer():
                 big['precoded'] = True
                 big['lc3_bytes_per_frame'] = global_config.octets_per_frame
                 filename = big_config[i].audio_source.replace('file:', '')
-
-                lc3_bytes = read_lc3_file(filename)
-                lc3_frames = iter(lc3_bytes)
-
-                if big_config[i].loop:
-                    lc3_frames = itertools.cycle(lc3_frames)
-                big['lc3_frames'] = lc3_frames
+                big['lc3_frames'] = _lc3_file_byte_gen(filename, loop=big_config[i].loop)
 
             # use wav files and code them entirely before streaming
             elif big_config[i].precode_wav and big_config[i].audio_source.endswith('.wav'):
@@ -884,6 +938,9 @@ class Streamer():
                     if lc3_frame == b'':  # Not all streams may stop at the same time
                         stream_finished[i] = True
                         continue
+
+                    for q_idx in range(big.get('num_bis', 1)):
+                        await big['iso_queues'][q_idx].write(lc3_frame)
                 else:  # code lc3 on the fly with perf counters
                     # Ensure frames generator exists (so we can aclose() on stop)
                     frames_gen = big.get('frames_gen')

src/auracast/server/multicast_frontend.py

@@ -100,6 +100,36 @@ QOS_PRESET_MAP = {
     "Robust": auracast_config.AuracastQosRobust(),
 }
 
+# Discrete advertising TX power steps in dBm supported by the Nordic SDC radio
+# PA. Sent through HCI_LE_Set_Extended_Advertising_Parameters; the controller
+# clamps to the nearest hardware step.
+TX_POWER_OPTIONS = [8, 7, 6, 5, 4, 3, 2, 0, -4, -8, -12, -16, -20]
+TX_POWER_DEFAULT = 8
+
+
+def _coerce_tx_power(value, default: int = TX_POWER_DEFAULT) -> int:
+    try:
+        v = int(value)
+    except (TypeError, ValueError):
+        return default
+    if v in TX_POWER_OPTIONS:
+        return v
+    return min(TX_POWER_OPTIONS, key=lambda s: abs(s - v))
+
+
+def _tx_power_selectbox(label: str, key: str, default: int, disabled: bool, help_text: str | None = None) -> int:
+    snapped = _coerce_tx_power(default)
+    idx = TX_POWER_OPTIONS.index(snapped)
+    return st.selectbox(
+        label,
+        TX_POWER_OPTIONS,
+        index=idx,
+        key=key,
+        format_func=lambda v: f"{v:+d} dBm",
+        disabled=disabled,
+        help=help_text or "Bluetooth advertising TX power for this radio. Higher values increase range; lower values reduce interference and power draw.",
+    )
+
 # Try loading persisted settings from backend
 saved_settings = {}
 try:
@@ -111,6 +141,9 @@ except Exception:
 
 # Define is_streaming early from the fetched status for use throughout the UI
 is_streaming = bool(saved_settings.get("is_streaming", False))
+textcast_is_streaming = bool(saved_settings.get("textcast_is_streaming", False))
+voskcast_is_streaming = bool(saved_settings.get("voskcast_is_streaming", False))
+whispercast_is_streaming = bool(saved_settings.get("whispercast_is_streaming", False))
 
 # Extract secondary status, if provided by the backend /status endpoint.
 secondary_status = saved_settings.get("secondary") or {}
@@ -155,6 +188,9 @@ options = [
     "Demo",
     "Analog",
     "Network - Dante",
+    "TextCast",
+    "VoskCast",
+    "WhisperCast",
     ]
 saved_audio_mode = saved_settings.get("audio_mode", "Demo")
 if saved_audio_mode not in options:
@@ -166,7 +202,7 @@ audio_mode = st.selectbox(
     "Audio Mode",
     options,
     index=options.index(saved_audio_mode) if saved_audio_mode in options else options.index("Demo"),
-    disabled=is_streaming,
+    disabled=is_streaming or textcast_is_streaming or voskcast_is_streaming or whispercast_is_streaming,
     help=(
         "Select the audio input source. Choose 'USB' for a connected USB audio device (via PipeWire), "
         "'Network' (AES67) for network RTP/AES67 sources, "
@@ -196,11 +232,94 @@ else:
     running_mode = backend_mode_mapped if (is_streaming and backend_mode_mapped) else audio_mode
 
 # Start/Stop buttons and status (moved to top)
-if audio_mode == "Demo":
+if audio_mode == "TextCast":
+    start_stream, stop_stream = render_stream_controls(textcast_is_streaming, "Start TextCast", "Stop TextCast", "TextCast", False)
+elif audio_mode == "VoskCast":
+    start_stream, stop_stream = render_stream_controls(voskcast_is_streaming, "Start VoskCast", "Stop VoskCast", "VoskCast", False)
+elif audio_mode == "Demo":
     start_stream, stop_stream = render_stream_controls(is_streaming, "Start Demo", "Stop Demo", running_mode, secondary_is_streaming)
 else:
     start_stream, stop_stream = render_stream_controls(is_streaming, "Start Auracast", "Stop Auracast", running_mode, secondary_is_streaming)
 
+# TextCast: DCP XML file uploader
+if audio_mode == "TextCast":
+    st.markdown("#### DCP Subtitle File")
+    dcp_file = st.file_uploader(
+        "Upload DCP XML subtitle file (.xml)",
+        type=["xml"],
+        disabled=textcast_is_streaming,
+        help="Upload a DCP-compliant subtitle XML file. Subtitles will be broadcast over Auracast.",
+    )
+    if dcp_file is not None:
+        content = dcp_file.read().decode("utf-8", errors="replace")
+        st.session_state['_textcast_dcp_content'] = content
+        st.session_state['_textcast_dcp_name'] = dcp_file.name
+        st.success(f"Loaded: {dcp_file.name} ({len(content):,} bytes)")
+    elif st.session_state.get('_textcast_dcp_name'):
+        st.info(f"Using previously uploaded file: {st.session_state['_textcast_dcp_name']}")
+    else:
+        st.warning("No subtitle file loaded. Upload a DCP XML file or use the sample below.")
+        if st.button("Load sample subtitle file", disabled=textcast_is_streaming):
+            import os as _os
+            _sample = _os.path.abspath(_os.path.join(
+                _os.path.dirname(__file__), '..', 'testdata', 'sample_subtitles.xml'))
+            try:
+                with open(_sample, 'r', encoding='utf-8') as _f:
+                    _content = _f.read()
+                st.session_state['_textcast_dcp_content'] = _content
+                st.session_state['_textcast_dcp_name'] = 'sample_subtitles.xml'
+                st.rerun()
+            except Exception as _e:
+                st.error(f"Could not load sample: {_e}")
+
+# WhisperCast: model size + input device
+if audio_mode == "WhisperCast":
+    st.markdown("#### Live Speech Recognition (faster-whisper)")
+    _whisper_default_model = saved_settings.get("whispercast_model", "tiny.en")
+    _whisper_default_device = saved_settings.get("whispercast_device", "ch1")
+    col_wm, col_wd = st.columns([2, 1])
+    with col_wm:
+        whisper_model_size = st.selectbox(
+            "Whisper Model",
+            ["tiny.en", "base.en", "small.en"],
+            index=["tiny.en", "base.en", "small.en"].index(_whisper_default_model)
+                  if _whisper_default_model in ["tiny.en", "base.en", "small.en"] else 0,
+            disabled=whispercast_is_streaming,
+            help="tiny.en (~39 MB, ~3-5s latency), base.en (~74 MB, ~5-8s latency)",
+        )
+    with col_wd:
+        whisper_device = st.selectbox(
+            "Input",
+            ["ch1", "ch2"],
+            index=0 if _whisper_default_device == "ch1" else 1,
+            disabled=whispercast_is_streaming,
+            help="Analog input channel",
+        )
+    st.caption("Model downloads automatically on first use. Each sentence appears after ~3s of speech.")
+
+# VoskCast: model path + input device
+if audio_mode == "VoskCast":
+    st.markdown("#### Live Speech Recognition (Vosk)")
+    _vosk_default_model = saved_settings.get("voskcast_model") or os.environ.get("VOSK_MODEL_PATH", "~/vosk-model-en-us")
+    _vosk_default_device = saved_settings.get("voskcast_device", "ch1")
+    col_model, col_dev = st.columns([3, 1])
+    with col_model:
+        vosk_model_path = st.text_input(
+            "Vosk Model Path",
+            value=_vosk_default_model,
+            disabled=voskcast_is_streaming,
+            help="Local path to the Vosk model directory. Download from https://alphacephei.com/vosk/models",
+        )
+    with col_dev:
+        vosk_device = st.selectbox(
+            "Input",
+            ["ch1", "ch2"],
+            index=0 if _vosk_default_device == "ch1" else 1,
+            disabled=voskcast_is_streaming,
+            help="Analog input channel (ch1 = left, ch2 = right)",
+        )
+    st.caption("Partial results appear immediately; final results are held for 4 s then cleared.")
+
 # 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)
@@ -355,6 +474,17 @@ 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(
@@ -398,6 +528,22 @@ if audio_mode == "Demo":
             disabled=is_streaming,
             help="Fast: 2 retransmissions, lower latency. Robust: 4 retransmissions, better reliability."
         )
+    # Per-radio TX power for Demo (independent for R1 and R2)
+    col_tx_r1, col_tx_r2 = st.columns(2, gap="small")
+    with col_tx_r1:
+        tx_power_r1 = _tx_power_selectbox(
+            "TX Power (R1)",
+            key="demo_tx_power_r1",
+            default=saved_settings.get('advertising_tx_power', TX_POWER_DEFAULT),
+            disabled=is_streaming,
+        )
+    with col_tx_r2:
+        tx_power_r2 = _tx_power_selectbox(
+            "TX Power (R2)",
+            key="demo_tx_power_r2",
+            default=saved_settings.get('secondary', {}).get('advertising_tx_power', TX_POWER_DEFAULT),
+            disabled=is_streaming,
+        )
     #st.info(f"Demo mode selected: {demo_selected} (Streams: {demo_stream_map[demo_selected]['streams']}, Rate: {demo_stream_map[demo_selected]['rate']} Hz)")
     quality = None  # Not used in demo mode
 else:
@@ -524,6 +670,13 @@ else:
                     help="Fast: 2 retransmissions, lower latency. Robust: 4 retransmissions, better reliability."
                 )
 
+            tx_power_r1 = _tx_power_selectbox(
+                "TX Power (R1)",
+                key="analog_tx_power_r1",
+                default=saved_settings.get('advertising_tx_power', TX_POWER_DEFAULT),
+                disabled=is_streaming,
+            )
+
             col_r1_name, col_r1_lang = st.columns([2, 1])
             with col_r1_name:
                 stream_name1 = st.text_input(
@@ -726,6 +879,13 @@ else:
                         help="Fast: 2 retransmissions, lower latency. Robust: 4 retransmissions, better reliability."
                     )
 
+                tx_power_r2 = _tx_power_selectbox(
+                    "TX Power (R2)",
+                    key="analog_tx_power_r2",
+                    default=saved_settings.get('secondary', {}).get('advertising_tx_power', TX_POWER_DEFAULT),
+                    disabled=is_streaming,
+                )
+
                 col_r2_name, col_r2_lang = st.columns([2, 1])
                 with col_r2_name:
                     stream_name2 = st.text_input(
@@ -785,6 +945,7 @@ else:
                     'immediate_rendering': immediate_rendering2,
                     'presentation_delay_ms': presentation_delay_ms2,
                     'qos_preset': qos_preset2,
+                    'tx_power': tx_power_r2,
                     'analog_gain_db_left': analog_gain_db_left,
                     'analog_gain_db_right': analog_gain_db_right,
                 }
@@ -801,6 +962,7 @@ else:
             'immediate_rendering': immediate_rendering1,
             'presentation_delay_ms': presentation_delay_ms1,
             'qos_preset': qos_preset1,
+            'tx_power': tx_power_r1,
             'stereo_mode': stereo_enabled,
             'analog_gain_db_left': analog_gain_db_left,
             'analog_gain_db_right': analog_gain_db_right,
@@ -1018,7 +1180,14 @@ else:
                     disabled=is_streaming,
                     help="Quality of Service preset for Radio 1"
                 )
-            
+
+            r1_tx_power = _tx_power_selectbox(
+                "TX Power (R1)",
+                key="dante_tx_power_r1",
+                default=saved_r1_config.get('advertising_tx_power', saved_settings.get('advertising_tx_power', TX_POWER_DEFAULT)),
+                disabled=is_streaming,
+            )
+
             # Per-stream configuration for Radio 1
             if dante_stereo_enabled:
                 st.write("**Stereo Stream Configuration (Radio 1)**")
@@ -1344,7 +1513,14 @@ else:
                         disabled=is_streaming,
                         help="Quality of Service preset for Radio 2"
                     )
-                
+
+                r2_tx_power = _tx_power_selectbox(
+                    "TX Power (R2)",
+                    key="dante_tx_power_r2",
+                    default=saved_r2_config.get('advertising_tx_power', saved_settings.get('secondary', {}).get('advertising_tx_power', TX_POWER_DEFAULT)),
+                    disabled=is_streaming,
+                )
+
                 # Per-stream configuration for Radio 2
                 st.write("**Stream Configuration (Radio 2)**")
                 r2_streams = []
@@ -1471,6 +1647,7 @@ else:
                 r2_immediate_rendering = False
                 r2_presentation_delay_ms = 40
                 r2_qos_preset = 'Fast'
+                r2_tx_power = TX_POWER_DEFAULT
         
         # Validate unique input devices for Network - Dante mode
         if audio_mode == "Network - Dante":
@@ -1502,6 +1679,7 @@ else:
             'immediate_rendering': r1_immediate_rendering,
             'presentation_delay_ms': r1_presentation_delay_ms,
             'qos_preset': r1_qos_preset,
+            'tx_power': r1_tx_power,
             'dante_stereo_mode': dante_stereo_enabled,
             'dante_stereo_left': dante_left_channel,
             'dante_stereo_right': dante_right_channel,
@@ -1517,6 +1695,7 @@ else:
             'immediate_rendering': r2_immediate_rendering if radio2_enabled else False,
             'presentation_delay_ms': r2_presentation_delay_ms if radio2_enabled else 40000,
             'qos_preset': r2_qos_preset if radio2_enabled else 'Fast',
+            'tx_power': r2_tx_power if radio2_enabled else TX_POWER_DEFAULT,
         } if radio2_enabled else None
         
     if audio_mode in ("USB", "Network"):
@@ -1595,6 +1774,13 @@ else:
                 help="Fast: 2 retransmissions, lower latency. Robust: 4 retransmissions, better reliability."
             )
 
+        tx_power = _tx_power_selectbox(
+            "TX Power",
+            key="usb_tx_power",
+            default=saved_settings.get('advertising_tx_power', TX_POWER_DEFAULT),
+            disabled=is_streaming,
+        )
+
         stream_name = st.text_input(
             "Channel Name",
             value=default_name,
@@ -1696,22 +1882,78 @@ else:
 if stop_stream:
     st.session_state['stream_started'] = False
     try:
-        r = requests.post(f"{BACKEND_URL}/stop_audio").json()
-        if audio_mode == "Demo":
-            st.session_state['demo_stream_started'] = False
-        if r['was_running']:
+        if audio_mode == "TextCast":
+            r = requests.post(f"{BACKEND_URL}/stop_textcast").json()
+        elif audio_mode == "VoskCast":
+            r = requests.post(f"{BACKEND_URL}/stop_voskcast").json()
+        elif audio_mode == "WhisperCast":
+            r = requests.post(f"{BACKEND_URL}/stop_whispercast").json()
+        else:
+            r = requests.post(f"{BACKEND_URL}/stop_audio").json()
+            if audio_mode == "Demo":
+                st.session_state['demo_stream_started'] = False
+        if r.get('was_running'):
             is_stopped = True
     except Exception as e:
         st.error(f"Error: {e}")
         
 
 if start_stream:
-    # Always send stop to ensure backend is in a clean state, regardless of current status
-    r = requests.post(f"{BACKEND_URL}/stop_audio").json()
-    # Small pause lets backend fully release audio devices before re-init
-    time.sleep(1)
+    if audio_mode == "TextCast":
+        uploaded = st.session_state.get('_textcast_dcp_content')
+        if not uploaded:
+            st.error("Upload a DCP XML file first.")
+        else:
+            try:
+                ru = requests.post(f"{BACKEND_URL}/upload_dcp", json={"xml": uploaded})
+                if not ru.ok:
+                    st.error(f"Upload failed: {ru.text}")
+                else:
+                    rs = requests.post(f"{BACKEND_URL}/start_textcast")
+                    if rs.ok:
+                        st.success("TextCast started.")
+                        st.rerun()
+                    else:
+                        st.error(f"Start failed: {rs.text}")
+            except Exception as e:
+                st.error(f"Error: {e}")
 
-    if audio_mode == "Demo":
+    elif audio_mode == "VoskCast":
+        try:
+            rs = requests.post(
+                f"{BACKEND_URL}/start_voskcast",
+                json={"model": vosk_model_path, "device": vosk_device},
+            )
+            if rs.ok:
+                st.success("VoskCast started.")
+                st.rerun()
+            else:
+                st.error(f"Start failed: {rs.text}")
+        except Exception as e:
+            st.error(f"Error: {e}")
+
+    elif audio_mode == "WhisperCast":
+        try:
+            rs = requests.post(
+                f"{BACKEND_URL}/start_whispercast",
+                json={"model": whisper_model_size, "device": whisper_device},
+            )
+            if rs.ok:
+                st.success("WhisperCast started.")
+                st.rerun()
+            else:
+                st.error(f"Start failed: {rs.text}")
+        except Exception as e:
+            st.error(f"Error: {e}")
+
+    else:
+
+    # Always send stop to ensure backend is in a clean state, regardless of current status
+     r = requests.post(f"{BACKEND_URL}/stop_audio").json()
+     # Small pause lets backend fully release audio devices before re-init
+     time.sleep(1)
+
+     if audio_mode == "Demo":
         demo_cfg = demo_stream_map[demo_selected]
         q = QUALITY_MAP[demo_cfg['quality']]
 
@@ -1726,12 +1968,22 @@ 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}
@@ -1748,6 +2000,7 @@ if start_stream:
             immediate_rendering=immediate_rendering,
             presentation_delay_us=int(presentation_delay_ms * 1000),
             qos_config=QOS_PRESET_MAP[qos_preset],
+            advertising_tx_power=tx_power_r1,
             bigs=bigs1
         )
         config2 = None
@@ -1760,6 +2013,7 @@ if start_stream:
                 immediate_rendering=immediate_rendering,
                 presentation_delay_us=int(presentation_delay_ms * 1000),
                 qos_config=QOS_PRESET_MAP[qos_preset],
+                advertising_tx_power=tx_power_r2,
                 bigs=bigs2
             )
 
@@ -1803,6 +2057,7 @@ 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']],
+                advertising_tx_power=int(cfg.get('tx_power', TX_POWER_DEFAULT)),
                 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=[
@@ -1890,6 +2145,7 @@ if start_stream:
                 immediate_rendering=bool(radio_cfg['immediate_rendering']),
                 presentation_delay_us=int(radio_cfg['presentation_delay_ms'] * 1000),
                 qos_config=QOS_PRESET_MAP[radio_cfg['qos_preset']],
+                advertising_tx_power=int(radio_cfg.get('tx_power', TX_POWER_DEFAULT)),
                 bigs=bigs
             )
 
@@ -1914,7 +2170,7 @@ if start_stream:
                     st.error(f"Failed to initialize Dante Radio 2: {r2.text}")
         except Exception as e:
             st.error(f"Error while starting Dante radios: {e}")
-    if audio_mode not in ("Demo", "Analog", "Network - Dante"):
+    if audio_mode not in ("Demo", "Analog", "Network - Dante", "VoskCast", "WhisperCast", "TextCast"):
         # USB/Network: single config as before, using shared controls
         q = QUALITY_MAP[quality]
         config = auracast_config.AuracastConfigGroup(
@@ -1925,6 +2181,7 @@ if start_stream:
             immediate_rendering=immediate_rendering,
             presentation_delay_us=int(presentation_delay_ms * 1000),
             qos_config=QOS_PRESET_MAP[qos_preset],
+            advertising_tx_power=tx_power,
             bigs=[
                 auracast_config.AuracastBigConfig(
                     code=(stream_passwort.strip() or None),

src/auracast/server/multicast_server.py

@@ -209,6 +209,16 @@ multicaster1: multicast_control.Multicaster | None = None
 multicaster2: multicast_control.Multicaster | None = None
 _stream_lock = asyncio.Lock()  # serialize initialize/stop_audio on API side
 
+# TextCast state
+_textcast_task: asyncio.Task | None = None
+DCP_UPLOAD_PATH = os.path.join(os.path.dirname(__file__), 'uploaded_subtitles.xml')
+
+# VoskCast state
+_voskcast_task: asyncio.Task | None = None
+
+# WhisperCast state
+_whispercast_task: asyncio.Task | None = None
+
 # 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.
@@ -592,6 +602,13 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
         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"
@@ -614,13 +631,15 @@ async def init_radio(transport: str, conf: auracast_config.AuracastConfigGroup,
             '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),
+            'advertising_tx_power': getattr(conf, 'advertising_tx_power', 8),
             '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:
@@ -696,6 +715,208 @@ async def _stop_audio_impl():
         log.error("Exception in /stop_audio: %s", traceback.format_exc())
         raise HTTPException(status_code=500, detail=str(e))
 
+@app.post("/upload_dcp")
+async def upload_dcp(payload: dict):
+    """Save DCP XML content for TextCast. Body: {"xml": "<DCSubtitle>..."}"""
+    xml_content = payload.get("xml", "")
+    if not xml_content.strip():
+        raise HTTPException(status_code=400, detail="Empty XML content")
+    try:
+        with open(DCP_UPLOAD_PATH, 'w', encoding='utf-8') as f:
+            f.write(xml_content)
+        log.info("DCP XML saved to %s (%d bytes)", DCP_UPLOAD_PATH, len(xml_content))
+        return {"status": "ok", "path": DCP_UPLOAD_PATH}
+    except Exception as e:
+        raise HTTPException(status_code=500, detail=str(e))
+
+
+@app.post("/start_textcast")
+async def start_textcast():
+    """Start text-over-Auracast broadcast using the uploaded DCP XML file."""
+    return await _on_ble_loop(_start_textcast_impl())
+
+
+async def _start_textcast_impl():
+    global _textcast_task
+    if not os.path.exists(DCP_UPLOAD_PATH):
+        raise HTTPException(status_code=400, detail="No DCP file uploaded. Use /upload_dcp first.")
+
+    # Stop any running audio/textcast first
+    await _stop_all()
+    await _stop_textcast_impl()
+
+    from auracast.text_multicast import broadcast_text
+    _textcast_task = asyncio.get_event_loop().create_task(
+        broadcast_text(DCP_UPLOAD_PATH, TRANSPORT1)
+    )
+
+    settings = {
+        'is_streaming': True,
+        'audio_mode': 'TextCast',
+        'textcast_is_streaming': True,
+        'timestamp': datetime.utcnow().isoformat(),
+    }
+    save_stream_settings(settings)
+    _led_on()
+    log.info("TextCast started (DCP: %s)", DCP_UPLOAD_PATH)
+    return {"status": "started"}
+
+
+@app.post("/stop_textcast")
+async def stop_textcast():
+    """Stop an active TextCast broadcast."""
+    return await _on_ble_loop(_stop_textcast_impl())
+
+
+async def _stop_textcast_impl():
+    global _textcast_task
+    was_running = False
+    if _textcast_task is not None and not _textcast_task.done():
+        was_running = True
+        _textcast_task.cancel()
+        try:
+            await asyncio.wait_for(asyncio.shield(_textcast_task), timeout=3.0)
+        except (asyncio.CancelledError, asyncio.TimeoutError, Exception):
+            pass
+        _textcast_task = None
+        _led_off()
+        settings = load_stream_settings() or {}
+        if settings.get('audio_mode') == 'TextCast':
+            settings['is_streaming'] = False
+            settings['textcast_is_streaming'] = False
+            settings['timestamp'] = datetime.utcnow().isoformat()
+            save_stream_settings(settings)
+        log.info("TextCast stopped")
+    return {"status": "stopped", "was_running": was_running}
+
+
+@app.post("/start_voskcast")
+async def start_voskcast(body: dict = {}):
+    """Start Vosk STT → TextCast. Body (optional): {"model": "...", "device": "ch1"}"""
+    return await _on_ble_loop(_start_voskcast_impl(body))
+
+
+async def _start_voskcast_impl(body: dict) -> dict:
+    global _voskcast_task
+    from auracast.vosk_textcast import broadcast_vosk, DEFAULT_MODEL_PATH
+
+    model = body.get('model') or DEFAULT_MODEL_PATH
+    device = body.get('device', 'ch1')
+
+    await _stop_all()
+    await _stop_textcast_impl()
+    await _stop_voskcast_impl()
+
+    _voskcast_task = asyncio.get_event_loop().create_task(
+        broadcast_vosk(TRANSPORT1, model, device)
+    )
+
+    settings = {
+        'is_streaming': True,
+        'audio_mode': 'VoskCast',
+        'voskcast_is_streaming': True,
+        'voskcast_device': device,
+        'voskcast_model': model,
+        'timestamp': datetime.utcnow().isoformat(),
+    }
+    save_stream_settings(settings)
+    _led_on()
+    log.info("VoskCast started (device=%s, model=%s)", device, model)
+    return {"status": "started"}
+
+
+@app.post("/stop_voskcast")
+async def stop_voskcast():
+    """Stop an active VoskCast broadcast."""
+    return await _on_ble_loop(_stop_voskcast_impl())
+
+
+async def _stop_voskcast_impl() -> dict:
+    global _voskcast_task
+    was_running = False
+    if _voskcast_task is not None and not _voskcast_task.done():
+        was_running = True
+        _voskcast_task.cancel()
+        try:
+            await asyncio.wait_for(asyncio.shield(_voskcast_task), timeout=4.0)
+        except (asyncio.CancelledError, asyncio.TimeoutError, Exception):
+            pass
+        _voskcast_task = None
+        _led_off()
+        settings = load_stream_settings() or {}
+        if settings.get('audio_mode') == 'VoskCast':
+            settings['is_streaming'] = False
+            settings['voskcast_is_streaming'] = False
+            settings['timestamp'] = datetime.utcnow().isoformat()
+            save_stream_settings(settings)
+        log.info("VoskCast stopped")
+    return {"status": "stopped", "was_running": was_running}
+
+
+@app.post("/start_whispercast")
+async def start_whispercast(body: dict = {}):
+    """Start faster-whisper → TextCast. Body (optional): {"model": "tiny.en", "device": "ch1"}"""
+    return await _on_ble_loop(_start_whispercast_impl(body))
+
+
+async def _start_whispercast_impl(body: dict) -> dict:
+    global _whispercast_task
+    from auracast.faster_whisper_textcast import broadcast_whisper
+
+    model = body.get('model', 'tiny.en')
+    device = body.get('device', 'ch1')
+
+    await _stop_all()
+    await _stop_textcast_impl()
+    await _stop_voskcast_impl()
+    await _stop_whispercast_impl()
+
+    _whispercast_task = asyncio.get_event_loop().create_task(
+        broadcast_whisper(TRANSPORT1, model, device)
+    )
+
+    settings = {
+        'is_streaming': True,
+        'audio_mode': 'WhisperCast',
+        'whispercast_is_streaming': True,
+        'whispercast_device': device,
+        'whispercast_model': model,
+        'timestamp': datetime.utcnow().isoformat(),
+    }
+    save_stream_settings(settings)
+    _led_on()
+    log.info("WhisperCast started (device=%s, model=%s)", device, model)
+    return {"status": "started"}
+
+
+@app.post("/stop_whispercast")
+async def stop_whispercast():
+    """Stop an active WhisperCast broadcast."""
+    return await _on_ble_loop(_stop_whispercast_impl())
+
+
+async def _stop_whispercast_impl() -> dict:
+    global _whispercast_task
+    was_running = False
+    if _whispercast_task is not None and not _whispercast_task.done():
+        was_running = True
+        _whispercast_task.cancel()
+        try:
+            await asyncio.wait_for(asyncio.shield(_whispercast_task), timeout=5.0)
+        except (asyncio.CancelledError, asyncio.TimeoutError, Exception):
+            pass
+        _whispercast_task = None
+        _led_off()
+        settings = load_stream_settings() or {}
+        if settings.get('audio_mode') == 'WhisperCast':
+            settings['is_streaming'] = False
+            settings['whispercast_is_streaming'] = False
+            settings['timestamp'] = datetime.utcnow().isoformat()
+            save_stream_settings(settings)
+        log.info("WhisperCast stopped")
+    return {"status": "stopped", "was_running": was_running}
+
+
 @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.
@@ -754,6 +975,15 @@ async def get_status():
     status["secondary"] = secondary
     status["secondary_is_streaming"] = bool(secondary.get("is_streaming", False))
     status["led_enabled"] = _LED_ENABLED
+    status["textcast_is_streaming"] = (
+        _textcast_task is not None and not _textcast_task.done()
+    )
+    status["voskcast_is_streaming"] = (
+        _voskcast_task is not None and not _voskcast_task.done()
+    )
+    status["whispercast_is_streaming"] = (
+        _whispercast_task is not None and not _whispercast_task.done()
+    )
 
     return status
 
@@ -794,11 +1024,12 @@ async def _autostart_from_settings():
         big_ids = settings.get('big_ids') or []
         big_addrs = settings.get('big_random_addresses') or []
         stream_password = settings.get('stream_password')
+        tx_power = int(settings.get('advertising_tx_power', 8))
         original_ts = settings.get('timestamp')
         previously_streaming = bool(settings.get('is_streaming'))
 
         log.info(
-            "[AUTOSTART][PRIMARY] loaded settings: previously_streaming=%s audio_mode=%s rate=%s octets=%s pres_delay=%s qos_preset=%s immediate_rendering=%s assisted_listening_stream=%s demo_sources=%s",
+            "[AUTOSTART][PRIMARY] loaded settings: previously_streaming=%s audio_mode=%s rate=%s octets=%s pres_delay=%s qos_preset=%s immediate_rendering=%s assisted_listening_stream=%s tx_power=%+d dBm demo_sources=%s",
             previously_streaming,
             audio_mode,
             rate,
@@ -807,6 +1038,7 @@ async def _autostart_from_settings():
             saved_qos_preset,
             immediate_rendering,
             assisted_listening_stream,
+            tx_power,
             (settings.get('demo_sources') or []),
         )
 
@@ -856,6 +1088,7 @@ 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,
+                advertising_tx_power=tx_power,
                 bigs=bigs,
             )
             # Set num_bis for stereo mode if needed
@@ -925,6 +1158,7 @@ async def _autostart_from_settings():
                     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),
+                    advertising_tx_power=tx_power,
                     bigs=bigs,
                 )
                 # Set num_bis for stereo mode if needed
@@ -960,10 +1194,11 @@ async def _autostart_from_settings():
         big_ids = settings.get('big_ids') or []
         big_addrs = settings.get('big_random_addresses') or []
         stream_password = settings.get('stream_password')
+        tx_power = int(settings.get('advertising_tx_power', 8))
         original_ts = settings.get('timestamp')
         previously_streaming = bool(settings.get('is_streaming'))
         log.info(
-            "[AUTOSTART][SECONDARY] loaded settings: previously_streaming=%s audio_mode=%s rate=%s octets=%s pres_delay=%s qos_preset=%s immediate_rendering=%s assisted_listening_stream=%s demo_sources=%s",
+            "[AUTOSTART][SECONDARY] loaded settings: previously_streaming=%s audio_mode=%s rate=%s octets=%s pres_delay=%s qos_preset=%s immediate_rendering=%s assisted_listening_stream=%s tx_power=%+d dBm demo_sources=%s",
             previously_streaming,
             audio_mode,
             rate,
@@ -972,6 +1207,7 @@ async def _autostart_from_settings():
             saved_qos_preset,
             immediate_rendering,
             assisted_listening_stream,
+            tx_power,
             (settings.get('demo_sources') or []),
         )
         if not previously_streaming:
@@ -1011,6 +1247,7 @@ 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,
+                advertising_tx_power=tx_power,
                 bigs=bigs,
             )
             conf.qos_config = QOS_PRESET_MAP.get(saved_qos_preset, QOS_PRESET_MAP["Fast"])
@@ -1080,6 +1317,7 @@ async def _autostart_from_settings():
                     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),
+                    advertising_tx_power=tx_power,
                     bigs=bigs,
                 )
                 conf.qos_config = QOS_PRESET_MAP.get(saved_qos_preset, QOS_PRESET_MAP["Fast"])

src/auracast/testdata/sample_subtitles.xml

@@ -0,0 +1,71 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<DCSubtitle Version="1.0">
+  <SubtitleID>a1b2c3d4-e5f6-7890-abcd-ef1234567890</SubtitleID>
+  <MovieTitle>Sample TextCast Subtitles</MovieTitle>
+  <ReelNumber>1</ReelNumber>
+  <Language>en</Language>
+  <LoadFont Id="Font1" URI="Arial.ttf"/>
+  <Font Id="Font1" Color="FFFFFFFF" Effect="none" Size="42" Italic="no">
+
+    <Subtitle SpotNumber="1" TimeIn="00:00:02:00" TimeOut="00:00:05:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Welcome to TextCast.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="2" TimeIn="00:00:06:00" TimeOut="00:00:09:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Text transmitted over Auracast BLE.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="3" TimeIn="00:00:10:00" TimeOut="00:00:13:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">No LC3 audio codec involved.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="4" TimeIn="00:00:14:00" TimeOut="00:00:17:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Raw ISO SDUs carry UTF-8 text.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="5" TimeIn="00:00:18:00" TimeOut="00:00:21:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">100 frames per second at 40 bytes.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="6" TimeIn="00:00:22:00" TimeOut="00:00:25:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Scrolling display on SH1106 OLED.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="7" TimeIn="00:00:26:00" TimeOut="00:00:29:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Each new line scrolls up the screen.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="8" TimeIn="00:00:30:00" TimeOut="00:00:33:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">The quick brown fox jumps over</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="9" TimeIn="00:00:34:00" TimeOut="00:00:37:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">the lazy dog.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="10" TimeIn="00:00:38:00" TimeOut="00:00:41:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Speech-to-text output goes here.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="11" TimeIn="00:00:42:00" TimeOut="00:00:45:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Latency is dominated by BLE BIG.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="12" TimeIn="00:00:46:00" TimeOut="00:00:49:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Typical end-to-end: under 50 ms.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="13" TimeIn="00:00:50:00" TimeOut="00:00:53:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">One transmitter, many receivers.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="14" TimeIn="00:00:54:00" TimeOut="00:00:57:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">Built on Bumble and Zephyr RTOS.</Text>
+    </Subtitle>
+
+    <Subtitle SpotNumber="15" TimeIn="00:00:58:00" TimeOut="00:01:01:00" FadeUpTime="0" FadeDownTime="0">
+      <Text HAlign="center" VAlign="bottom">End of demonstration. Thank you.</Text>
+    </Subtitle>
+
+  </Font>
+</DCSubtitle>

src/auracast/text_multicast.py

@@ -0,0 +1,155 @@
+"""Text-over-Auracast transmitter.
+
+Reads a DCP XML subtitle file and broadcasts each subtitle as raw ISO SDUs.
+No LC3 encoding is used.  The BIG is advertised with codec_id=LC3 (required
+for BAP sync) but the SDU payload is plain UTF-8 text with a magic header.
+
+Frame format (SDU_SIZE bytes total):
+  Byte 0      : TEXT_MAGIC (0xAA)  – identifies this as a text SDU
+  Byte 1      : text length N      – 0 means idle/clear
+  Bytes 2..N+1: UTF-8 text
+  Bytes N+2.. : zero padding to SDU_SIZE
+
+Usage:
+    poetry run python -m auracast.text_multicast \\
+        --dcp ./auracast/testdata/sample_subtitles.xml \\
+        --transport serial:/dev/ttyAMA3,1000000,rtscts
+"""
+from __future__ import annotations
+
+import argparse
+import asyncio
+import logging
+import os
+
+from auracast import auracast_config, multicast
+from auracast.dcp_parser import parse_dcp_xml
+
+TEXT_MAGIC = 0xAA
+SDU_SIZE = 64            # octets_per_frame; 62 usable text bytes per frame
+SDU_INTERVAL_US = 10_000 # 10 ms → 100 SDUs/sec
+BROADCAST_NAME = 'TextCast'
+
+logging.basicConfig(
+    level=logging.INFO,
+    format='%(asctime)s %(levelname)s %(name)s: %(message)s',
+)
+log = logging.getLogger('text_multicast')
+
+
+def _make_text_frame(text: str) -> bytes:
+    """Encode a subtitle string into a fixed-size TEXT SDU."""
+    text_bytes = text.encode('utf-8')[: SDU_SIZE - 2]
+    frame = bytes([TEXT_MAGIC, len(text_bytes)]) + text_bytes
+    return frame + bytes(SDU_SIZE - len(frame))
+
+
+def _make_idle_frame() -> bytes:
+    """Return an idle frame (magic=0, signals 'no active subtitle')."""
+    return bytes(SDU_SIZE)
+
+
+async def _text_stream(bigs: dict, subtitles: list, loop: bool = True) -> None:
+    """Main text streaming loop.
+
+    Writes one SDU every ~10 ms (flow-controlled by the BLE controller).
+    Subtitle timing is derived from the frame counter: frame N ≈ N × 10 ms.
+    When *loop* is True (default) the subtitle list repeats indefinitely.
+    """
+    iso_queue = bigs['big0']['iso_queue']
+    frame_interval_s = SDU_INTERVAL_US / 1_000_000
+    frame_count = 0
+    sub_idx = 0
+    n = len(subtitles)
+    last_log_sub = -1
+    loop_count = 0
+    # Total duration of one pass: end of last subtitle + 2 s gap before restart
+    _loop_gap_s = 2.0
+    _pass_duration_s = subtitles[-1].time_out + _loop_gap_s if n > 0 else 0.0
+
+    log.info("Streaming %d subtitle(s) (loop=%s).  Press Ctrl-C to stop.", n, loop)
+
+    while True:
+        now_s = frame_count * frame_interval_s
+
+        # Advance past subtitles whose time_out has passed
+        while sub_idx < n and now_s >= subtitles[sub_idx].time_out:
+            sub_idx += 1
+
+        # Determine what to send
+        if sub_idx < n and now_s >= subtitles[sub_idx].time_in:
+            frame = _make_text_frame(subtitles[sub_idx].text)
+            if sub_idx != last_log_sub:
+                log.info("[loop %d %05.1fs] %s", loop_count, now_s, subtitles[sub_idx].text)
+                last_log_sub = sub_idx
+        else:
+            frame = _make_idle_frame()
+
+        await iso_queue.write(frame)
+        frame_count += 1
+
+        # End of pass
+        if n > 0 and now_s >= _pass_duration_s:
+            if loop:
+                loop_count += 1
+                log.info("Loop %d complete – restarting.", loop_count)
+                frame_count = 0
+                sub_idx = 0
+                last_log_sub = -1
+            else:
+                log.info("All subtitles transmitted.  Exiting.")
+                break
+
+
+async def broadcast_text(dcp_path: str, transport: str, loop: bool = True) -> None:
+    subtitles = parse_dcp_xml(dcp_path)
+    if not subtitles:
+        log.error("No subtitles found in %s", dcp_path)
+        return
+    log.info("Loaded %d subtitle(s) from %s", len(subtitles), dcp_path)
+
+    config = auracast_config.AuracastConfigGroup(
+        bigs=[
+            auracast_config.AuracastBigConfig(
+                name=BROADCAST_NAME,
+                program_info='Text Broadcast',
+                language='eng',
+                audio_source='file:dummy',  # not used – streamer loop is replaced
+                iso_que_len=4,
+            ),
+        ],
+        auracast_sampling_rate_hz=16000,
+        octets_per_frame=SDU_SIZE,
+        frame_duration_us=SDU_INTERVAL_US,
+        presentation_delay_us=40_000,
+        qos_config=auracast_config.AuracastQosRobust(),
+        transport=transport,
+    )
+
+    async with multicast.create_device(config) as device:
+        bigs = await multicast.init_broadcast(device, config, config.bigs)
+        await _text_stream(bigs, subtitles, loop=loop)
+
+
+def main() -> None:
+    parser = argparse.ArgumentParser(description='Auracast text (subtitle) transmitter')
+    parser.add_argument('--dcp', required=True, help='Path to DCP XML subtitle file')
+    parser.add_argument(
+        '--transport',
+        default=os.environ.get(
+            'AURACAST_TRANSPORT',
+            'serial:/dev/ttyAMA3,1000000,rtscts',
+        ),
+        help='Bumble HCI transport string (default: $AURACAST_TRANSPORT or ttyAMA3)',
+    )
+    parser.add_argument(
+        '--no-loop',
+        action='store_true',
+        help='Play subtitles once and exit instead of looping indefinitely',
+    )
+    args = parser.parse_args()
+    multicast.run_async(broadcast_text(args.dcp, args.transport, loop=not args.no_loop))
+
+
+if __name__ == '__main__':
+    main()

src/auracast/utils/read_lc3_file.py

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

src/auracast/vosk_textcast.py

@@ -0,0 +1,270 @@
+"""Vosk speech-to-text → TextCast streamer.
+
+Captures mono audio from an analog ALSA/sounddevice input, runs Vosk
+offline ASR in a background thread, and broadcasts recognised text over
+the TextCast BLE broadcast using the same SDU framing as text_multicast.py.
+
+Usage (CLI):
+    poetry run python -m auracast.vosk_textcast \\
+        --model /path/to/vosk-model-en-us \\
+        --device ch1 \\
+        --transport serial:/dev/ttyAMA3,1000000,rtscts
+
+Environment:
+    VOSK_MODEL_PATH   – default Vosk model directory
+    AURACAST_TRANSPORT – default HCI transport string
+"""
+from __future__ import annotations
+
+import argparse
+import asyncio
+import json
+import logging
+import os
+import threading
+import time
+from typing import Optional
+
+import numpy as np
+import samplerate
+import sounddevice as sd
+
+from auracast import auracast_config, multicast
+from auracast.text_multicast import (
+    SDU_SIZE,
+    SDU_INTERVAL_US,
+    _make_text_frame,
+    _make_idle_frame,
+)
+
+log = logging.getLogger('vosk_textcast')
+
+VOSK_SAMPLE_RATE = 16_000   # Vosk models expect 16 kHz
+CAPTURE_SAMPLE_RATE = 48_000 # Hardware capture rate (always 48 kHz)
+BLOCK_FRAMES_48K = 4800      # 100 ms blocks at 48 kHz → 1600 frames at 16 kHz
+CAPTION_HOLD_S = 4.0         # Keep caption visible N seconds after last speech
+BROADCAST_NAME = 'LiveCaption'
+
+DEFAULT_MODEL_PATH = os.environ.get(
+    'VOSK_MODEL_PATH',
+    os.path.expanduser('~/vosk-model-en-us'),
+)
+
+
+def _tail_to_fit(text: str, max_bytes: int) -> str:
+    """Return the tail of *text* that fits in *max_bytes* UTF-8 bytes."""
+    encoded = text.encode('utf-8')
+    if len(encoded) <= max_bytes:
+        return text
+    tail = encoded[-max_bytes:].decode('utf-8', errors='ignore')
+    sp = tail.find(' ')
+    return tail[sp + 1:] if sp != -1 else tail
+
+
+def _new_words(old: str, new: str) -> str:
+    """Return the words appended to *new* beyond the shared prefix with *old*.
+
+    If *new* doesn't start with *old* (different utterance), return *new* in full.
+    """
+    old_words = old.split()
+    new_words = new.split()
+    if new_words[:len(old_words)] == old_words:
+        extra = new_words[len(old_words):]
+        return ' '.join(extra)
+    return new
+
+
+def _resolve_device(device: str) -> Optional[int]:
+    """Return sounddevice index for a name or numeric string, or None for default."""
+    if not device:
+        return None
+    if device.isdigit():
+        return int(device)
+    for i, d in enumerate(sd.query_devices()):
+        if d['name'] == device and d['max_input_channels'] > 0:
+            return i
+    log.warning("Device '%s' not found in sounddevice list – using default input", device)
+    return None
+
+
+async def _iso_write_loop(bigs: dict, shared: dict, lock: threading.Lock) -> None:
+    """ISO SDU write loop.
+
+    Runs at ~10 ms per iteration (flow-controlled by the BLE controller).
+    Sends the current recognised text (partial or final) as-is.
+    """
+    iso_queue = bigs['big0']['iso_queue']
+    last_sent: str = ''
+
+    while True:
+        now = time.monotonic()
+        with lock:
+            text: str = shared.get('text', '')
+            expiry: float = shared.get('expiry', 0.0)
+
+        if text and now < expiry:
+            display_text = _tail_to_fit(text, SDU_SIZE - 2)
+            if display_text != last_sent:
+                log.info("Caption: %s", display_text)
+                last_sent = display_text
+            frame = _make_text_frame(display_text)
+        else:
+            if last_sent:
+                log.info("Caption cleared")
+                last_sent = ''
+                with lock:
+                    shared['text'] = ''
+            frame = _make_idle_frame()
+
+        await iso_queue.write(frame)
+
+
+def _vosk_thread(
+    model_path: str,
+    device: str,
+    shared: dict,
+    lock: threading.Lock,
+    stop_event: threading.Event,
+) -> None:
+    """Blocking audio capture + Vosk recognition loop.  Runs in a daemon thread."""
+    try:
+        from vosk import KaldiRecognizer, Model  # type: ignore
+    except ImportError:
+        log.error("vosk is not installed. Run: poetry add vosk")
+        return
+
+    log.info("Loading Vosk model from %s …", model_path)
+    model = Model(model_path)
+    rec = KaldiRecognizer(model, VOSK_SAMPLE_RATE)
+    rec.SetMaxAlternatives(0)
+    rec.SetWords(False)
+
+    resampler = samplerate.Resampler('sinc_fastest', channels=1)
+    ratio = VOSK_SAMPLE_RATE / CAPTURE_SAMPLE_RATE
+
+    dev_idx = _resolve_device(device)
+    last_word_count = [0]  # word count of last partial sent to display
+
+    def _cb(indata: np.ndarray, frames: int, time_info, status) -> None:
+        if status:
+            log.warning("Audio status: %s", status)
+        if stop_event.is_set():
+            raise sd.CallbackStop()
+
+        # Resample 48 kHz → 16 kHz
+        mono = indata[:, 0].astype(np.float32)
+        downsampled = resampler.process(mono, ratio, end_of_input=False)
+        pcm16 = (downsampled * 32767).astype(np.int16).tobytes()
+
+        if rec.AcceptWaveform(pcm16):
+            result = json.loads(rec.Result())
+            final_text = result.get('text', '').strip()
+            if final_text:
+                log.info("Final: %s", final_text)
+                with lock:
+                    shared['text'] = _tail_to_fit(final_text, SDU_SIZE - 2)
+                    shared['expiry'] = time.monotonic() + CAPTION_HOLD_S
+                last_word_count[0] = 0  # reset for next sentence
+        else:
+            partial_text = json.loads(rec.PartialResult()).get('partial', '').strip()
+            if partial_text:
+                wc = len(partial_text.split())
+                if wc > last_word_count[0]:  # new word arrived
+                    last_word_count[0] = wc
+                    with lock:
+                        shared['text'] = _tail_to_fit(partial_text, SDU_SIZE - 2)
+                        shared['expiry'] = time.monotonic() + CAPTION_HOLD_S
+
+    try:
+        with sd.InputStream(
+            samplerate=CAPTURE_SAMPLE_RATE,
+            blocksize=BLOCK_FRAMES_48K,
+            device=dev_idx,
+            dtype='float32',
+            channels=1,
+            callback=_cb,
+        ):
+            log.info("Vosk listening on device '%s' (idx=%s) …", device, dev_idx)
+            stop_event.wait()
+    except Exception as exc:
+        log.error("Vosk audio thread error: %s", exc, exc_info=True)
+
+
+async def broadcast_vosk(
+    transport: str,
+    model_path: str = DEFAULT_MODEL_PATH,
+    device: str = 'ch1',
+) -> None:
+    """Start a Vosk STT → TextCast broadcast.  Runs until cancelled."""
+    model_path = os.path.expanduser(model_path)
+    if not os.path.exists(model_path):
+        raise FileNotFoundError(
+            f"Vosk model not found at '{model_path}'. "
+            "Download from https://alphacephei.com/vosk/models and set VOSK_MODEL_PATH."
+        )
+
+    config = auracast_config.AuracastConfigGroup(
+        bigs=[
+            auracast_config.AuracastBigConfig(
+                name=BROADCAST_NAME,
+                program_info='Live Captions',
+                language='eng',
+                audio_source='file:dummy',
+                iso_que_len=4,
+            ),
+        ],
+        auracast_sampling_rate_hz=16000,
+        octets_per_frame=SDU_SIZE,
+        frame_duration_us=SDU_INTERVAL_US,
+        presentation_delay_us=40_000,
+        qos_config=auracast_config.AuracastQosRobust(),
+        transport=transport,
+    )
+
+    shared: dict = {'text': '', 'expiry': 0.0}
+    lock = threading.Lock()
+    stop_event = threading.Event()
+
+    async with multicast.create_device(config) as ble_device:
+        bigs = await multicast.init_broadcast(ble_device, config, config.bigs)
+
+        t = threading.Thread(
+            target=_vosk_thread,
+            args=(model_path, device, shared, lock, stop_event),
+            daemon=True,
+        )
+        t.start()
+        log.info("VoskCast started (device=%s, model=%s)", device, model_path)
+
+        try:
+            await _iso_write_loop(bigs, shared, lock)
+        except asyncio.CancelledError:
+            log.info("VoskCast cancelled – shutting down")
+            stop_event.set()
+            t.join(timeout=3.0)
+            raise
+
+
+def main() -> None:
+    parser = argparse.ArgumentParser(description='Vosk STT → Auracast TextCast')
+    parser.add_argument(
+        '--model',
+        default=DEFAULT_MODEL_PATH,
+        help=f'Path to Vosk model directory (default: {DEFAULT_MODEL_PATH})',
+    )
+    parser.add_argument(
+        '--device',
+        default='ch1',
+        help='sounddevice input device name or index (default: ch1)',
+    )
+    parser.add_argument(
+        '--transport',
+        default=os.environ.get('AURACAST_TRANSPORT', 'serial:/dev/ttyAMA3,1000000,rtscts'),
+        help='Bumble HCI transport string',
+    )
+    args = parser.parse_args()
+    multicast.run_async(broadcast_vosk(args.transport, args.model, args.device))
+
+
+if __name__ == '__main__':
+    main()