Update engine

This commit is contained in:
Xin Wang
2026-02-23 17:16:18 +08:00
parent 01c0de0a4d
commit c6c84b5af9
9 changed files with 991 additions and 186 deletions

View File

@@ -59,8 +59,12 @@ class MicrophoneClient:
url: str,
sample_rate: int = 16000,
chunk_duration_ms: int = 20,
app_id: str = "assistant_demo",
channel: str = "mic_client",
config_version_id: str = "local-dev",
input_device: int = None,
output_device: int = None
output_device: int = None,
track_debug: bool = False,
):
"""
Initialize microphone client.
@@ -76,8 +80,12 @@ class MicrophoneClient:
self.sample_rate = sample_rate
self.chunk_duration_ms = chunk_duration_ms
self.chunk_samples = int(sample_rate * chunk_duration_ms / 1000)
self.app_id = app_id
self.channel = channel
self.config_version_id = config_version_id
self.input_device = input_device
self.output_device = output_device
self.track_debug = track_debug
# WebSocket connection
self.ws = None
@@ -106,6 +114,17 @@ class MicrophoneClient:
# Verbose mode for streaming LLM responses
self.verbose = False
@staticmethod
def _event_ids_suffix(event: dict) -> str:
data = event.get("data") if isinstance(event.get("data"), dict) else {}
keys = ("turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id")
parts = []
for key in keys:
value = data.get(key, event.get(key))
if value:
parts.append(f"{key}={value}")
return f" [{' '.join(parts)}]" if parts else ""
async def connect(self) -> None:
"""Connect to WebSocket server."""
@@ -114,20 +133,30 @@ class MicrophoneClient:
self.running = True
print("Connected!")
# Send invite command
# WS v1 handshake: hello -> session.start
await self.send_command({
"command": "invite",
"option": {
"codec": "pcm",
"sampleRate": self.sample_rate
}
"type": "hello",
"version": "v1",
})
await self.send_command({
"type": "session.start",
"audio": {
"encoding": "pcm_s16le",
"sample_rate_hz": self.sample_rate,
"channels": 1,
},
"metadata": {
"appId": self.app_id,
"channel": self.channel,
"configVersionId": self.config_version_id,
},
})
async def send_command(self, cmd: dict) -> None:
"""Send JSON command to server."""
if self.ws:
await self.ws.send(json.dumps(cmd))
print(f"→ Command: {cmd.get('command', 'unknown')}")
print(f"→ Command: {cmd.get('type', 'unknown')}")
async def send_chat(self, text: str) -> None:
"""Send chat message (text input)."""
@@ -136,7 +165,7 @@ class MicrophoneClient:
self.first_audio_received = False
await self.send_command({
"command": "chat",
"type": "input.text",
"text": text
})
print(f"→ Chat: {text}")
@@ -144,13 +173,14 @@ class MicrophoneClient:
async def send_interrupt(self) -> None:
"""Send interrupt command."""
await self.send_command({
"command": "interrupt"
"type": "response.cancel",
"graceful": False,
})
async def send_hangup(self, reason: str = "User quit") -> None:
"""Send hangup command."""
await self.send_command({
"command": "hangup",
"type": "session.stop",
"reason": reason
})
@@ -295,43 +325,48 @@ class MicrophoneClient:
async def _handle_event(self, event: dict) -> None:
"""Handle incoming event."""
event_type = event.get("event", "unknown")
event_type = event.get("type", event.get("event", "unknown"))
ids = self._event_ids_suffix(event)
if self.track_debug:
print(f"[track-debug] event={event_type} trackId={event.get('trackId')}{ids}")
if event_type == "answer":
print("← Session ready!")
elif event_type == "speaking":
print("User speech detected")
elif event_type == "silence":
print("← User silence detected")
elif event_type == "transcript":
if event_type in {"hello.ack", "session.started"}:
print(f"← Session ready!{ids}")
elif event_type == "config.resolved":
print(f"Config resolved: {event.get('config', {}).get('output', {})}{ids}")
elif event_type == "input.speech_started":
print(f"← User speech detected{ids}")
elif event_type == "input.speech_stopped":
print(f"← User silence detected{ids}")
elif event_type in {"transcript", "transcript.delta", "transcript.final"}:
# Display user speech transcription
text = event.get("text", "")
is_final = event.get("isFinal", False)
is_final = event_type == "transcript.final" or bool(event.get("isFinal"))
if is_final:
# Clear the interim line and print final
print(" " * 80, end="\r") # Clear previous interim text
print(f"→ You: {text}")
print(f"→ You: {text}{ids}")
else:
# Interim result - show with indicator (overwrite same line)
display_text = text[:60] + "..." if len(text) > 60 else text
print(f" [listening] {display_text}".ljust(80), end="\r")
elif event_type == "ttfb":
elif event_type in {"ttfb", "metrics.ttfb"}:
# Server-side TTFB event
latency_ms = event.get("latencyMs", 0)
print(f"← [TTFB] Server reported latency: {latency_ms}ms")
elif event_type == "llmResponse":
elif event_type in {"llmResponse", "assistant.response.delta", "assistant.response.final"}:
# LLM text response
text = event.get("text", "")
is_final = event.get("isFinal", False)
is_final = event_type == "assistant.response.final" or bool(event.get("isFinal"))
if is_final:
# Print final LLM response
print(f"← AI: {text}")
elif self.verbose:
# Show streaming chunks only in verbose mode
display_text = text[:60] + "..." if len(text) > 60 else text
print(f" [streaming] {display_text}")
elif event_type == "trackStart":
print("← Bot started speaking")
print(f" [streaming] {display_text}{ids}")
elif event_type in {"trackStart", "output.audio.start"}:
print(f"← Bot started speaking{ids}")
# IMPORTANT: Accept audio again after trackStart
self._discard_audio = False
self._audio_sequence += 1
@@ -342,13 +377,13 @@ class MicrophoneClient:
# Clear any old audio in buffer
with self.audio_output_lock:
self.audio_output_buffer = b""
elif event_type == "trackEnd":
print("← Bot finished speaking")
elif event_type in {"trackEnd", "output.audio.end"}:
print(f"← Bot finished speaking{ids}")
# Reset TTFB tracking after response completes
self.request_start_time = None
self.first_audio_received = False
elif event_type == "interrupt":
print("← Bot interrupted!")
elif event_type in {"interrupt", "response.interrupted"}:
print(f"← Bot interrupted!{ids}")
# IMPORTANT: Discard all audio until next trackStart
self._discard_audio = True
# Clear audio buffer immediately
@@ -357,12 +392,12 @@ class MicrophoneClient:
self.audio_output_buffer = b""
print(f" (cleared {buffer_ms:.0f}ms, discarding audio until new track)")
elif event_type == "error":
print(f"← Error: {event.get('error')}")
elif event_type == "hangup":
print(f"← Hangup: {event.get('reason')}")
print(f"← Error: {event.get('error')}{ids}")
elif event_type in {"hangup", "session.stopped"}:
print(f"← Hangup: {event.get('reason')}{ids}")
self.running = False
else:
print(f"← Event: {event_type}")
print(f"← Event: {event_type}{ids}")
async def interactive_mode(self) -> None:
"""Run interactive mode for text chat."""
@@ -573,6 +608,26 @@ async def main():
action="store_true",
help="Show streaming LLM response chunks"
)
parser.add_argument(
"--app-id",
default="assistant_demo",
help="Stable app/assistant identifier for server-side config lookup"
)
parser.add_argument(
"--channel",
default="mic_client",
help="Client channel name"
)
parser.add_argument(
"--config-version-id",
default="local-dev",
help="Optional config version identifier"
)
parser.add_argument(
"--track-debug",
action="store_true",
help="Print event trackId for protocol debugging"
)
args = parser.parse_args()
@@ -583,8 +638,12 @@ async def main():
client = MicrophoneClient(
url=args.url,
sample_rate=args.sample_rate,
app_id=args.app_id,
channel=args.channel,
config_version_id=args.config_version_id,
input_device=args.input_device,
output_device=args.output_device
output_device=args.output_device,
track_debug=args.track_debug,
)
client.verbose = args.verbose

View File

@@ -52,9 +52,21 @@ if not PYAUDIO_AVAILABLE and not SD_AVAILABLE:
class SimpleVoiceClient:
"""Simple voice client with reliable audio playback."""
def __init__(self, url: str, sample_rate: int = 16000):
def __init__(
self,
url: str,
sample_rate: int = 16000,
app_id: str = "assistant_demo",
channel: str = "simple_client",
config_version_id: str = "local-dev",
track_debug: bool = False,
):
self.url = url
self.sample_rate = sample_rate
self.app_id = app_id
self.channel = channel
self.config_version_id = config_version_id
self.track_debug = track_debug
self.ws = None
self.running = False
@@ -75,6 +87,17 @@ class SimpleVoiceClient:
# Interrupt handling - discard audio until next trackStart
self._discard_audio = False
@staticmethod
def _event_ids_suffix(event: dict) -> str:
data = event.get("data") if isinstance(event.get("data"), dict) else {}
keys = ("turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id")
parts = []
for key in keys:
value = data.get(key, event.get(key))
if value:
parts.append(f"{key}={value}")
return f" [{' '.join(parts)}]" if parts else ""
async def connect(self):
"""Connect to server."""
@@ -83,12 +106,25 @@ class SimpleVoiceClient:
self.running = True
print("Connected!")
# Send invite
# WS v1 handshake: hello -> session.start
await self.ws.send(json.dumps({
"command": "invite",
"option": {"codec": "pcm", "sampleRate": self.sample_rate}
"type": "hello",
"version": "v1",
}))
print("-> invite")
await self.ws.send(json.dumps({
"type": "session.start",
"audio": {
"encoding": "pcm_s16le",
"sample_rate_hz": self.sample_rate,
"channels": 1,
},
"metadata": {
"appId": self.app_id,
"channel": self.channel,
"configVersionId": self.config_version_id,
},
}))
print("-> hello/session.start")
async def send_chat(self, text: str):
"""Send chat message."""
@@ -96,8 +132,8 @@ class SimpleVoiceClient:
self.request_start_time = time.time()
self.first_audio_received = False
await self.ws.send(json.dumps({"command": "chat", "text": text}))
print(f"-> chat: {text}")
await self.ws.send(json.dumps({"type": "input.text", "text": text}))
print(f"-> input.text: {text}")
def play_audio(self, audio_data: bytes):
"""Play audio data immediately."""
@@ -152,34 +188,39 @@ class SimpleVoiceClient:
else:
# JSON event
event = json.loads(msg)
etype = event.get("event", "?")
etype = event.get("type", event.get("event", "?"))
ids = self._event_ids_suffix(event)
if self.track_debug:
print(f"[track-debug] event={etype} trackId={event.get('trackId')}{ids}")
if etype == "transcript":
if etype in {"transcript", "transcript.delta", "transcript.final"}:
# User speech transcription
text = event.get("text", "")
is_final = event.get("isFinal", False)
is_final = etype == "transcript.final" or bool(event.get("isFinal"))
if is_final:
print(f"<- You said: {text}")
print(f"<- You said: {text}{ids}")
else:
print(f"<- [listening] {text}", end="\r")
elif etype == "ttfb":
elif etype in {"ttfb", "metrics.ttfb"}:
# Server-side TTFB event
latency_ms = event.get("latencyMs", 0)
print(f"<- [TTFB] Server reported latency: {latency_ms}ms")
elif etype == "trackStart":
elif etype in {"trackStart", "output.audio.start"}:
# New track starting - accept audio again
self._discard_audio = False
print(f"<- {etype}")
elif etype == "interrupt":
print(f"<- {etype}{ids}")
elif etype in {"interrupt", "response.interrupted"}:
# Interrupt - discard audio until next trackStart
self._discard_audio = True
print(f"<- {etype} (discarding audio until new track)")
elif etype == "hangup":
print(f"<- {etype}")
print(f"<- {etype}{ids} (discarding audio until new track)")
elif etype in {"hangup", "session.stopped"}:
print(f"<- {etype}{ids}")
self.running = False
break
elif etype == "config.resolved":
print(f"<- config.resolved {event.get('config', {}).get('output', {})}{ids}")
else:
print(f"<- {etype}")
print(f"<- {etype}{ids}")
except asyncio.TimeoutError:
continue
@@ -270,6 +311,10 @@ async def main():
parser.add_argument("--text", help="Send text and play response")
parser.add_argument("--list-devices", action="store_true")
parser.add_argument("--sample-rate", type=int, default=16000)
parser.add_argument("--app-id", default="assistant_demo")
parser.add_argument("--channel", default="simple_client")
parser.add_argument("--config-version-id", default="local-dev")
parser.add_argument("--track-debug", action="store_true")
args = parser.parse_args()
@@ -277,7 +322,14 @@ async def main():
list_audio_devices()
return
client = SimpleVoiceClient(args.url, args.sample_rate)
client = SimpleVoiceClient(
args.url,
args.sample_rate,
app_id=args.app_id,
channel=args.channel,
config_version_id=args.config_version_id,
track_debug=args.track_debug,
)
await client.run(args.text)

View File

@@ -36,8 +36,18 @@ def generate_sine_wave(duration_ms=1000):
return audio_data
async def receive_loop(ws, ready_event: asyncio.Event):
async def receive_loop(ws, ready_event: asyncio.Event, track_debug: bool = False):
"""Listen for incoming messages from the server."""
def event_ids_suffix(data):
payload = data.get("data") if isinstance(data.get("data"), dict) else {}
keys = ("turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id")
parts = []
for key in keys:
value = payload.get(key, data.get(key))
if value:
parts.append(f"{key}={value}")
return f" [{' '.join(parts)}]" if parts else ""
print("👂 Listening for server responses...")
async for msg in ws:
timestamp = datetime.now().strftime("%H:%M:%S")
@@ -46,7 +56,10 @@ async def receive_loop(ws, ready_event: asyncio.Event):
try:
data = json.loads(msg.data)
event_type = data.get('type', 'Unknown')
print(f"[{timestamp}] 📨 Event: {event_type} | {msg.data[:150]}...")
ids = event_ids_suffix(data)
print(f"[{timestamp}] 📨 Event: {event_type}{ids} | {msg.data[:150]}...")
if track_debug:
print(f"[{timestamp}] [track-debug] event={event_type} trackId={data.get('trackId')}{ids}")
if event_type == "session.started":
ready_event.set()
except json.JSONDecodeError:
@@ -113,7 +126,7 @@ async def send_sine_loop(ws):
print("\n✅ Finished streaming test audio.")
async def run_client(url, file_path=None, use_sine=False):
async def run_client(url, file_path=None, use_sine=False, track_debug: bool = False):
"""Run the WebSocket test client."""
session = aiohttp.ClientSession()
try:
@@ -121,7 +134,7 @@ async def run_client(url, file_path=None, use_sine=False):
async with session.ws_connect(url) as ws:
print("✅ Connected!")
session_ready = asyncio.Event()
recv_task = asyncio.create_task(receive_loop(ws, session_ready))
recv_task = asyncio.create_task(receive_loop(ws, session_ready, track_debug=track_debug))
# Send v1 hello + session.start handshake
await ws.send_json({"type": "hello", "version": "v1"})
@@ -131,7 +144,12 @@ async def run_client(url, file_path=None, use_sine=False):
"encoding": "pcm_s16le",
"sample_rate_hz": SAMPLE_RATE,
"channels": 1
}
},
"metadata": {
"appId": "assistant_demo",
"channel": "test_websocket",
"configVersionId": "local-dev",
},
})
print("📤 Sent v1 hello/session.start")
await asyncio.wait_for(session_ready.wait(), timeout=8)
@@ -168,9 +186,10 @@ if __name__ == "__main__":
parser.add_argument("--url", default=SERVER_URL, help="WebSocket endpoint URL")
parser.add_argument("--file", help="Path to PCM/WAV file to stream")
parser.add_argument("--sine", action="store_true", help="Use sine wave generation (default)")
parser.add_argument("--track-debug", action="store_true", help="Print event trackId for protocol debugging")
args = parser.parse_args()
try:
asyncio.run(run_client(args.url, args.file, args.sine))
asyncio.run(run_client(args.url, args.file, args.sine, args.track_debug))
except KeyboardInterrupt:
print("\n👋 Client stopped.")

View File

@@ -57,10 +57,15 @@ class WavFileClient:
url: str,
input_file: str,
output_file: str,
app_id: str = "assistant_demo",
channel: str = "wav_client",
config_version_id: str = "local-dev",
sample_rate: int = 16000,
chunk_duration_ms: int = 20,
wait_time: float = 15.0,
verbose: bool = False
verbose: bool = False,
track_debug: bool = False,
tail_silence_ms: int = 800,
):
"""
Initialize WAV file client.
@@ -77,11 +82,17 @@ class WavFileClient:
self.url = url
self.input_file = Path(input_file)
self.output_file = Path(output_file)
self.app_id = app_id
self.channel = channel
self.config_version_id = config_version_id
self.sample_rate = sample_rate
self.chunk_duration_ms = chunk_duration_ms
self.chunk_samples = int(sample_rate * chunk_duration_ms / 1000)
self.wait_time = wait_time
self.verbose = verbose
self.track_debug = track_debug
self.tail_silence_ms = max(0, int(tail_silence_ms))
self.frame_bytes = 640 # 16k mono pcm_s16le, 20ms
# WebSocket connection
self.ws = None
@@ -125,6 +136,17 @@ class WavFileClient:
# Replace problematic characters for console output
safe_message = message.encode('ascii', errors='replace').decode('ascii')
print(f"{direction} {safe_message}")
@staticmethod
def _event_ids_suffix(event: dict) -> str:
data = event.get("data") if isinstance(event.get("data"), dict) else {}
keys = ("turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id")
parts = []
for key in keys:
value = data.get(key, event.get(key))
if value:
parts.append(f"{key}={value}")
return f" [{' '.join(parts)}]" if parts else ""
async def connect(self) -> None:
"""Connect to WebSocket server."""
@@ -144,7 +166,12 @@ class WavFileClient:
"encoding": "pcm_s16le",
"sample_rate_hz": self.sample_rate,
"channels": 1
}
},
"metadata": {
"appId": self.app_id,
"channel": self.channel,
"configVersionId": self.config_version_id,
},
})
async def send_command(self, cmd: dict) -> None:
@@ -216,6 +243,10 @@ class WavFileClient:
end_sample = min(sent_samples + chunk_size, total_samples)
chunk = audio_data[sent_samples:end_sample]
chunk_bytes = chunk.tobytes()
if len(chunk_bytes) % self.frame_bytes != 0:
# v1 audio framing requires 640-byte (20ms) PCM units.
pad = self.frame_bytes - (len(chunk_bytes) % self.frame_bytes)
chunk_bytes += b"\x00" * pad
# Send to server
if self.ws:
@@ -232,6 +263,16 @@ class WavFileClient:
# Delay to simulate real-time streaming
# Server expects audio at real-time pace for VAD/ASR to work properly
await asyncio.sleep(self.chunk_duration_ms / 1000)
# Add a short silence tail to help VAD/EOU close the final utterance.
if self.tail_silence_ms > 0 and self.ws:
tail_frames = max(1, self.tail_silence_ms // 20)
silence = b"\x00" * self.frame_bytes
for _ in range(tail_frames):
await self.ws.send(silence)
self.bytes_sent += len(silence)
await asyncio.sleep(0.02)
self.log_event("", f"Sent trailing silence: {self.tail_silence_ms}ms")
self.send_completed = True
elapsed = time.time() - self.send_start_time
@@ -284,16 +325,22 @@ class WavFileClient:
async def _handle_event(self, event: dict) -> None:
"""Handle incoming event."""
event_type = event.get("type", "unknown")
ids = self._event_ids_suffix(event)
if self.track_debug:
print(f"[track-debug] event={event_type} trackId={event.get('trackId')}{ids}")
if event_type == "hello.ack":
self.log_event("", "Handshake acknowledged")
self.log_event("", f"Handshake acknowledged{ids}")
elif event_type == "session.started":
self.session_ready = True
self.log_event("", "Session ready!")
self.log_event("", f"Session ready!{ids}")
elif event_type == "config.resolved":
config = event.get("config", {})
self.log_event("", f"Config resolved (output={config.get('output', {})}){ids}")
elif event_type == "input.speech_started":
self.log_event("", "Speech detected")
self.log_event("", f"Speech detected{ids}")
elif event_type == "input.speech_stopped":
self.log_event("", "Silence detected")
self.log_event("", f"Silence detected{ids}")
elif event_type == "transcript.delta":
text = event.get("text", "")
display_text = text[:60] + "..." if len(text) > 60 else text
@@ -301,35 +348,35 @@ class WavFileClient:
elif event_type == "transcript.final":
text = event.get("text", "")
print(" " * 80, end="\r")
self.log_event("", f"→ You: {text}")
self.log_event("", f"→ You: {text}{ids}")
elif event_type == "metrics.ttfb":
latency_ms = event.get("latencyMs", 0)
self.log_event("", f"[TTFB] Server latency: {latency_ms}ms")
elif event_type == "assistant.response.delta":
text = event.get("text", "")
if self.verbose and text:
self.log_event("", f"LLM: {text}")
self.log_event("", f"LLM: {text}{ids}")
elif event_type == "assistant.response.final":
text = event.get("text", "")
if text:
self.log_event("", f"LLM Response (final): {text[:100]}{'...' if len(text) > 100 else ''}")
self.log_event("", f"LLM Response (final): {text[:100]}{'...' if len(text) > 100 else ''}{ids}")
elif event_type == "output.audio.start":
self.track_started = True
self.response_start_time = time.time()
self.waiting_for_first_audio = True
self.log_event("", "Bot started speaking")
self.log_event("", f"Bot started speaking{ids}")
elif event_type == "output.audio.end":
self.track_ended = True
self.log_event("", "Bot finished speaking")
self.log_event("", f"Bot finished speaking{ids}")
elif event_type == "response.interrupted":
self.log_event("", "Bot interrupted!")
self.log_event("", f"Bot interrupted!{ids}")
elif event_type == "error":
self.log_event("!", f"Error: {event.get('message')}")
self.log_event("!", f"Error: {event.get('message')}{ids}")
elif event_type == "session.stopped":
self.log_event("", f"Session stopped: {event.get('reason')}")
self.log_event("", f"Session stopped: {event.get('reason')}{ids}")
self.running = False
else:
self.log_event("", f"Event: {event_type}")
self.log_event("", f"Event: {event_type}{ids}")
def save_output_wav(self) -> None:
"""Save received audio to output WAV file."""
@@ -473,6 +520,21 @@ async def main():
default=16000,
help="Target sample rate for audio (default: 16000)"
)
parser.add_argument(
"--app-id",
default="assistant_demo",
help="Stable app/assistant identifier for server-side config lookup"
)
parser.add_argument(
"--channel",
default="wav_client",
help="Client channel name"
)
parser.add_argument(
"--config-version-id",
default="local-dev",
help="Optional config version identifier"
)
parser.add_argument(
"--chunk-duration",
type=int,
@@ -490,6 +552,17 @@ async def main():
action="store_true",
help="Enable verbose output"
)
parser.add_argument(
"--track-debug",
action="store_true",
help="Print event trackId for protocol debugging"
)
parser.add_argument(
"--tail-silence-ms",
type=int,
default=800,
help="Trailing silence to send after WAV playback for EOU detection (default: 800)"
)
args = parser.parse_args()
@@ -497,10 +570,15 @@ async def main():
url=args.url,
input_file=args.input,
output_file=args.output,
app_id=args.app_id,
channel=args.channel,
config_version_id=args.config_version_id,
sample_rate=args.sample_rate,
chunk_duration_ms=args.chunk_duration,
wait_time=args.wait_time,
verbose=args.verbose
verbose=args.verbose,
track_debug=args.track_debug,
tail_silence_ms=args.tail_silence_ms,
)
await client.run()

View File

@@ -401,6 +401,9 @@
const targetSampleRate = 16000;
const playbackStopRampSec = 0.008;
const appId = "assistant_demo";
const channel = "web_client";
const configVersionId = "local-dev";
function logLine(type, text, data) {
const time = new Date().toLocaleTimeString();
@@ -604,15 +607,35 @@
logLine("sys", `${cmd.type}`, cmd);
}
function eventIdsSuffix(event) {
const data = event && typeof event.data === "object" && event.data ? event.data : {};
const keys = ["turn_id", "utterance_id", "response_id", "tool_call_id", "tts_id"];
const parts = [];
for (const key of keys) {
const value = data[key] || event[key];
if (value) parts.push(`${key}=${value}`);
}
return parts.length ? ` [${parts.join(" ")}]` : "";
}
function handleEvent(event) {
const type = event.type || "unknown";
logLine("event", type, event);
const ids = eventIdsSuffix(event);
logLine("event", `${type}${ids}`, event);
if (type === "hello.ack") {
sendCommand({
type: "session.start",
audio: { encoding: "pcm_s16le", sample_rate_hz: targetSampleRate, channels: 1 },
metadata: {
appId,
channel,
configVersionId,
},
});
}
if (type === "config.resolved") {
logLine("sys", "config.resolved", event.config || {});
}
if (type === "transcript.final") {
if (event.text) {
setInterim("You", "");