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Keep realtime conversation items ordered across turns

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This commit is contained in:
Mikei386
2026-09-16 14:42:30 +02:00
parent b9dcadedd5
commit 97c8072b3d
3 changed files with 82 additions and 13 deletions
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services/athena-realtime-voice/README.md
+4
View File
@@ -16,6 +16,10 @@ The bridge announces each committed user audio item before sending its completed
transcript with the same item ID. OpenClaw requires that sequence to persist the
transcript; omitting the item caused “Realtime transcript refers to an unknown
speech item” in the browser.
Spoken assistant replies are also announced as conversation items with matching
transcript IDs and explicit predecessor IDs. Without them, OpenClaw's ordered
transcript storage can wait on a missing assistant item when the next user turn
arrives, leaving later questions unanswered.
Required service environment:
services/athena-realtime-voice/server.py
+46 -6
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@@ -12,6 +12,7 @@ import hashlib
import hmac
import io
import json
import logging
import os
import time
import uuid
@@ -21,6 +22,7 @@ from fractions import Fraction
import numpy as np
from aiohttp import ClientSession, FormData, web
from aiortc import MediaStreamTrack, RTCPeerConnection, RTCSessionDescription
from aiortc.mediastreams import MediaStreamError
from av import AudioFrame, AudioResampler
from cryptography.exceptions import InvalidSignature
from cryptography.hazmat.primitives.serialization import load_pem_public_key
@@ -34,6 +36,12 @@ USED_TOKENS = web.AppKey("used_tokens", dict)
PEERS = web.AppKey("peers", set)
HTTP = web.AppKey("http", ClientSession)
PUBLIC_KEY_URL = web.AppKey("public_key_url", str)
logger = logging.getLogger("athena_realtime_voice")
logger.setLevel(logging.INFO)
logger.propagate = False
_handler = logging.StreamHandler()
_handler.setFormatter(logging.Formatter("%(asctime)s %(levelname)s %(message)s"))
logger.addHandler(_handler)
def decode_payload(payload: str, timestamp: int) -> dict:
@@ -170,8 +178,10 @@ class RealtimeSession:
self.speech = bytearray()
self.silence_since: float | None = None
self.call_id: str | None = None
self.last_item_id: str | None = None
self.pending_reply: str | None = None
self.turn_task: asyncio.Task | None = None
self.dropped_speech_while_busy = False
self.vad_threshold = float(os.environ.get("VAD_THRESHOLD", "0.018"))
self.silence_ms = int(os.environ.get("SILENCE_MS", "750"))
@@ -187,6 +197,11 @@ class RealtimeSession:
except json.JSONDecodeError:
return
kind = event.get("type")
if kind in {"conversation.item.create", "response.create", "response.cancel"}:
logger.info("client event=%s item_type=%s call_matches=%s busy=%s pending_reply=%s",
kind, (event.get("item") or {}).get("type"),
(event.get("item") or {}).get("call_id") == self.call_id,
self.busy, self.pending_reply is not None)
if kind == "session.update":
self.emit({"type": "session.updated", "session": {"id": "athena-local"}})
elif kind == "conversation.item.create":
@@ -218,12 +233,21 @@ class RealtimeSession:
await self.on_audio(pcm)
except asyncio.CancelledError:
raise
except MediaStreamError:
return
except Exception:
logger.exception("incoming audio stream stopped unexpectedly")
return
async def on_audio(self, pcm: bytes) -> None:
if self.busy or not pcm:
if not pcm:
return
if self.busy:
if not self.dropped_speech_while_busy and pcm_rms(pcm) >= self.vad_threshold:
self.dropped_speech_while_busy = True
logger.info("microphone speech ignored while response is busy")
return
self.dropped_speech_while_busy = False
voiced = pcm_rms(pcm) >= self.vad_threshold
now = time.monotonic()
if voiced and not self.speaking:
@@ -231,6 +255,7 @@ class RealtimeSession:
self.speech.clear()
self.silence_since = None
self.emit({"type": "input_audio_buffer.speech_started"})
logger.info("speech started")
if not self.speaking:
return
self.speech.extend(pcm)
@@ -242,6 +267,7 @@ class RealtimeSession:
if elapsed_silence >= self.silence_ms or len(self.speech) >= SAMPLE_RATE * 2 * 45:
self.speaking = False
self.emit({"type": "input_audio_buffer.speech_stopped"})
logger.info("speech stopped bytes=%s", len(self.speech))
audio = bytes(self.speech)
self.speech.clear()
if len(audio) >= SAMPLE_RATE * 2 // 4:
@@ -265,14 +291,18 @@ class RealtimeSession:
return
item_id = f"item_{uuid.uuid4().hex}"
# OpenClaw tracks transcript items before accepting their text.
self.emit({"type": "input_audio_buffer.committed", "item_id": item_id})
self.emit({"type": "input_audio_buffer.committed", "item_id": item_id,
"previous_item_id": self.last_item_id})
self.last_item_id = item_id
self.emit({"type": "conversation.item.input_audio_transcription.completed",
"item_id": item_id, "transcript": text})
logger.info("transcript completed item_id=%s", item_id)
await self.consult(text)
except asyncio.CancelledError:
self.busy = False
raise
except Exception:
logger.exception("speech transcription failed")
self.busy = False
self.emit({"type": "error", "error": {"message": "Athena STT failed"}})
@@ -289,13 +319,16 @@ class RealtimeSession:
"arguments": json.dumps({"prompt": text}, ensure_ascii=False),
}],
}})
logger.info("consult requested call_id=%s response_id=%s", self.call_id, response_id)
async def speak(self, text: str) -> None:
if not text.strip():
self.busy = False
return
response_id = f"resp_{uuid.uuid4().hex}"
item_id = f"item_{uuid.uuid4().hex}"
try:
logger.info("speaking response_id=%s", response_id)
self.emit({"type": "response.created", "response": {"id": response_id}})
async with self.http.post(
os.environ["ATHENA_API_BASE_URL"].rstrip("/") + "/audio/speech",
@@ -306,18 +339,25 @@ class RealtimeSession:
response.raise_for_status()
pcm = read_wav(await response.read())
self.track.enqueue(pcm)
self.emit({"type": "response.output_audio_transcript.done", "transcript": text})
item = {"id": item_id, "type": "message", "role": "assistant",
"status": "completed"}
self.emit({"type": "conversation.item.created", "item": item,
"previous_item_id": self.last_item_id})
self.last_item_id = item_id
self.emit({"type": "response.output_audio_transcript.done",
"item_id": item_id, "transcript": text})
# Keep half-duplex active until the queued audio has actually played.
await asyncio.sleep(len(pcm) / (SAMPLE_RATE * 2) + 0.2)
self.emit({"type": "conversation.item.done", "item": item})
self.emit({"type": "response.done", "response": {
"id": response_id, "status": "completed", "output": [
{"id": f"item_{uuid.uuid4().hex}", "type": "message", "role": "assistant", "status": "completed"}
],
"id": response_id, "status": "completed", "output": [item],
}})
logger.info("speech response complete response_id=%s", response_id)
except asyncio.CancelledError:
self.emit({"type": "response.cancelled", "response": {"id": response_id}})
raise
except Exception:
logger.exception("speech synthesis failed")
self.emit({"type": "error", "error": {"message": "Athena TTS failed"}})
finally:
self.busy = False
services/athena-realtime-voice/test_smoke.py
+31 -6
View File
@@ -31,8 +31,8 @@ class Microphone(MediaStreamTrack):
async def recv(self):
await asyncio.sleep(max(0, self.start + self.count * .02 - time.monotonic()))
# 0.6 s speech followed by silence to trigger server VAD.
if 4 <= self.count < 44:
# Two utterances separated by enough time for the first TTS reply.
if 4 <= self.count < 44 or 750 <= self.count < 790:
index = np.arange(FRAME_SAMPLES) + self.count * FRAME_SAMPLES
samples = (8000 * np.sin(2 * np.pi * 440 * index / SAMPLE_RATE)).astype("<i2")
else:
@@ -132,16 +132,20 @@ class VoiceSmokeTest(unittest.IsolatedAsyncioTestCase):
voice_runner, voice_url = await serve(create_app())
pc = RTCPeerConnection()
events = []
event_times = []
done = asyncio.Event()
audio_received = asyncio.Event()
completed_responses = 0
function_calls = 0
channel = pc.createDataChannel("oai-events")
pc.addTrack(Microphone())
microphone = Microphone()
pc.addTrack(microphone)
@pc.on("track")
def on_track(track):
if track.kind == "audio":
async def consume():
for _ in range(150):
for _ in range(600):
frame = await track.recv()
if np.max(np.abs(frame.to_ndarray())) > 100:
audio_received.set()
@@ -154,11 +158,14 @@ class VoiceSmokeTest(unittest.IsolatedAsyncioTestCase):
@channel.on("message")
def on_message(raw):
nonlocal completed_responses, function_calls
event = json.loads(raw)
events.append(event)
event_times.append((round(time.monotonic() - microphone.start, 1), event.get("type")))
if event.get("type") == "response.done":
output = event.get("response", {}).get("output", [])
if output and output[0].get("type") == "function_call":
function_calls += 1
call = output[0]
channel.send(json.dumps({"type": "conversation.item.create", "item": {
"type": "function_call_output", "call_id": call["call_id"],
@@ -166,6 +173,8 @@ class VoiceSmokeTest(unittest.IsolatedAsyncioTestCase):
}}))
channel.send(json.dumps({"type": "response.create"}))
elif output and output[0].get("type") == "message":
completed_responses += 1
if completed_responses == 2:
done.set()
try:
@@ -186,13 +195,29 @@ class VoiceSmokeTest(unittest.IsolatedAsyncioTestCase):
answer = await response.text()
await pc.setRemoteDescription(RTCSessionDescription(sdp=answer, type="answer"))
try:
await asyncio.wait_for(done.wait(), timeout=20)
await asyncio.wait_for(done.wait(), timeout=25)
except asyncio.TimeoutError:
self.fail(f"No completed voice response; events: {[e.get('type') for e in events]}")
self.fail(f"No completed voice response; state={pc.connectionState}, "
f"mic_frames={microphone.count}, events: {event_times}")
await asyncio.wait_for(audio_received.wait(), timeout=5)
self.assertEqual(function_calls, 2)
self.assertTrue(any(event.get("type") ==
"conversation.item.input_audio_transcription.completed"
for event in events))
user_items = [event for event in events
if event.get("type") == "input_audio_buffer.committed"]
assistant_items = [event for event in events
if event.get("type") == "conversation.item.created"
and event.get("item", {}).get("role") == "assistant"]
assistant_transcripts = [event for event in events
if event.get("type") == "response.output_audio_transcript.done"]
self.assertEqual(len(user_items), 2)
self.assertEqual(len(assistant_items), 2)
self.assertEqual(len(assistant_transcripts), 2)
self.assertEqual(assistant_items[0]["previous_item_id"], user_items[0]["item_id"])
self.assertEqual(assistant_transcripts[0]["item_id"], assistant_items[0]["item"]["id"])
self.assertEqual(user_items[1]["previous_item_id"], assistant_items[0]["item"]["id"])
self.assertEqual(assistant_items[1]["previous_item_id"], user_items[1]["item_id"])
committed = next((index, event) for index, event in enumerate(events)
if event.get("type") == "input_audio_buffer.committed")
completed = next((index, event) for index, event in enumerate(events)