qwen-audio-3.0-realtime-plus
即時語音對話模型,接受語音與文字,並以語音或文字回應。設計重點是自然的交談節奏,以及與工具互動。
關於模型
此說明介紹模型的能力。Ailoot 可用的輸入、輸出與選項,依所選方法列於下方。
Qwen-Audio-3.0-Realtime-Plus 是 Alibaba 的即時語音對話模型。它接受語音與文字,以音訊與文字回應,並支援自然交換話語的雙向對話。聲音可分段傳送與接收,讓互動隨對話進行。
判斷對方是否說完
可透過聲音活動、語意完整度或明確控制,判斷一次發話的邊界。在模型開發商提供的模式中,須在傳送第一段音訊前選定模式;開始對話後若要變更,必須建立新的連線。這種控制是否可用,取決於選定的方法。語意模式會考量停頓與填充詞:短暫猶豫不一定表示使用者已說完。使用者的新發話可中斷正在形成的回應,讓對話依下一段發話繼續。語意模式下,可用指定對話者的語音樣本引導辨識,以減少其他人聲及背景雜音的干擾。
指令設定助理角色及交談風格。語音表達可考量語調、速度與情緒;3.0 版本提供五種系統聲音,以及依語音樣本設定的聲音。模型支援 11 種語言,包括俄語,也支援模型開發商列出的漢語方言變體。在模型開發商的設定方式中,聲音於第一次更新工作階段設定時指定;同一段對話之後傳入的新聲音會被忽略。這些設定須由選定的呼叫方式支援。
工具與對話歷史
支援函式呼叫。模型提出所需動作,連接的應用程式執行動作,再將結果傳回對話。這讓對話能圍繞實際資料展開,並在操作完成後繼續。語音助理可利用工具結果,同時保留對話脈絡。模型開發商的功能也包括內建網路搜尋,前提是選定的方法有提供。網路搜尋與函式呼叫不能在同一工作階段同時啟用。
預設歷史保留 20 輪對話,可設定的範圍為一至 50 輪。累積音訊的上限為 300 秒;超過限制時,較舊的歷史片段會被刪除。這與單次發話的長度不同:歷史設定決定繼續對話時,哪些先前素材仍可使用。
文字脈絡與聲音
正式脈絡長度為 40 960 個 Token。輸入長度上限為 16 384 個 Token,輸出上限為 8192;推理模式也採用相同的輸入與輸出上限,另外限制推理鏈最多為 1024 個 Token。
串流聲音使用單聲道、16 位元 PCM:輸入為 16 kHz,輸出為 24 kHz。文字與音訊資料屬於同一段對話。此特定模型未宣稱支援視覺輸入或輸出圖片;專長是語音、文字及語音互動。
Ailoot · 即時語音 GET /v1/realtime?model=qwen-audio-3.0-realtime-plus qwen-audio-3.0-realtime-plus
qwen-audio-3.0-realtime-plus入門
即時語音對話模型,接受語音與文字,並以語音或文字回應。設計重點是自然的交談節奏,以及與工具互動。
- 接受的輸入
- 文字 · 音訊
- 結果
- 文字 · 音訊
Ailoot 可用 · 輸入與方法
Ailoot · 即時語音 GET /v1/realtime?model=qwen-audio-3.0-realtime-plus
- 輸入
text,audio- 輸出
text,audio- 輸出格式
- 工作階段事件
| 欄位 | 類型 | 要求 | 值與限制 | 預設值 |
|---|---|---|---|---|
欄位query.model | 類型string | 要求必填 | 值與限制 qwen-audio-3.0-realtime-plus | 預設值— |
欄位session.input_audio_format | 類型string | 要求選填 | 值與限制 pcm16, pcm | 預設值— |
欄位session.output_audio_format | 類型string | 要求選填 | 值與限制 pcm16, pcm | 預設值— |
欄位session.instructions | 類型string | 要求選填 | 值與限制 限制適用對象: 所選的工作階段政策 | 預設值— |
欄位session.max_history_turns | 類型integer | 要求選填 | 值與限制 最小值: 1 限制適用對象: 所選的工作階段政策 | 預設值— |
欄位input_audio_buffer.append.audio | 類型base64_string | 要求必填 | 值與限制 輸入內容: PCM16 單聲道;輸入 16 kHz,輸出 24 kHz;位元組長度須非零且為偶數 | 預設值— |
輸入素材組合
realtime_text_audio
方法參數與生命週期
| 參數 | 規格 |
|---|---|
參數material_types | 規格text, audio |
參數required | 規格null |
參數optional_controls | 規格null |
參數content_type | 規格WebSocket |
生命週期: exact retained published method
方法結構與 ID 均為精確值;供應商能力與准入狀態分開。
方案
Ailoot 方法費率
GET /v1/realtime?model=qwen-audio-3.0-realtime-plus- 輸入
- US$0.0008 / 1K
- 輸出
- US$0.0064 / 1K
- 輸入
- US$0.0048 每分鐘音訊
- 輸出
- US$0.018 每分鐘輸出音訊
Ailoot 會在請求前預留估算金額。取得結果後,依實測用量結算最終費用。串流結算採用最終用量紀錄。預留金額與結算的運作方式 →
請求範例
Ailoot · 即時語音
# Install the documented client once: python3 -m pip install websockets
import base64, json, os, time, wave
from pathlib import Path
from websockets.sync.client import connect
key = os.environ.get("AILOOT_API_KEY", "")
if not key.strip():
raise SystemExit("Set your AILOOT_API_KEY before connecting")
file = Path(os.environ.get("AILOOT_TEXT_FILE", ""))
if not file.is_file(): raise SystemExit("Set AILOOT_TEXT_FILE to your actual UTF-8 material")
material = file.read_text(encoding="utf-8")
if not material.strip(): raise SystemExit("Text material must not be empty")
out = Path("realtime-results")
out.mkdir(exist_ok=True)
audio, texts, complete = {}, {}, []
current_response = None
pending = False
deadline = time.monotonic() + 570 # Local client bound; lower server/session limits still apply.
def receive(ws, log):
global current_response, pending
left = deadline - time.monotonic()
if left <= 0:
raise TimeoutError("Client session limit; preserve IDs and partial results, do not replay")
event = json.loads(ws.recv(timeout=min(60, left)))
log.write(json.dumps(event, ensure_ascii=False) + "\n")
log.flush()
kind = event.get("type")
if kind == "error":
raise RuntimeError(json.dumps(event, ensure_ascii=False))
if kind == "response.created":
current_response = event["response"]["id"]
if kind in {"response.audio.delta", "response.output_audio.delta", "response.text.delta", "response.output_text.delta", "response.audio_transcript.delta", "response.output_audio_transcript.delta", "response.text.done", "response.audio_transcript.done"}:
identity = (event["response_id"], event["item_id"], event["output_index"], event["content_index"])
if kind in {"response.audio.delta", "response.output_audio.delta"}:
raw = base64.b64decode(event["delta"], validate=True)
if len(raw) % 2: raise ValueError("Output PCM16 chunk must have an even byte length")
audio.setdefault(identity, bytearray()).extend(raw)
elif kind.endswith(".done"):
texts[identity] = event.get("text", event.get("transcript", ""))
else:
texts[identity] = texts.get(identity, "") + event["delta"]
print(event["delta"], end="", flush=True)
if kind == "response.done":
response = event["response"]
complete.append(response)
if current_response is None: current_response = response["id"]
if response["id"] == current_response: pending = False
return event
def wait_for(ws, log, kind):
while True:
event = receive(ws, log)
if event.get("type") == kind: return event
def send(ws, event):
ws.send(json.dumps(event, ensure_ascii=False))
try:
# One exact model query; no reconnect/retry loop and no provider credentials.
with connect("wss://api.ailoot.ai/v1/realtime?model=qwen-audio-3.0-realtime-plus", additional_headers={"Authorization": "Bearer " + key}) as ws:
with (out / "events.jsonl").open("w", encoding="utf-8") as log:
wait_for(ws, log, "session.created")
send(ws, {"type":"session.update", "session":{"modalities":["audio","text"], "input_audio_format":"pcm", "output_audio_format":"pcm", "turn_detection":None}})
wait_for(ws, log, "session.updated")
send(ws, {"type":"conversation.item.create", "item":{"type":"message", "role":"user", "content":[{"type":"input_text", "text":material}]}})
send(ws, {"type":"response.create"})
pending = True
try:
while pending: receive(ws, log)
except KeyboardInterrupt:
# Cancel the actual current inference, then preserve its returned status.
send(ws, {"type":"response.cancel"})
while pending: receive(ws, log)
if not complete or complete[-1].get("status") != "completed":
raise RuntimeError("Response did not complete normally; inspect saved events and status")
finally:
manifest = {"responses":complete, "parts":[]}
for number, identity in enumerate(dict.fromkeys(list(audio) + list(texts))):
record = {"response_id":identity[0], "item_id":identity[1], "output_index":identity[2], "content_index":identity[3]}
if identity in audio:
raw = bytes(audio[identity])
pcm_file = out / ("audio-" + str(number) + ".pcm")
wav_file = out / ("audio-" + str(number) + ".wav")
pcm_file.write_bytes(raw)
with wave.open(str(wav_file), "wb") as wav:
wav.setnchannels(1); wav.setsampwidth(2); wav.setframerate(24000); wav.writeframes(raw)
record.update({"pcm":str(pcm_file), "wav":str(wav_file), "sample_rate_hz":24000})
if identity in texts:
text_file = out / ("text-" + str(number) + ".txt")
text_file.write_text(texts[identity], encoding="utf-8")
record["text"] = str(text_file)
manifest["parts"].append(record)
(out / "results.json").write_text(json.dumps(manifest, ensure_ascii=False, indent=2), encoding="utf-8")
print("\nSaved every received event and result in", out)
# Install the documented client once: python3 -m pip install websockets
import base64, json, os, time, wave
from pathlib import Path
from websockets.sync.client import connect
key = os.environ.get("AILOOT_API_KEY", "")
if not key.strip():
raise SystemExit("Set your AILOOT_API_KEY before connecting")
file = Path(os.environ.get("AILOOT_AUDIO_FILE", ""))
if not file.is_file(): raise SystemExit("Set AILOOT_AUDIO_FILE to your raw PCM16 mono 16-kHz recording")
material = file.read_bytes()
if not material or len(material) % 2 or material.startswith(b"RIFF"):
raise SystemExit("Use nonempty signed little-endian PCM16 mono 16-kHz bytes, not a WAV header")
out = Path("realtime-results")
out.mkdir(exist_ok=True)
audio, texts, complete = {}, {}, []
current_response = None
pending = False
deadline = time.monotonic() + 570 # Local client bound; lower server/session limits still apply.
def receive(ws, log):
global current_response, pending
left = deadline - time.monotonic()
if left <= 0:
raise TimeoutError("Client session limit; preserve IDs and partial results, do not replay")
event = json.loads(ws.recv(timeout=min(60, left)))
log.write(json.dumps(event, ensure_ascii=False) + "\n")
log.flush()
kind = event.get("type")
if kind == "error":
raise RuntimeError(json.dumps(event, ensure_ascii=False))
if kind == "response.created":
current_response = event["response"]["id"]
if kind in {"response.audio.delta", "response.output_audio.delta", "response.text.delta", "response.output_text.delta", "response.audio_transcript.delta", "response.output_audio_transcript.delta", "response.text.done", "response.audio_transcript.done"}:
identity = (event["response_id"], event["item_id"], event["output_index"], event["content_index"])
if kind in {"response.audio.delta", "response.output_audio.delta"}:
raw = base64.b64decode(event["delta"], validate=True)
if len(raw) % 2: raise ValueError("Output PCM16 chunk must have an even byte length")
audio.setdefault(identity, bytearray()).extend(raw)
elif kind.endswith(".done"):
texts[identity] = event.get("text", event.get("transcript", ""))
else:
texts[identity] = texts.get(identity, "") + event["delta"]
print(event["delta"], end="", flush=True)
if kind == "response.done":
response = event["response"]
complete.append(response)
if current_response is None: current_response = response["id"]
if response["id"] == current_response: pending = False
return event
def wait_for(ws, log, kind):
while True:
event = receive(ws, log)
if event.get("type") == kind: return event
def send(ws, event):
ws.send(json.dumps(event, ensure_ascii=False))
try:
# One exact model query; no reconnect/retry loop and no provider credentials.
with connect("wss://api.ailoot.ai/v1/realtime?model=qwen-audio-3.0-realtime-plus", additional_headers={"Authorization": "Bearer " + key}) as ws:
with (out / "events.jsonl").open("w", encoding="utf-8") as log:
wait_for(ws, log, "session.created")
send(ws, {"type":"session.update", "session":{"modalities":["audio","text"], "input_audio_format":"pcm", "output_audio_format":"pcm", "turn_detection":None}})
wait_for(ws, log, "session.updated")
for offset in range(0, len(material), 640):
send(ws, {"type":"input_audio_buffer.append", "audio":base64.b64encode(material[offset:offset+640]).decode("ascii")})
time.sleep(0.02)
send(ws, {"type":"input_audio_buffer.commit"})
wait_for(ws, log, "input_audio_buffer.committed")
send(ws, {"type":"response.create"})
pending = True
try:
while pending: receive(ws, log)
except KeyboardInterrupt:
# Cancel the actual current inference, then preserve its returned status.
send(ws, {"type":"response.cancel"})
while pending: receive(ws, log)
if not complete or complete[-1].get("status") != "completed":
raise RuntimeError("Response did not complete normally; inspect saved events and status")
finally:
manifest = {"responses":complete, "parts":[]}
for number, identity in enumerate(dict.fromkeys(list(audio) + list(texts))):
record = {"response_id":identity[0], "item_id":identity[1], "output_index":identity[2], "content_index":identity[3]}
if identity in audio:
raw = bytes(audio[identity])
pcm_file = out / ("audio-" + str(number) + ".pcm")
wav_file = out / ("audio-" + str(number) + ".wav")
pcm_file.write_bytes(raw)
with wave.open(str(wav_file), "wb") as wav:
wav.setnchannels(1); wav.setsampwidth(2); wav.setframerate(24000); wav.writeframes(raw)
record.update({"pcm":str(pcm_file), "wav":str(wav_file), "sample_rate_hz":24000})
if identity in texts:
text_file = out / ("text-" + str(number) + ".txt")
text_file.write_text(texts[identity], encoding="utf-8")
record["text"] = str(text_file)
manifest["parts"].append(record)
(out / "results.json").write_text(json.dumps(manifest, ensure_ascii=False, indent=2), encoding="utf-8")
print("\nSaved every received event and result in", out)
接收並儲存結果
Ailoot · 即時語音
工作階段事件
保留所有已完成的結果
請保留所有完整傳回的回應、檔案與內容區塊。
請依每個實際回應 ID,組合所有回傳的音訊增量片段與文字/事件結果。
需要檢查原始回應時
請保留實際的工作階段/回應/事件 ID 與紀錄憑證。
錯誤與後續步驟
| 回應 | 條件 | 下一步 |
|---|---|---|
回應426 · invalid_request_error | 條件未進行 WebSocket 升級 | 下一步請使用 WebSocket 用戶端,並帶入 Upgrade: websocket。 |
別名
模型別名
Ailoot qwen-audio-3.0-realtime-plus
方法路徑別名
請使用上方列出的 API 方法路徑。
相關呼叫用 ID
FAQ · qwen-audio-3.0-realtime-plus
- 可以使用哪些輸入與結果?
GET /v1/realtime?model=qwen-audio-3.0-realtime-plus·qwen-audio-3.0-realtime-plus:text,audio→text,audio- 適用哪些限制?
GET /v1/realtime?model=qwen-audio-3.0-realtime-plus·qwen-audio-3.0-realtime-plus:session.max_history_turns.minimum:1