Alibaba Qwen Releases Qwen-Audio-3.1-Realtime: A Full-Duplex Voice Model Trained to Think, Act, and Decide When to Speak
https://www.marktechpost.com/2026/09/28/alibaba-qwen-releases-qwen-audio-3-1-realtime-a-full-duplex-voice-model-trained-to-think-act-and-decide-when-to-speak/📌 Qwen-Audio-3.1:能自己決定何時開口的全雙工語音模型
TL;DR:阿里 Qwen 推出全雙工語音代理模型 Qwen-Audio-3.1-Realtime,能自主判斷聆聽、插話與沉默的時機,同時大砍 API 價格。
多數語音助理仍是「你說完、我才開始想」的半雙工邏輯,打斷、搶話、該閉嘴時卻硬要插嘴的尷尬並不少見。Qwen-Audio-3.1-Realtime 想解決的正是這個問題:讓模型自己判斷什麼時候該聽、該說、該停、該接續。
🤔 背景:一套 5 模型的音訊堆疊
Alibaba 的 Qwen 團隊發布 Qwen-Audio-3.1,涵蓋 ASR、TTS 與即時互動的 5 模型組合,主力是為會呼叫工具的語音代理設計的全雙工模型 Qwen-Audio-3.1-Realtime。同時大幅調降價格:Realtime 降約 85%、TTS 降約 70%、ASR 最高降 95%。目前僅提供受管理的 API 服務,qwen-audio-3.1-realtime-plus 透過 WebSocket 在 QwenCloud 上線,官方未釋出開源權重。
🧩 三個模型接力完成一次對話
系統由兩個共用 Audio Encoder 與 LLM 設計的模型組成:一個全雙工決策模型持續判斷「該繼續聽、該開口、該停止,還是該接續」;一個語音轉文字模型把回應內容寫成文字;接著一個具備上下文感知能力的語音渲染器,把文字轉成串流語音,同時參考對話歷史、聲音線索與聲學情境。
模型規格上,文字與語音都可作為輸入與輸出,context 視窗達 26.2 萬 tokens(最多 24.5 萬輸入、1.6 萬輸出),預設限制為每分鐘 60 次請求、10 萬 tokens。功能涵蓋 function calling、網頁搜尋、結構化輸出、context cache 與 fine-tuning。另一款搭配模型 Qwen-Audio-3.1-ASR-Flash-Filetrans 則專攻離線長音訊轉錄,支援熱詞、語者分離、標點符號,以及多語言與中文方言辨識。
📊 訓練分三層:Think、Act、Speak and Coordinate
官方將訓練拆成三層。第一層 Core-Cocktail SFT,用百萬小時等級的配對資料,把 audio 模型重新錨定回其文字 LLM 的能力基礎。第二層是 Multimodality OPD(on-policy distillation):由一個 Text Teacher 與一個凍結的 Audio Reference,針對學生模型自己產生的軌跡逐 token 評分,而不是模仿預先寫好的標準答案。第三層則針對同理心、語用意圖、聲學場景訓練出各自的 domain expert,以 GRPO 訓練,再透過 Multi-Teacher OPD 合併成單一可部署模型。
每個訓練領域都綁定一組工具池、一個具狀態的 JSON 資料庫,以及一份自然語言業務政策,領域本身則是從開源工具與 MCP 伺服器定義中取樣而來。每個任務只有三種結局:完成寫入、合理拒絕,或判定請求無法支援。評分邏輯先檢查最終狀態,再檢查是否有權限進行的寫入操作,最後才看行為層面的判斷,也就是說回應說得再流暢,只要狀態檢查沒過關也拿不到分;GRPO 的獎勵則分別在對話、里程碑、輪次三個層級給出。
在搜尋訓練上,官方設計了一個懲罰重複查詢的獎勵機制:當模型發出的查詢數量超過參考查詢數量時,獎勵會按比例被打折。結果是平均每次搜尋呼叫的查詢數從 4.37 降到 1.05,但觸發搜尋的 F1 分數也從 60.87% 微幅降到 58.61%。
📊 全雙工判斷的進步與代價
這一層決定模型何時、是否、如何開口。在 Full-Duplex-Bench v1.5 上,模型誤把「別人在跟第三方講話」當成該回應的比例,從 0.13 降到 0.03;v3.0 上的填充語(filler)出現率從 0.7590 降到 0.2960。但也有取捨:被打斷之後,模型「不該恢復卻恢復說話」的比例從 0.035 上升到 0.130,打斷後的停止延遲是 1.116 秒,相較 GPT-Realtime-2 的 0.383 秒明顯較慢。
其他評測上,對比 3.0 版本,Audio MultiChallenge 分數從 47.12 提升到 52.21,14 種語言的 BBA 平均分從 81.7% 升到 88.1%,FLEURS 的詞錯誤率從 9.01 降到 3.98(官方註明 τ-Voice 的數字是用半雙工的語音轉文字方式量測,不能直接與正式全雙工結果比較)。在 50 場人類 red-team 對話測試中,GPT-Realtime-2 仍以 96.00% 對 92.00% 領先。
💡 定價
Qwen-Audio-3.1-Realtime 的價格為每百萬 audio input tokens 6.4 美元、每百萬 text input tokens 0.8 美元,輸出(文字加語音)每百萬 tokens 24 美元,純文字輸出不收費。官方也提醒,不同供應商對音訊的 tokenize 方式不同,跨廠牌價格不能直接類比。Qwen-Audio-3.1-ASR-Flash-Filetrans 的價格則是輸入 0.15 美元、輸出 0.47 美元(每百萬 tokens)。
⚠️ 限制
打斷後「不當恢復說話」的比例上升,加上停止延遲慢於 GPT-Realtime-2,顯示全雙工判斷的擬真度與反應速度目前仍有妥協;人類 red-team 評測也顯示整體表現尚未超越 GPT-Realtime-2。此外目前沒有開源權重,只能透過 QwenCloud 的受管理 API 使用。
🎯 實務啟示
如果你正在打造會呼叫工具的語音代理(客服、語音下單、多輪任務型助理),Qwen-Audio-3.1-Realtime 把「何時該說話」做成獨立的決策層,而不是單純的靜音偵測,有助於降低搶話與尷尬沉默。正式導入前,建議針對你的實際打斷場景測試延遲與誤判率,並留意目前只能透過 API 使用、無法自行部署的限制。
🔗 來源
- 標題:Alibaba Qwen Releases Qwen-Audio-3.1-Realtime: A Full-Duplex Voice Model Trained to Think, Act, and Decide When to Speak
- 作者/機構:Asif Razzaq,MarkTechPost
- 連結:https://www.marktechpost.com/2026/09/28/alibaba-qwen-releases-qwen-audio-3-1-realtime-a-full-duplex-voice-model-trained-to-think-act-and-decide-when-to-speak/
#Qwen #Alibaba #VoiceAI #FullDuplex #SpeechAI #AIAgents #TTS #ASR #GRPO #ConversationalAI
原始資料 MarkTechPost · 收集於 2026-09-29
摘要原文
Alibaba’s Qwen team has released Qwen-Audio-3.1 , a 5-model audio stack spanning ASR, TTS and realtime interaction. The main model is Qwen-Audio-3.1-Realtime , a full-duplex speech model built for voice agents that call tools. Qwen also cut prices : about 85% on Realtime, about 70% on TTS and up to 95% on ASR. Is it deployable? Yes, as a managed API. qwen-audio-3.1-realtime-plus is live on QwenCloud over WebSocket. No open weights were announced. The model page lists text and audio as both input and output. Context is 262K tokens, with 245K max input and 16K max output. Default limits are 60 requests and 100K tokens per minute. Pricing is $6.4 per 1M audio input tokens and $0.8 per 1M text input tokens. Text and audio output costs $24 per 1M tokens, with output text not charged. Key features include function calling, web search, structured outputs, context cache and fine-tuning. A companion model, Qwen-Audio-3.1-ASR-Flash-Filetrans , targets offline long-audio transcription. It supports hot words, speaker separation, punctuation and multilingual plus Chinese dialect recognition. It costs $0.15 input and $0.47 output per 1M tokens. The system runs 2 models with the same Audio Encoder and LLM design. A full-duplex decision model predicts whether to keep listening, speak, stop or resume. A speech-to-text model writes the response content as text. A context-aware voice renderer then turns that text into streaming speech. It conditions on conversation history, voice cues and acoustic context. Training is organized into 3 layers: Think, Act, and Speak and Coordinate. Core-Cocktail SFT re-anchors the audio model to its source text LLM using million-hour-scale paired data. Multimodality OPD follows. A Text Teacher and a frozen Audio Reference score each token of the student’s own trajectory. This is on-policy distillation , not imitation of pre-written answers. Domain experts for empathy, pragmatic intent and acoustic scenes are then trained with GRPO. Multi-Teacher OPD merges them into 1 deployable model. Each training domain bundles a tool pool, a stateful JSON database and a natural-language business policy. Domains are seeded from open-source tool and MCP server definitions. Every task defines 1 of 3 outcomes: a write, a justified refusal, or an unsupported request. Scoring checks terminal state, then permitted writes, then behavioral assertions. A fluent reply cannot rescue a failed state check. GRPO receives rewards at dialogue, milestone and turn level. Search training penalizes redundant queries with r query = q min ( 1 , n ref n pred ) r_{\text{query}} = q \min \left( 1, \frac{n_{\text{ref}}}{n_{\text{pred}}} \right) . Mean queries per search call fell from 4.37 to 1.05. Trigger F1 slipped from 60.87% to 58.61%. This layer decides whether, when and how to speak. On Full-Duplex-Bench v1.5, replies to people talking to someone else fell from 0.13 to 0.03. On v3.0, the filler rate dropped from 0.7590 to 0.2960. There are trade-offs. After interruptions, the unwanted resume rate rose from 0.035 to 0.130. Interruption stop latency is 1.116 seconds, versus 0.383 for GPT-Realtime-2. Explore the Think, Act, Speak loop, duplex decisions, a scored training episode and the search reward. Against 3.0, Audio MultiChallenge rises from 47.12 to 52.21. The 14-language BBA average climbs from 81.7% to 88.1%. FLEURS WER falls from 9.01 to 3.98. The τ-Voice figures use a half-duplex speech-to-text adaptation. They are not comparable to official full-duplex results. GPT-Realtime-2 still leads the 50-session human red-team study, 96.00% versus 92.00%. Prices are list rates checked September 28, 2026. Token rates are not directly comparable, since each provider tokenizes audio differently. Check out the Paper , Qwen-Audio-3.1-ASR , and Qwen-Audio-3.1-Realtime . All credit goes to the researcher of this project. Also, feel free to follow us on Twitter and don’t forget to join our 150k+ML SubReddit and Subscribe to our Newsletter . Wait! are you on telegram? now you can join us on telegram as well. Need to partner with us for promoting your GitHub Repo OR Hugging Face Page OR Product Release OR Webinar etc.? Connect with us The post Alibaba Qwen Releases Qwen-Audio-3.1-Realtime: A Full-Duplex Voice Model Trained to Think, Act, and Decide When to Speak appeared first on MarkTechPost .
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