feat(voice): 阿里百炼 STT/TTS 主链路接通(base64 STT + cosyvoice TTS按voice_id + REST /audio/* + voice_input.deviceId)

- voice/dashscope.go 新建：DashscopeSTT(qwen-audio-3.0-asr-flash, base64 data URI→text) + DashscopeTTS(cosyvoice 按 voice_id, 取非流式 24h URL 下载字节)
- cloud.go newDashboardSTT/TTS 由空 endpoint stub 改为真实百炼 adapter
- chain.go gating 从 DashscopeAPIKey!="" 改为 voiceConfigured(cfg); silent 兜底保留
- config.VoiceConfig 扩展 DASHSCOPE_BASE_URL/VOICE_STT_*/VOICE_TTS_* 字段(必填 DASHSCOPE_API_KEY)
- gateway: POST /api/v1/audio/{transcribe,speech} (owner+审计+MaxBytesReader); voiceInputRequest 增 deviceId 并透传 runner.Run
- 实测: cosyvoice-v3.5-plus+longxiaochun 回 418(引擎拒绝), 改用 cosyvoice-v3-flash+longxiaochun_v3; TTS 你好我是昔涟->106124B wav -> STT 回 你好我是西联, 链返回真实字节/文本
- 单测: dashscope_test/config voice defaults/gateway audio_test; go build/vet/test 全绿
- 声音复刻(响昔涟音色)待公网参考音频; 不做 create_voice

.env.example

@@ -66,7 +66,20 @@ IOT_PROVIDER=mock
 # ---- 语音服务 ----
 TTS_PROVIDER=edge-tts
 STT_PROVIDER=dashscope
-# DASHSCOPE_API_KEY=sk-xxxxx
+# DASHSCOPE_API_KEY=sk-xxxxx   # ⚠️ 必填才能接通阿里百炼语音；未填时走降级链 (silent)。绝不打日志。
+# --- 阿里百炼 DashScope 语音（STT + TTS 主链路）---
+# DASHSCOPE_BASE_URL=https://dashscope.aliyuncs.com/api/v1
+# VOICE_STT_MODEL=qwen-audio-3.0-asr-flash
+# VOICE_STT_LANGUAGE=zh
+# 已实测：cosyvoice-v3.5-plus / cosyvoice-v2 走 HTTP SpeechSynthesizer 会回 418(引擎拒绝)；
+# 可用组合为 cosyvoice-v3-flash + 对应 v3 音色(longxiaochun_v3 龙小淳·女助手等)。
+# 各模型音色不可混用（v2 名如 longxiaochun 用于 v3 模型会 418）。
+# VOICE_TTS_MODEL=cosyvoice-v3-flash
+# VOICE_TTS_VOICE_ID=longxiaochun_v3
+# VOICE_TTS_FORMAT=wav
+# VOICE_TTS_SAMPLE_RATE=24000
+# 声音复刻（响昔涟音色）：先用参考音频（公网 URL）在百炼「声音复刻」建自定义音色，
+# 再把返回的 voice_id 填到 VOICE_TTS_VOICE_ID。TODO：待公网参考音频；本项目不做 create_voice。
 
 # ---- 后台思考 ----
 ENABLE_BACKGROUND_THINKING=true

Docs/api-client.md

@@ -29,6 +29,12 @@
 | 6 | `/api/v1/sessions/{id}/messages` | GET | JWT(owner/admin) | ✅ | **热分页拉历史**：`?limit=&before=&platform=` 默认 `limit=50`（上限 200）、`before`=上一页最旧 `createdAt`（毫秒），按时间倒序 `{data:[{id,sessionId,role,content,mode?,mood?,image?,thinking?,platform?,deviceId?,senderName?,createdAt}]}`；`platform` 为**可选**来源平台过滤（design/16 §4.4）——Web 端不传（=全量共享主会话），IM 适配器传**本平台**（如 `telegram`）只拉本渠道消息；`deviceId` 为**客户端具体端标识**（design/16 §4.5，如 `web:chrome:a1b2c3` / `weixin-ilink:conf0:wx_uid`）；`senderName` 为**第三方发送者昵称**（design/16 §4.6，群聊/陌生人私聊消息带，主人私聊为空，`omitempty`）；非法 `limit`/`before` 回 `400`；未注入 HistoryStore 回 `501` |
 | 7 | `/api/v1/sessions/{id}` | DELETE | JWT(owner/admin) | ✅ | **删除次级会话及其消息** → `{data:{deleted:<id>}}`；主会话不可删回 `403 forbidden`；未注入 HistoryStore 回 `501` |
 | 8 | `/api/v1/personas` | GET | JWT | ✅ | `{data:[persona...]}` |
+| 9 | `/api/v1/audio/transcribe` | POST | JWT(owner) | ✅ | **语音→文本**：body `{audio:"<base64>",format?,deviceId?}` → `{data:{text}}`（走后端 STT，百炼 `qwen-audio-3.0-asr-flash`） |
+| 10 | `/api/v1/audio/speech` | POST | JWT(owner) | ✅ | **文本→音频**：body `{text,voice?,format?}` → `200` 音频字节（`Content-Type: audio/<format>`，非 JSON envelope，走后端 TTS，百炼 `cosyvoice-v3-flash`） |
+
+> **语音 REST 源（`/api/v1/audio/*`，均 owner-only + 审计）**：原生/移动端用它上传/获取音频，**绕开 WS 4KB 帧限**。
+> - `POST /api/v1/audio/transcribe`：body `{audio:"<base64>", format?:"wav", deviceId?}` → `200 {data:{text}}`。`audio` 为 base64（或 data: URI）；`format` 为编码（`wav`/`mp3`/`ogg`，缺省 `wav`）；空 audio 回 `400`；未注入语音服务 `500 voice_unavailable`、识别失败 `500 stt_failed`。
+> - `POST /api/v1/audio/speech`：body `{text, voice?, format?}` → `200` 原始音频字节（`Content-Type: audio/<format>`），**非 JSON envelope**。`voice`/`format` 目前遵循服务端 `VOICE_TTS_VOICE_ID`/`VOICE_TTS_FORMAT` 配置（后续支持按请求覆盖）；空 text 回 `400`；未注入语音服务 `500 voice_unavailable`、合成失败 `500 tts_failed`、静默空音频 `500 tts_empty`。**声音复刻（响昔涟音色）**：先用参考音频（公网 URL）在百炼「声音复刻」建自定义音色，再把返回的 `voice_id` 配到 `VOICE_TTS_VOICE_ID`（本项目不做 create_voice）。
 
 ## 2. REST 端点（管理，`/api/v1/admin/*`）
 
@@ -126,7 +132,7 @@
 - **向后兼容**：未映射/无法确定的事件仍走 `stream_chunk` + `stream_end`（非流式回退路径）。
 - **上行（客户端→服务端）**: `mode_switch` / `chat`（跑 loop 并广播流式） / `voice_input` / `subagent_ask` / `task_output_query` / `interrupt`。
 - **`interrupt`**（上行，客户端→服务端）：`{sessionId}` → 会话编排器中断（design/15 §3）——取消该会话当前在跑的任务（context cancel）、清空其待处理队列、回 `session_interrupted`；用户说「停下/算了/换个话题」时发。命中空闲会话为 no-op。
-- **`voice_input`**（上行，客户端→服务端）：`_{audio?, format?, text?, sessionId?}`。`audio` 为 base64（或 data: URI）音频，`format` 为其编码（`wav`/`mp3`/`ogg`）；`text` 为已转写文本时跳过 STT。服务端走 STT→agent 聊天→TTS（design/06 §1），回复经 `voice_output` 定向回该连接。**真 STT/TTS 需配云端 endpoint**；无凭据时后端走降级链（cloud 端点为占位，edge-tts/espeak 未实现，最终 `silent` 返回空音频）。无注入语音服务时回 `error` 帧（`voice unavailable`）。
+- **`voice_input`**（上行，客户端→服务端）：`_{audio?, format?, text?, sessionId?, deviceId?}`。`audio` 为 base64（或 data: URI）音频，`format` 为其编码（`wav`/`mp3`/`ogg`）；`text` 为已转写文本时跳过 STT；`deviceId` 为**客户端具体端标识**（design/16 §4.5，如 `voiceassist:myapp:v1`，可选），随该语音轮次落库、聊天气泡可标注来源。服务端走 STT→agent 聊天→TTS（design/06 §1），回复经 `voice_output` 定向回该连接。**真 STT/TTS 已接通阿里百炼**（`DASHSCOPE_API_KEY` 必填：STT=`qwen-audio-3.0-asr-flash`、TTS=`cosyvoice-v3-flash`+`longxiaochun_v3`）；无凭据时后端走降级链（cloud → edge-tts/espeak（未实现）→ `silent` 空音频）。无注入语音服务时回 `error` 帧（`voice unavailable`）。
 - **`voice_output`**（下行，服务端→客户端）：`{audio, format, text, reply}`。`audio` 为 base64 合成音频，`format` 为其编码，`text` 为识别文本，`reply` 为助手回复文本。
 - **向后兼容**：未映射/无法确定的事件仍走 `stream_chunk` + `stream_end`（非流式回退路径）。
 
@@ -196,6 +202,7 @@
 - 语音：上行 `voice_input`（base64 音频或已转写 text），下行 `voice_output`（合成音频 base64 + text）。
 
 ## 5. 变更记录
+- `v0.18`（**后端语音接通阿里百炼 STT + TTS 主链路**）：`POST /api/v1/audio/transcribe`（**新增**，owner + 审计，body `{audio,format?,deviceId?}` → `{data:{text}}`，走 STT）；`POST /api/v1/audio/speech`（**新增**，owner + 审计，body `{text,voice?,format?}` → 音频字节 `audio/<format>`，走 TTS）——**REST 语音源**，原生/移动端经它绕开 WS 4KB 帧限上传/获取音频；WS `voice_input` 上行新增 **`deviceId`**（随语音轮次落库，design/16 §4.5）。后端 `internal/runtime/multimodal/voice` 的 cloud rung 由空 endpoint stub 改为**真实百炼 adapter**：STT `qwen-audio-3.0-asr-flash`（base64 原始字节→文本）、TTS `cosyvoice-v3-flash`（按 `voice_id` 合成，取非流式返回的 24h 音频 URL 下载字节）。**实测**（`DASHSCOPE_API_KEY` 实跑）：TTS「你好，我是昔涟」→ 106124 字节 wav；把该 wav 喂给 STT → 文本「你好，我是西联。」（TTS 发音把「昔涟」读得近似「西联」，STT 识别正确）；语音链 (cloud→…→silent) 返回真实字节/文本。**已实测修正**：`cosyvoice-v3.5-plus`/`longxiaochun`（v2 音色）走 HTTP `SpeechSynthesizer` 会回 `418`（引擎拒绝），改用 **`cosyvoice-v3-flash` + `longxiaochun_v3`**（v3 音色，各模型音色不可混用）。**声音复刻（响昔涟音色）仍待**：需参考音频（公网 URL）在百炼建自定义音色后把 `voice_id` 填入 `VOICE_TTS_VOICE_ID`（本项目不做 create_voice）。config 新增 `DASHSCOPE_BASE_URL`/`VOICE_STT_\*`/`VOICE_TTS_\*`（`VOICE_TTS_SAMPLE_RATE` 默认 24000）。
 - `v0.17`（系统设置安全与存储：吊销全部 API key / 轮换 JWT 密钥 / 备份 / 恢复，design/11 §4 + design/10 §3 模块14）：新增 `POST /api/v1/admin/system/{revoke-keys,rotate-jwt,backup,restore}`（均 **owner-only** + 审计）。`revoke-keys` 走 `auth.APIKey.RevokeAll()`（清空 key 表，返回 `{revoked:n}`）；`rotate-jwt` 走 `auth.JWTService.RotateSecret`（≥32 位，**新密钥不回显/不落日志**，现有 token 立即失效）；`backup` 用 SQLite `VACUUM INTO` 快照到 `<dataDir>/backup-<ts>.db`（含已提交 WAL 内容）返回 `{path,size}`；`restore` 先校验路径在 dataDir 内（**`filepath.Rel` 防路径穿越**，逃逸回 `400`）→ **先自动备份当前 DB** → 替换 DB 文件 → 返回 `{restored:true,backup,note:"需重启后生效"}`（运行进程持有 DB 句柄，热替换需重启可见）。gateway 新增 `Deps.DatabasePath`（composition root 传入实际打开的 SQLite 路径）。`编辑环境变量 / 重启服务 / 检查更新` 仍为后置，未实现、不伪造。
 - `v0.17`（KOOK 适配器面板 + 每适配器专属页，design/17 §2）：`GET /api/v1/admin/channels/{platform}/view`、`GET /api/v1/admin/channels/{platform}/friends?type=`、`POST /api/v1/admin/channels/{platform}/friend/handle`、`POST /api/v1/admin/channels/{platform}/message/test`（**新增**，均 owner-only + 审计，带可选 `?accountId=`，仅 `kook` 实现，其它平台 `501`）；`GET /api/v1/admin/channels` 每条账号附 `capabilities`（面板按平台渲染专属卡片）。KOOK 面板：好友/申请、服务器/频道/成员、私信会话、发消息测试——经适配器 REST（`Authorization: Bot <token>`）拉真实数据，token 不打印不落日志，子项失败记入 `errors` 不伪造。
 - `v0.16`（运行时配置体系：渠道适配器配置落盘 + 热更新，design/04 §3.1）：`PUT /api/v1/admin/channels/{platform}/config`（**新增**，owner-only + 审计）body `{id?,token?,baseURL?,remoteURL?,listenAddr?,mode?,accessToken?,selfID?,config?,allowGroups?,denyGroups?,allowUsers?,denyUsers?,groupMsgPerMin?,msgPerMin?,maxMessages?}` → `201 {persisted:true,path,platform,accountId}`——读当前有效配置 → `UpsertChannel` → 落盘 `channels.json`（本地、gitignore、0600）→ `bridge.Reload(platform)` **热替换运行中 adapter**（进行中消息用旧、新消息用新），无需重启；token 写入文件但**不回显/不落日志**。`DELETE /api/v1/admin/channels/{platform}/config?id=`（**新增**）删除账号并热更新。`GET /api/v1/admin/channels` 每条渠道附 `accounts`（每平台已配账号，token 脱敏掩码）。`.env`（`CHANNELS_JSON`/`CHANNEL_*_TOKEN`）退为兜底。

internal/config/config.go

@@ -97,11 +97,35 @@ type IoTConfig struct {
 	Provider string `env:"IOT_PROVIDER" default:"mock"`
 }
 
-// VoiceConfig holds TTS/STT provider settings.
+// VoiceConfig holds TTS/STT provider settings and the Alibaba Bailian DashScope
+// credentials/models that back the cloud rung of the voice chain.
 type VoiceConfig struct {
-	TTSProvider     string `env:"TTS_PROVIDER" default:"edge-tts"`
-	STTProvider     string `env:"STT_PROVIDER" default:"dashscope"`
+	// TTSProvider selects the non-cloud local TTS rung: "edge-tts" or "espeak".
+	TTSProvider string `env:"TTS_PROVIDER" default:"edge-tts"`
+	// STTProvider selects the cloud STT provider ("dashscope").
+	STTProvider string `env:"STT_PROVIDER" default:"dashscope"`
+	// DashscopeAPIKey is the Bailian API key backing STT + TTS. When empty the
+	// dashscope adapters return ErrNotConfigured and the chain degrades.
 	DashscopeAPIKey string `env:"DASHSCOPE_API_KEY"`
+
+	// DashscopeBaseURL is the Bailian API base; the STT/TTS service paths are
+	// appended to it.
+	DashscopeBaseURL string `env:"DASHSCOPE_BASE_URL" default:"https://dashscope.aliyuncs.com/api/v1"`
+
+	// STTModel is the DashScope speech-transcription model.
+	STTModel    string `env:"VOICE_STT_MODEL" default:"qwen-audio-3.0-asr-flash"`
+	STTLanguage string `env:"VOICE_STT_LANGUAGE" default:"zh"`
+
+	// TTSModel is the DashScope speech-synthesis model. cosyvoice-v3.5-plus and
+	// cosyvoice-v2 reject the HTTP SpeechSynthesizer call (engine 418); verified
+	// working is cosyvoice-v3-flash.
+	TTSModel string `env:"VOICE_TTS_MODEL" default:"cosyvoice-v3-flash"`
+	// TTSVoiceID selects the synthesized timbre. Voices are model-scoped (a v3
+	// model needs a _v3 voice). After cloning a custom voice in Bailian
+	// (音色复刻) fill its voice_id here.
+	TTSVoiceID    string `env:"VOICE_TTS_VOICE_ID" default:"longxiaochun_v3"`
+	TTSFormat     string `env:"VOICE_TTS_FORMAT" default:"wav"`
+	TTSSampleRate int    `env:"VOICE_TTS_SAMPLE_RATE" default:"24000"`
 }
 
 // BackgroundConfig holds the autonomous thinking settings. QuietStartHour and

internal/config/config_test.go

@@ -162,6 +162,53 @@ func TestLoad_CapacityEnvOverrides(t *testing.T) {
 	}
 }
 
+func TestLoad_VoiceDefaults(t *testing.T) {
+	t.Setenv("JWT_SECRET", strings.Repeat("s", 40))
+
+	cfg, err := Load("")
+	if err != nil {
+		t.Fatalf("Load() error: %v", err)
+	}
+	if cfg.Voice.DashscopeBaseURL != "https://dashscope.aliyuncs.com/api/v1" {
+		t.Errorf("DashscopeBaseURL = %q, want the default Bailian URL", cfg.Voice.DashscopeBaseURL)
+	}
+	if cfg.Voice.STTModel != "qwen-audio-3.0-asr-flash" {
+		t.Errorf("STTModel = %q, want qwen-audio-3.0-asr-flash", cfg.Voice.STTModel)
+	}
+	if cfg.Voice.STTLanguage != "zh" {
+		t.Errorf("STTLanguage = %q, want zh", cfg.Voice.STTLanguage)
+	}
+	if cfg.Voice.TTSModel != "cosyvoice-v3-flash" {
+		t.Errorf("TTSModel = %q, want cosyvoice-v3-flash", cfg.Voice.TTSModel)
+	}
+	if cfg.Voice.TTSVoiceID != "longxiaochun_v3" {
+		t.Errorf("TTSVoiceID = %q, want longxiaochun_v3", cfg.Voice.TTSVoiceID)
+	}
+	if cfg.Voice.TTSFormat != "wav" {
+		t.Errorf("TTSFormat = %q, want wav", cfg.Voice.TTSFormat)
+	}
+	if cfg.Voice.TTSSampleRate != 24000 {
+		t.Errorf("TTSSampleRate = %d, want 24000", cfg.Voice.TTSSampleRate)
+	}
+}
+
+func TestLoad_VoiceEnvOverrides(t *testing.T) {
+	t.Setenv("JWT_SECRET", strings.Repeat("s", 40))
+	t.Setenv("VOICE_TTS_VOICE_ID", "custom-voice-1")
+	t.Setenv("VOICE_STT_MODEL", "qwen-audio-3.0-asr-flash")
+
+	cfg, err := Load("")
+	if err != nil {
+		t.Fatalf("Load() error: %v", err)
+	}
+	if cfg.Voice.TTSVoiceID != "custom-voice-1" {
+		t.Errorf("TTSVoiceID = %q, want custom-voice-1", cfg.Voice.TTSVoiceID)
+	}
+	if cfg.Voice.STTModel != "qwen-audio-3.0-asr-flash" {
+		t.Errorf("STTModel = %q, want qwen-audio-3.0-asr-flash", cfg.Voice.STTModel)
+	}
+}
+
 func TestParseDotEnv_QuotedAndExport(t *testing.T) {
 	path := writeEnvFile(t, `export FOO=bar
 BAZ='quoted value'

internal/runtime/multimodal/voice/chain.go

@@ -80,8 +80,8 @@ func WithProviders(steps ...Provider) Option {
 // New assembles a chain from cfg using the fixed degradation order
 // cloud → local → edge-tts/espeak → silent.
 //
-//   - cloud is included only when a cloud voice setting is present (today an
-//     API key) so a mis-configured build degrades quietly.
+//   - cloud is included only when a cloud voice credential or model is present
+//     (voiceConfigured) so a mis-configured build degrades quietly.
 //   - local is a pure adaptation point and is only included when supplied via
 //     WithProviders; by default it is skipped.
 //   - the edge-tts or espeak rung is included according to cfg.TTSProvider.
@@ -98,7 +98,7 @@ func New(cfg config.VoiceConfig, opts ...Option) *Chain {
 	}
 
 	steps := make([]Provider, 0, 5)
-	if cfg.DashscopeAPIKey != "" {
+	if voiceConfigured(cfg) {
 		steps = append(steps, CloudProvider(newDashboardSTT(cfg), newDashboardTTS(cfg)))
 	}
 	switch cfg.TTSProvider {
@@ -110,3 +110,12 @@ func New(cfg config.VoiceConfig, opts ...Option) *Chain {
 	steps = append(steps, SilentProvider())
 	return NewChain(steps...)
 }
+
+// voiceConfigured reports whether any cloud voice credential or model is set, so
+// the cloud rung is only added when the runtime intends to use DashScope. Any
+// single present field (API key, base URL, or an STT/TTS model) makes the cloud
+// rung worth trying; a missing key then makes the adapter return ErrNotConfigured
+// and the chain degrades quietly.
+func voiceConfigured(cfg config.VoiceConfig) bool {
+	return cfg.DashscopeAPIKey != "" || cfg.DashscopeBaseURL != "" || cfg.STTModel != "" || cfg.TTSModel != ""
+}

internal/runtime/multimodal/voice/cloud.go

@@ -201,16 +201,15 @@ func discardLogger() *slog.Logger {
 	return slog.New(slog.NewTextHandler(io.Discard, nil))
 }
 
-// newDashboardSTT returns the cloud STT for a config. The endpoint is an
-// adaptation point and is empty until config gains a voice URL; a call then
-// returns ErrNotConfigured and the chain degrades.
+// newDashboardSTT returns the cloud STT adapter for a DashScope config. A call
+// returns ErrNotConfigured when no API key is set so the chain degrades.
 func newDashboardSTT(cfg config.VoiceConfig) STT {
-	return NewHTTPSTT("", cfg.DashscopeAPIKey)
+	return NewDashscopeSTT(cfg)
 }
 
-// newDashboardTTS returns the cloud TTS for a config.
+// newDashboardTTS returns the cloud TTS adapter for a DashScope config.
 func newDashboardTTS(cfg config.VoiceConfig) TTS {
-	return NewHTTPTTS("", cfg.DashscopeAPIKey)
+	return NewDashscopeTTS(cfg)
 }
 
 // newEdgeTTS returns the edge-tts adapter point (not yet implemented).

internal/runtime/multimodal/voice/dashscope.go

@@ -0,0 +1,397 @@
+package voice
+
+import (
+	"bytes"
+	"context"
+	"encoding/base64"
+	"encoding/json"
+	"fmt"
+	"io"
+	"net/http"
+	"strings"
+
+	"git.yeij.top/AskaEth/Cyrene_v2/internal/config"
+)
+
+// DashScope (Alibaba Bailian) STT/TTS adapters for the cloud rung of the voice
+// chain. Requests follow the DashScope public API:
+//
+//   - STT:   POST {base}/services/aigc/multimodal-generation/generation
+//            (qwen-audio-3.0-asr-flash, base64 input audio)
+//   - TTS:   POST {base}/services/audio/tts/SpeechSynthesizer
+//            (cosyvoice-v3-flash, synthesized by voice_id)
+//
+// Both take a non-streaming response here: STT returns the transcript in the
+// output envelope; TTS (no X-DashScope-SSE header) returns a 24h audio URL that
+// we download so the caller always gets the actual bytes. Audio format ↔ mime
+// is mapped in both directions (mimeOf / the data: URI in the STT request).
+
+const (
+	// defaultSTTSampleRate is the input audio sample rate the ASR expects.
+	defaultSTTSampleRate = 16000
+	// dashscopeSTTPath is the multimodal-generation service path (ASR).
+	dashscopeSTTPath = "/services/aigc/multimodal-generation/generation"
+	// dashscopeTTSPath is the SpeechSynthesizer service path (TTS).
+	dashscopeTTSPath = "/services/audio/tts/SpeechSynthesizer"
+	// defaultDashscopeBaseURL is the Bailian API root when none is configured.
+	defaultDashscopeBaseURL = "https://dashscope.aliyuncs.com/api/v1"
+)
+
+// DashscopeSTT transcribes audio via DashScope multimodal-generation.
+type DashscopeSTT struct {
+	endpoint   string
+	apiKey     string
+	model      string
+	format     string
+	language   string
+	sampleRate int
+	client     *http.Client
+}
+
+// NewDashscopeSTT builds the DashScope STT adapter from cfg. Missing model /
+// language / base URL fall back to the DashScope defaults.
+func NewDashscopeSTT(cfg config.VoiceConfig) *DashscopeSTT {
+	model := firstNonEmpty(cfg.STTModel, "qwen-audio-3.0-asr-flash")
+	language := firstNonEmpty(cfg.STTLanguage, "zh")
+	return &DashscopeSTT{
+		endpoint:   dashscopeEndpoint(cfg.DashscopeBaseURL) + dashscopeSTTPath,
+		apiKey:     cfg.DashscopeAPIKey,
+		model:      model,
+		format:     "wav",
+		language:   language,
+		sampleRate: defaultSTTSampleRate,
+		client:     defaultHTTPClient(),
+	}
+}
+
+// STT posts the audio as a base64 data: URI and returns the recognized text.
+func (s *DashscopeSTT) STT(ctx context.Context, audio Audio) (string, error) {
+	if s.apiKey == "" {
+		return "", ErrNotConfigured
+	}
+	if len(audio.Data) == 0 {
+		return "", fmt.Errorf("voice: %w: empty audio", ErrProviderFailed)
+	}
+	if err := ctx.Err(); err != nil {
+		return "", err
+	}
+	format := audio.Format
+	if format == "" {
+		format = "wav"
+	}
+	payload, err := json.Marshal(dashscopeSTTRequest{
+		Model: s.model,
+		Input: dashscopeSTTInput{
+			Messages: []dashscopeSTTMessage{{
+				Role: "user",
+				Content: []dashscopeSTTContent{{
+					Type: "input_audio",
+					InputAudio: dashscopeInputAudio{
+						Data: "data:" + mimeOf(format) + ";base64," + base64.StdEncoding.EncodeToString(audio.Data),
+					},
+				}},
+			}},
+		},
+		Parameters: dashscopeSTTParameters{
+			Format:        format,
+			SampleRate:    s.sampleRate,
+			LanguageHints: []string{s.language},
+		},
+	})
+	if err != nil {
+		return "", fmt.Errorf("voice: %w: marshal: %v", ErrProviderFailed, err)
+	}
+	body, err := dashscopePost(ctx, s.client, s.apiKey, s.endpoint, payload, "disable")
+	if err != nil {
+		return "", err
+	}
+	text, err := extractSTTText(body)
+	if err != nil {
+		return "", fmt.Errorf("voice: %w: decode: %v", ErrProviderFailed, err)
+	}
+	return text, nil
+}
+
+// DashscopeTTS synthesizes speech video via DashScope SpeechSynthesizer.
+type DashscopeTTS struct {
+	endpoint   string
+	apiKey     string
+	model      string
+	voiceID    string
+	format     string
+	sampleRate int
+	client     *http.Client
+}
+
+// NewDashscopeTTS builds the DashScope TTS adapter from cfg. Missing model /
+// voice / format / base URL fall back to the DashScope defaults.
+func NewDashscopeTTS(cfg config.VoiceConfig) *DashscopeTTS {
+	model := firstNonEmpty(cfg.TTSModel, "cosyvoice-v3-flash")
+	voiceID := firstNonEmpty(cfg.TTSVoiceID, "longxiaochun_v3")
+	format := firstNonEmpty(cfg.TTSFormat, "wav")
+	sampleRate := cfg.TTSSampleRate
+	if sampleRate == 0 {
+		sampleRate = 24000
+	}
+	return &DashscopeTTS{
+		endpoint:   dashscopeEndpoint(cfg.DashscopeBaseURL) + dashscopeTTSPath,
+		apiKey:     cfg.DashscopeAPIKey,
+		model:      model,
+		voiceID:    voiceID,
+		format:     format,
+		sampleRate: sampleRate,
+		client:     defaultHTTPClient(),
+	}
+}
+
+// TTS synthesizes audio for text. The non-streaming DashScope response carries a
+// 24h audio URL (or a base64 blob); we resolve it to bytes and return them so the
+// caller holds a complete audio file.
+func (t *DashscopeTTS) TTS(ctx context.Context, text string) (Audio, error) {
+	if t.apiKey == "" {
+		return Audio{}, ErrNotConfigured
+	}
+	if strings.TrimSpace(text) == "" {
+		return Audio{}, ErrEmptyText
+	}
+	if err := ctx.Err(); err != nil {
+		return Audio{}, err
+	}
+	payload, err := json.Marshal(dashscopeTTSRequest{
+		Model: t.model,
+		Input: dashscopeTTSInput{
+			Text:       text,
+			Voice:      t.voiceID,
+			Format:     t.format,
+			SampleRate: t.sampleRate,
+		},
+	})
+	if err != nil {
+		return Audio{}, fmt.Errorf("voice: %w: marshal: %v", ErrProviderFailed, err)
+	}
+	// No X-DashScope-SSE header → non-streaming URL response.
+	body, err := dashscopePost(ctx, t.client, t.apiKey, t.endpoint, payload, "")
+	if err != nil {
+		return Audio{}, err
+	}
+	data, err := t.audioBytes(ctx, body)
+	if err != nil {
+		return Audio{}, err
+	}
+	return Audio{Format: t.format, Data: data}, nil
+}
+
+// audioBytes resolves the synthesized audio from the DashScope response: a 24h
+// URL is downloaded, or an inline base64 blob is decoded.
+func (t *DashscopeTTS) audioBytes(ctx context.Context, body []byte) ([]byte, error) {
+	var r dashscopeTTSResponse
+	if err := json.Unmarshal(body, &r); err != nil {
+		return nil, fmt.Errorf("voice: %w: decode: %v", ErrProviderFailed, err)
+	}
+	if url := r.Output.Audio.URL; url != "" {
+		data, err := t.download(ctx, url)
+		if err != nil {
+			return nil, err
+		}
+		return data, nil
+	}
+	if b64 := r.Output.Audio.Data; b64 != "" {
+		data, err := base64.StdEncoding.DecodeString(b64)
+		if err != nil {
+			return nil, fmt.Errorf("voice: %w: decode audio: %v", ErrProviderFailed, err)
+		}
+		return data, nil
+	}
+	return nil, fmt.Errorf("voice: %w: no audio in response", ErrProviderFailed)
+}
+
+// download fetches an audio URL into memory, capping the size so an oversized
+// media response cannot exhaust memory.
+func (t *DashscopeTTS) download(ctx context.Context, url string) ([]byte, error) {
+	req, err := http.NewRequestWithContext(ctx, http.MethodGet, url, nil)
+	if err != nil {
+		return nil, fmt.Errorf("voice: %w: build: %v", ErrProviderFailed, err)
+	}
+	resp, err := t.client.Do(req)
+	if err != nil {
+		return nil, fmt.Errorf("voice: %w: download: %v", ErrProviderFailed, err)
+	}
+	defer func() { _ = resp.Body.Close() }()
+	if resp.StatusCode != http.StatusOK {
+		return nil, fmt.Errorf("voice: %w: download status %d", ErrProviderFailed, resp.StatusCode)
+	}
+	data, err := io.ReadAll(io.LimitReader(resp.Body, cloudMaxResponseBytes))
+	if err != nil {
+		return nil, fmt.Errorf("voice: %w: download read: %v", ErrProviderFailed, err)
+	}
+	return data, nil
+}
+
+// dashscopeEndpoint returns the DashScope API base with a trailing slash trimmed,
+// defaulting to the public Bailian root when none is configured.
+func dashscopeEndpoint(baseURL string) string {
+	base := strings.TrimRight(baseURL, "/")
+	if base == "" {
+		return defaultDashscopeBaseURL
+	}
+	return base
+}
+
+// firstNonEmpty returns a when non-empty, else b.
+func firstNonEmpty(a, b string) string {
+	if a != "" {
+		return a
+	}
+	return b
+}
+
+// dashscopePost POSTs a JSON payload with the DashScope bearer auth and an
+// optional X-DashScope-SSE header, returning the decoded response body. A non-200
+// status is wrapped with the DashScope error body.
+func dashscopePost(ctx context.Context, client *http.Client, apiKey, url string, payload []byte, sse string) ([]byte, error) {
+	req, err := http.NewRequestWithContext(ctx, http.MethodPost, url, bytes.NewReader(payload))
+	if err != nil {
+		return nil, fmt.Errorf("voice: %w: build: %v", ErrProviderFailed, err)
+	}
+	req.Header.Set("Content-Type", "application/json")
+	if apiKey != "" {
+		req.Header.Set("Authorization", "Bearer "+apiKey)
+	}
+	if sse != "" {
+		req.Header.Set("X-DashScope-SSE", sse)
+	}
+	resp, err := client.Do(req)
+	if err != nil {
+		return nil, fmt.Errorf("voice: %w: request: %v", ErrProviderFailed, err)
+	}
+	defer func() { _ = resp.Body.Close() }()
+	body, err := io.ReadAll(io.LimitReader(resp.Body, cloudMaxResponseBytes))
+	if err != nil {
+		return nil, fmt.Errorf("voice: %w: read: %v", ErrProviderFailed, err)
+	}
+	if resp.StatusCode != http.StatusOK {
+		return nil, fmt.Errorf("voice: %w: dashscope status %d: %s", ErrProviderFailed, resp.StatusCode, strings.TrimSpace(string(body)))
+	}
+	return body, nil
+}
+
+// extractSTTText pulls the transcript out of a DashScope ASR response. The ASR
+// output shape varies by model, so several candidate paths are tried.
+func extractSTTText(body []byte) (string, error) {
+	var r dashscopeSTTResponse
+	if err := json.Unmarshal(body, &r); err != nil {
+		return "", err
+	}
+	if s := r.Output.Output.Sentence.Text; s != "" {
+		return s, nil
+	}
+	if s := r.Output.Output.Text; s != "" {
+		return s, nil
+	}
+	if s := r.Output.Text; s != "" {
+		return s, nil
+	}
+	for _, ch := range r.Output.Choices {
+		if len(ch.Message.Content) > 0 {
+			if s := ch.Message.Content[0].Text; s != "" {
+				return s, nil
+			}
+		}
+	}
+	return r.Text, nil
+}
+
+// mimeOf maps an audio format to its MIME type for the data: URI / Content-Type.
+func mimeOf(format string) string {
+	switch strings.ToLower(format) {
+	case "mp3", "mpeg":
+		return "audio/mpeg"
+	case "ogg", "oga":
+		return "audio/ogg"
+	case "webm":
+		return "audio/webm"
+	default:
+		return "audio/wav"
+	}
+}
+
+// dashscopeSTTResponse is the DashScope ASR response envelope.
+type dashscopeSTTResponse struct {
+	Output struct {
+		Output struct {
+			Sentence struct {
+				Text string `json:"text"`
+			} `json:"sentence"`
+			Text string `json:"text"`
+		} `json:"output"`
+		Text    string `json:"text"`
+		Choices []struct {
+			Message struct {
+				Content []struct {
+					Text string `json:"text"`
+				} `json:"content"`
+			} `json:"message"`
+		} `json:"choices"`
+	} `json:"output"`
+	Text string `json:"text"`
+}
+
+// dashscopeTTSResponse is the DashScope TTS response envelope.
+type dashscopeTTSResponse struct {
+	Output struct {
+		Audio struct {
+			URL  string `json:"url"`
+			Data string `json:"data"`
+		} `json:"audio"`
+	} `json:"output"`
+}
+
+// dashscopeSTTRequest is the ASR request body.
+type dashscopeSTTRequest struct {
+	Model      string                 `json:"model"`
+	Input      dashscopeSTTInput      `json:"input"`
+	Parameters dashscopeSTTParameters `json:"parameters"`
+}
+
+// dashscopeSTTInput carries the multimodal messages list.
+type dashscopeSTTInput struct {
+	Messages []dashscopeSTTMessage `json:"messages"`
+}
+
+// dashscopeSTTMessage is one ASR user turn.
+type dashscopeSTTMessage struct {
+	Role    string                `json:"role"`
+	Content []dashscopeSTTContent `json:"content"`
+}
+
+// dashscopeSTTContent is one audio content block.
+type dashscopeSTTContent struct {
+	Type       string              `json:"type"`
+	InputAudio dashscopeInputAudio `json:"input_audio"`
+}
+
+// dashscopeInputAudio carries the base64 data: URI audio.
+type dashscopeInputAudio struct {
+	Data string `json:"data"`
+}
+
+// dashscopeSTTParameters tunes the ASR request.
+type dashscopeSTTParameters struct {
+	Format        string   `json:"format"`
+	SampleRate    int      `json:"sample_rate"`
+	LanguageHints []string `json:"language_hints,omitempty"`
+}
+
+// dashscopeTTSRequest is the TTS request body.
+type dashscopeTTSRequest struct {
+	Model string            `json:"model"`
+	Input dashscopeTTSInput `json:"input"`
+}
+
+// dashscopeTTSInput carries text + voice + output format.
+type dashscopeTTSInput struct {
+	Text       string `json:"text"`
+	Voice      string `json:"voice"`
+	Format     string `json:"format"`
+	SampleRate int    `json:"sample_rate"`
+}

internal/runtime/multimodal/voice/dashscope_live_test.go

@@ -0,0 +1,96 @@
+//go:build dashscope_live
+
+package voice
+
+import (
+	"bytes"
+	"context"
+	"encoding/binary"
+	"os"
+	"testing"
+
+	"git.yeij.top/AskaEth/Cyrene_v2/internal/config"
+)
+
+// TestDashscopeLive verifies the real Alibaba Bailian STT + TTS end to end. It is
+// excluded from the default suite (build tag dashscope_live) so `go test ./...`
+// never hits the paid API. Run it explicitly with the key sourced from .env:
+//
+//	set -a; source .env; set +a
+//	go test -tags dashscope_live -run TestDashscopeLive -v ./internal/runtime/multimodal/voice/
+func TestDashscopeLive(t *testing.T) {
+	key := os.Getenv("DASHSCOPE_API_KEY")
+	if key == "" {
+		t.Skip("DASHSCOPE_API_KEY not set")
+	}
+	cfg := config.VoiceConfig{DashscopeAPIKey: key}
+
+	// TTS: synthesize speech for a short line.
+	tts := NewDashscopeTTS(cfg)
+	audio, err := tts.TTS(context.Background(), "你好，我是昔涟")
+	if err != nil {
+		t.Fatalf("TTS error: %v", err)
+	}
+	t.Logf("TTS: format=%q bytes=%d", audio.Format, len(audio.Data))
+	if len(audio.Data) == 0 {
+		t.Fatal("TTS returned empty audio")
+	}
+
+	// STT closed loop: transcribe the synthesized speech back to text.
+	stt := NewDashscopeSTT(cfg)
+	text, err := stt.STT(context.Background(), Audio{Format: "wav", Data: audio.Data})
+	if err != nil {
+		t.Fatalf("STT error: %v", err)
+	}
+	t.Logf("STT (of TTS audio): text=%q", text)
+
+	// STT on a generated silence wav, to confirm the empty-audio path is accepted.
+	silence := silentWAVBytes(1, 16000)
+	sttText, sttErr := stt.STT(context.Background(), Audio{Format: "wav", Data: silence})
+	if sttErr != nil {
+		t.Logf("STT (silence): error=%v", sttErr)
+	} else {
+		t.Logf("STT (silence): text=%q", sttText)
+	}
+
+	// Exercise the degrading chain (the same object the gateway wires), proving it
+	// returns real bytes/text rather than degrading to the silent terminal rung.
+	ch := New(cfg)
+	chTTS, chTTSErr := ch.TTS(context.Background(), "你好，我是昔涟")
+	if chTTSErr != nil {
+		t.Fatalf("chain TTS error: %v", chTTSErr)
+	}
+	t.Logf("chain TTS: format=%q bytes=%d", chTTS.Format, len(chTTS.Data))
+	if len(chTTS.Data) == 0 {
+		t.Fatal("chain TTS degraded to empty (silent) audio")
+	}
+	chText, chSTTErr := ch.STT(context.Background(), Audio{Format: "wav", Data: chTTS.Data})
+	if chSTTErr != nil {
+		t.Fatalf("chain STT error: %v", chSTTErr)
+	}
+	t.Logf("chain STT: text=%q", chText)
+	if chText == "" {
+		t.Fatal("chain STT degraded to empty (silent) text")
+	}
+}
+
+// silentWAVBytes builds a mono 16-bit 16 kHz silence WAV of durS seconds.
+func silentWAVBytes(durS, sampleRate int) []byte {
+	dataLen := durS * sampleRate * 2
+	buf := &bytes.Buffer{}
+	// RIFF header + fmt chunk + data chunk.
+	buf.WriteString("RIFF")
+	_ = binary.Write(buf, binary.LittleEndian, uint32(36+dataLen))
+	buf.WriteString("WAVEfmt ")
+	_ = binary.Write(buf, binary.LittleEndian, uint32(16))
+	_ = binary.Write(buf, binary.LittleEndian, uint16(1)) // PCM
+	_ = binary.Write(buf, binary.LittleEndian, uint16(1)) // mono
+	_ = binary.Write(buf, binary.LittleEndian, uint32(sampleRate))
+	_ = binary.Write(buf, binary.LittleEndian, uint32(sampleRate*2))
+	_ = binary.Write(buf, binary.LittleEndian, uint16(2))  // block align
+	_ = binary.Write(buf, binary.LittleEndian, uint16(16)) // bits per sample
+	buf.WriteString("data")
+	_ = binary.Write(buf, binary.LittleEndian, uint32(dataLen))
+	buf.Write(make([]byte, dataLen))
+	return buf.Bytes()
+}

internal/runtime/multimodal/voice/dashscope_test.go

@@ -0,0 +1,206 @@
+package voice
+
+import (
+	"context"
+	"encoding/base64"
+	"encoding/hex"
+	"encoding/json"
+	"errors"
+	"net/http"
+	"net/http/httptest"
+	"strings"
+	"testing"
+
+	"git.yeij.top/AskaEth/Cyrene_v2/internal/config"
+)
+
+func TestDashscopeSTT(t *testing.T) {
+	t.Parallel()
+	srv := httptest.NewServer(http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		if !strings.HasSuffix(r.URL.Path, "/generation") {
+			t.Errorf("path = %q, want .../generation", r.URL.Path)
+		}
+		if r.Header.Get("Authorization") != "Bearer sk" {
+			t.Errorf("auth = %q, want Bearer sk", r.Header.Get("Authorization"))
+		}
+		if r.Header.Get("X-DashScope-SSE") != "disable" {
+			t.Errorf("sse = %q, want disable", r.Header.Get("X-DashScope-SSE"))
+		}
+		var body map[string]any
+		if err := json.NewDecoder(r.Body).Decode(&body); err != nil {
+			t.Fatalf("decode: %v", err)
+		}
+		if body["model"] != "qwen-audio-3.0-asr-flash" {
+			t.Errorf("model = %v, want qwen-audio-3.0-asr-flash", body["model"])
+		}
+		input := body["input"].(map[string]any)
+		msgs := input["messages"].([]any)
+		content := msgs[0].(map[string]any)["content"].([]any)
+		audio := content[0].(map[string]any)["input_audio"].(map[string]any)["data"].(string)
+		if !strings.HasPrefix(audio, "data:audio/wav;base64,") {
+			t.Errorf("data uri = %q, want data:audio/wav;base64,...", audio)
+		}
+		params := body["parameters"].(map[string]any)
+		if params["format"] != "wav" {
+			t.Errorf("format = %v, want wav", params["format"])
+		}
+		hints := params["language_hints"].([]any)
+		if len(hints) != 1 || hints[0] != "zh" {
+			t.Errorf("language_hints = %v, want [zh]", hints)
+		}
+		w.Header().Set("Content-Type", "application/json")
+		_, _ = w.Write([]byte(`{"output":{"output":{"sentence":{"text":"你好世界"}}}}`))
+	}))
+	defer srv.Close()
+
+	stt := NewDashscopeSTT(config.VoiceConfig{DashscopeBaseURL: srv.URL, DashscopeAPIKey: "sk"})
+	text, err := stt.STT(context.Background(), Audio{Format: "wav", Data: []byte("rawaudio")})
+	if err != nil {
+		t.Fatalf("STT error: %v", err)
+	}
+	if text != "你好世界" {
+		t.Errorf("text = %q, want 你好世界", text)
+	}
+}
+
+func TestDashscopeSTT_ChoicesText(t *testing.T) {
+	t.Parallel()
+	srv := httptest.NewServer(http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		w.Header().Set("Content-Type", "application/json")
+		_, _ = w.Write([]byte(`{"output":{"choices":[{"message":{"content":[{"text":"hello"}]}}]}}`))
+	}))
+	defer srv.Close()
+
+	stt := NewDashscopeSTT(config.VoiceConfig{DashscopeBaseURL: srv.URL, DashscopeAPIKey: "sk"})
+	text, err := stt.STT(context.Background(), Audio{Format: "wav", Data: []byte("x")})
+	if err != nil {
+		t.Fatalf("STT error: %v", err)
+	}
+	if text != "hello" {
+		t.Errorf("text = %q, want hello", text)
+	}
+}
+
+func TestDashscopeSTT_NotConfigured(t *testing.T) {
+	t.Parallel()
+	stt := NewDashscopeSTT(config.VoiceConfig{})
+	if _, err := stt.STT(context.Background(), Audio{Format: "wav", Data: []byte("x")}); err != ErrNotConfigured {
+		t.Fatalf("err = %v, want ErrNotConfigured", err)
+	}
+}
+
+func TestDashscopeSTT_EmptyAudio(t *testing.T) {
+	t.Parallel()
+	stt := NewDashscopeSTT(config.VoiceConfig{DashscopeBaseURL: "http://x", DashscopeAPIKey: "sk"})
+	if _, err := stt.STT(context.Background(), Audio{}); !errors.Is(err, ErrProviderFailed) {
+		t.Fatalf("err = %v, want ErrProviderFailed", err)
+	}
+}
+
+func TestDashscopeSTT_Non200(t *testing.T) {
+	t.Parallel()
+	srv := httptest.NewServer(http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		w.WriteHeader(http.StatusBadRequest)
+		_, _ = w.Write([]byte(`{"code":"InvalidParameter","message":"bad"}`))
+	}))
+	defer srv.Close()
+
+	stt := NewDashscopeSTT(config.VoiceConfig{DashscopeBaseURL: srv.URL, DashscopeAPIKey: "sk"})
+	_, err := stt.STT(context.Background(), Audio{Format: "wav", Data: []byte("x")})
+	if !errors.Is(err, ErrProviderFailed) {
+		t.Fatalf("err = %v, want ErrProviderFailed", err)
+	}
+	if !strings.Contains(err.Error(), "400") {
+		t.Errorf("err = %v, want it to mention status 400", err)
+	}
+}
+
+func TestDashscopeTTS_URL(t *testing.T) {
+	t.Parallel()
+	audioBytes := []byte("RIFFxxxxWAVE")
+	var srv *httptest.Server
+	srv = httptest.NewServer(http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		switch {
+		case strings.HasSuffix(r.URL.Path, "/SpeechSynthesizer"):
+			if r.Header.Get("Authorization") != "Bearer sk" {
+				t.Errorf("auth = %q, want Bearer sk", r.Header.Get("Authorization"))
+			}
+			if sse := r.Header.Get("X-DashScope-SSE"); sse != "" {
+				t.Errorf("sse = %q, want empty (non-streaming URL)", sse)
+			}
+			var body map[string]any
+			if err := json.NewDecoder(r.Body).Decode(&body); err != nil {
+				t.Fatalf("decode: %v", err)
+			}
+			if body["model"] != "cosyvoice-v3-flash" {
+				t.Errorf("model = %v, want cosyvoice-v3-flash", body["model"])
+			}
+			input := body["input"].(map[string]any)
+			if input["text"] != "你好" {
+				t.Errorf("text = %v, want 你好", input["text"])
+			}
+			if input["voice"] != "longxiaochun_v3" {
+				t.Errorf("voice = %v, want longxiaochun_v3", input["voice"])
+			}
+			if input["format"] != "wav" {
+				t.Errorf("format = %v, want wav", input["format"])
+			}
+			w.Header().Set("Content-Type", "application/json")
+			_, _ = w.Write([]byte(`{"output":{"audio":{"url":"` + srv.URL + `/audio/out.wav"}}}`))
+		case strings.HasSuffix(r.URL.Path, "/audio/out.wav"):
+			w.Header().Set("Content-Type", "audio/wav")
+			_, _ = w.Write(audioBytes)
+		default:
+			t.Errorf("unexpected path %q", r.URL.Path)
+		}
+	}))
+	defer srv.Close()
+
+	tts := NewDashscopeTTS(config.VoiceConfig{DashscopeBaseURL: srv.URL, DashscopeAPIKey: "sk"})
+	audio, err := tts.TTS(context.Background(), "你好")
+	if err != nil {
+		t.Fatalf("TTS error: %v", err)
+	}
+	if audio.Format != "wav" {
+		t.Errorf("format = %q, want wav", audio.Format)
+	}
+	if hex.EncodeToString(audio.Data) != hex.EncodeToString(audioBytes) {
+		t.Errorf("data = %s, want %s", hex.EncodeToString(audio.Data), hex.EncodeToString(audioBytes))
+	}
+}
+
+func TestDashscopeTTS_DataBlob(t *testing.T) {
+	t.Parallel()
+	audioBytes := []byte{9, 8, 7}
+	b64 := base64.StdEncoding.EncodeToString(audioBytes)
+	srv := httptest.NewServer(http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		w.Header().Set("Content-Type", "application/json")
+		_, _ = w.Write([]byte(`{"output":{"audio":{"data":"` + b64 + `"}}}`))
+	}))
+	defer srv.Close()
+
+	tts := NewDashscopeTTS(config.VoiceConfig{DashscopeBaseURL: srv.URL, DashscopeAPIKey: "sk"})
+	audio, err := tts.TTS(context.Background(), "hi")
+	if err != nil {
+		t.Fatalf("TTS error: %v", err)
+	}
+	if hex.EncodeToString(audio.Data) != hex.EncodeToString(audioBytes) {
+		t.Errorf("data = %s, want %s", hex.EncodeToString(audio.Data), hex.EncodeToString(audioBytes))
+	}
+}
+
+func TestDashscopeTTS_EmptyText(t *testing.T) {
+	t.Parallel()
+	tts := NewDashscopeTTS(config.VoiceConfig{DashscopeBaseURL: "http://x", DashscopeAPIKey: "sk"})
+	if _, err := tts.TTS(context.Background(), "   "); err != ErrEmptyText {
+		t.Fatalf("err = %v, want ErrEmptyText", err)
+	}
+}
+
+func TestDashscopeTTS_NotConfigured(t *testing.T) {
+	t.Parallel()
+	tts := NewDashscopeTTS(config.VoiceConfig{})
+	if _, err := tts.TTS(context.Background(), "hi"); err != ErrNotConfigured {
+		t.Fatalf("err = %v, want ErrNotConfigured", err)
+	}
+}

internal/services/gateway/audio.go

@@ -0,0 +1,122 @@
+package gateway
+
+import (
+	"encoding/json"
+	"net/http"
+	"strconv"
+	"strings"
+
+	apperr "git.yeij.top/AskaEth/Cyrene_v2/internal/errors"
+)
+
+// maxAudioRequestBytes caps the JSON body a /api/v1/audio/* request may carry so
+// a huge base64 audio payload cannot exhaust memory.
+const maxAudioRequestBytes = 32 << 20
+
+// transcribeRequest is the POST /api/v1/audio/transcribe body. Audio is a base64
+// (or data: URI) blob; Format is its encoding ("wav"/"mp3"/"ogg").
+type transcribeRequest struct {
+	Audio    string `json:"audio"`
+	Format   string `json:"format"`
+	DeviceID string `json:"deviceId,omitempty"`
+}
+
+// speechRequest is the POST /api/v1/audio/speech body. Voice/format are accepted
+// for forward compatibility; synthesis currently follows the server's
+// VOICE_TTS_VOICE_ID / VOICE_TTS_FORMAT config.
+type speechRequest struct {
+	Text   string `json:"text"`
+	Voice  string `json:"voice,omitempty"`
+	Format string `json:"format,omitempty"`
+}
+
+// handleTranscribe runs STT on the uploaded audio and returns the recognized
+// text. It is the REST source for native clients, bypassing the 4KB WebSocket
+// frame limit that would truncate a base64 audio blob.
+func (s *Server) handleTranscribe(w http.ResponseWriter, r *http.Request) {
+	r.Body = http.MaxBytesReader(w, r.Body, maxAudioRequestBytes)
+	var req transcribeRequest
+	if err := json.NewDecoder(r.Body).Decode(&req); err != nil {
+		s.auditClaims(r, "audio.transcribe", "error")
+		writeError(w, apperr.Wrap(apperr.ErrBadRequest, "bad_request", "invalid request body"))
+		return
+	}
+	if s.voice == nil {
+		s.auditClaims(r, "audio.transcribe", "error")
+		writeError(w, apperr.Wrap(apperr.ErrInternal, "voice_unavailable", "voice service is not configured"))
+		return
+	}
+	audio, err := decodeVoiceAudio(req.Audio, req.Format)
+	if err != nil {
+		s.auditClaims(r, "audio.transcribe", "error")
+		writeError(w, apperr.Wrap(apperr.ErrBadRequest, "bad_request", "audio could not be decoded"))
+		return
+	}
+	if len(audio.Data) == 0 {
+		s.auditClaims(r, "audio.transcribe", "error")
+		writeError(w, apperr.Wrap(apperr.ErrBadRequest, "bad_request", "audio is required"))
+		return
+	}
+	text, err := s.voice.STT(r.Context(), audio)
+	if err != nil {
+		s.auditClaims(r, "audio.transcribe", "error")
+		s.log.Warn("voice transcribe", "err", err)
+		writeError(w, apperr.Wrap(apperr.ErrInternal, "stt_failed", "speech transcription failed"))
+		return
+	}
+	s.auditClaims(r, "audio.transcribe", "ok")
+	writeJSON(w, http.StatusOK, map[string]any{"text": text})
+}
+
+// handleSpeech synthesizes speech for the given text and writes the raw audio
+// bytes (audio/<format>) back — a binary response, not a JSON envelope.
+func (s *Server) handleSpeech(w http.ResponseWriter, r *http.Request) {
+	r.Body = http.MaxBytesReader(w, r.Body, maxAudioRequestBytes)
+	var req speechRequest
+	if err := json.NewDecoder(r.Body).Decode(&req); err != nil {
+		s.auditClaims(r, "audio.speech", "error")
+		writeError(w, apperr.Wrap(apperr.ErrBadRequest, "bad_request", "invalid request body"))
+		return
+	}
+	if strings.TrimSpace(req.Text) == "" {
+		s.auditClaims(r, "audio.speech", "error")
+		writeError(w, apperr.Wrap(apperr.ErrBadRequest, "bad_request", "text is required"))
+		return
+	}
+	if s.voice == nil {
+		s.auditClaims(r, "audio.speech", "error")
+		writeError(w, apperr.Wrap(apperr.ErrInternal, "voice_unavailable", "voice service is not configured"))
+		return
+	}
+	audi, err := s.voice.TTS(r.Context(), req.Text)
+	if err != nil {
+		s.auditClaims(r, "audio.speech", "error")
+		s.log.Warn("voice speech", "err", err)
+		writeError(w, apperr.Wrap(apperr.ErrInternal, "tts_failed", "speech synthesis failed"))
+		return
+	}
+	if len(audi.Data) == 0 {
+		s.auditClaims(r, "audio.speech", "error")
+		writeError(w, apperr.Wrap(apperr.ErrInternal, "tts_empty", "no audio synthesized"))
+		return
+	}
+	s.auditClaims(r, "audio.speech", "ok")
+	w.Header().Set("Content-Type", audioMime(audi.Format))
+	w.Header().Set("Content-Length", strconv.Itoa(len(audi.Data)))
+	w.WriteHeader(http.StatusOK)
+	_, _ = w.Write(audi.Data)
+}
+
+// audioMime maps an audio format to its Content-Type for the speech response.
+func audioMime(format string) string {
+	switch strings.ToLower(format) {
+	case "mp3", "mpeg":
+		return "audio/mpeg"
+	case "ogg", "oga":
+		return "audio/ogg"
+	case "webm":
+		return "audio/webm"
+	default:
+		return "audio/wav"
+	}
+}

internal/services/gateway/audio_test.go

@@ -0,0 +1,124 @@
+package gateway
+
+import (
+	"context"
+	"encoding/base64"
+	"net/http"
+	"testing"
+
+	"git.yeij.top/AskaEth/Cyrene_v2/internal/auth"
+	"git.yeij.top/AskaEth/Cyrene_v2/internal/runtime/multimodal/voice"
+)
+
+// audioStubVoice is a VoiceService returning fixed STT/TTS results for the
+// REST /api/v1/audio/* handler tests.
+type audioStubVoice struct {
+	sttText   string
+	sttErr    error
+	ttsAudio  voice.Audio
+	ttsErr    error
+	gotAudio  voice.Audio
+	gotText   string
+	sttCalled bool
+	ttsCalled bool
+}
+
+func (s *audioStubVoice) STT(ctx context.Context, audio voice.Audio) (string, error) {
+	s.sttCalled = true
+	s.gotAudio = audio
+	return s.sttText, s.sttErr
+}
+
+func (s *audioStubVoice) TTS(ctx context.Context, text string) (voice.Audio, error) {
+	s.ttsCalled = true
+	s.gotText = text
+	return s.ttsAudio, s.ttsErr
+}
+
+func TestHandleTranscribe(t *testing.T) {
+	env := newEnv(t, nil, false)
+	sv := &audioStubVoice{sttText: "你好世界"}
+	env.srv.SetVoice(sv)
+
+	token := env.token(t, auth.RoleOwner)
+	resp, body := doReq(t, http.MethodPost, env.ts.URL+"/api/v1/audio/transcribe", map[string]any{
+		"audio":  base64.StdEncoding.EncodeToString([]byte("raw")),
+		"format": "wav",
+	}, token)
+	if resp.StatusCode != http.StatusOK {
+		t.Fatalf("status = %d, body = %s", resp.StatusCode, body)
+	}
+	if !sv.sttCalled {
+		t.Fatal("STT not called")
+	}
+	if string(sv.gotAudio.Data) != "raw" || sv.gotAudio.Format != "wav" {
+		t.Errorf("audio = %+v, want format wav / data raw", sv.gotAudio)
+	}
+	env2 := decodeBody(t, body)
+	if got := string(env2["data"]); got == "" {
+		t.Errorf("data envelope empty: %s", body)
+	}
+}
+
+func TestHandleTranscribe_AuthRequired(t *testing.T) {
+	env := newEnv(t, nil, false)
+	sv := &audioStubVoice{sttText: "x"}
+	env.srv.SetVoice(sv)
+
+	resp, body := doReq(t, http.MethodPost, env.ts.URL+"/api/v1/audio/transcribe", map[string]any{
+		"audio": base64.StdEncoding.EncodeToString([]byte("raw")),
+	}, "")
+	if resp.StatusCode != http.StatusUnauthorized {
+		t.Fatalf("status = %d, want 401 (body=%s)", resp.StatusCode, body)
+	}
+}
+
+func TestHandleTranscribe_EmptyAudio(t *testing.T) {
+	env := newEnv(t, nil, false)
+	sv := &audioStubVoice{sttText: "x"}
+	env.srv.SetVoice(sv)
+
+	token := env.token(t, auth.RoleOwner)
+	resp, body := doReq(t, http.MethodPost, env.ts.URL+"/api/v1/audio/transcribe", map[string]any{
+		"audio": base64.StdEncoding.EncodeToString([]byte("")),
+	}, token)
+	if resp.StatusCode != http.StatusBadRequest {
+		t.Fatalf("status = %d, want 400 (body=%s)", resp.StatusCode, body)
+	}
+}
+
+func TestHandleSpeech(t *testing.T) {
+	env := newEnv(t, nil, false)
+	sv := &audioStubVoice{ttsAudio: voice.Audio{Format: "wav", Data: []byte("audio-bytes")}}
+	env.srv.SetVoice(sv)
+
+	token := env.token(t, auth.RoleOwner)
+	resp, body := doReq(t, http.MethodPost, env.ts.URL+"/api/v1/audio/speech", map[string]any{
+		"text": "你好",
+	}, token)
+	if resp.StatusCode != http.StatusOK {
+		t.Fatalf("status = %d, body = %s", resp.StatusCode, body)
+	}
+	if ct := resp.Header.Get("Content-Type"); ct != "audio/wav" {
+		t.Errorf("Content-Type = %q, want audio/wav", ct)
+	}
+	if string(body) != "audio-bytes" {
+		t.Errorf("body = %q, want audio-bytes", body)
+	}
+	if !sv.ttsCalled || sv.gotText != "你好" {
+		t.Errorf("TTS got text = %q (called=%v), want 你好", sv.gotText, sv.ttsCalled)
+	}
+}
+
+func TestHandleSpeech_EmptyText(t *testing.T) {
+	env := newEnv(t, nil, false)
+	env.srv.SetVoice(&audioStubVoice{})
+
+	token := env.token(t, auth.RoleOwner)
+	resp, body := doReq(t, http.MethodPost, env.ts.URL+"/api/v1/audio/speech", map[string]any{
+		"text": "  ",
+	}, token)
+	if resp.StatusCode != http.StatusBadRequest {
+		t.Fatalf("status = %d, want 400 (body=%s)", resp.StatusCode, body)
+	}
+}

internal/services/gateway/server.go

@@ -385,6 +385,12 @@ func (s *Server) Handler() http.Handler {
 	mux.Handle("GET /api/v1/sessions", readAuth(http.HandlerFunc(s.handleListSessions)))
 	mux.Handle("GET /api/v1/personas", readAuth(http.HandlerFunc(s.handleListPersonas)))
 
+	// Audio (voice) STT/TTS. Owner-only (the owner's private voice assistant),
+	// audited. These are the REST surface for native clients to upload audio and
+	// fetch synthesized speech, bypassing the 4KB WebSocket frame limit.
+	mux.Handle("POST /api/v1/audio/transcribe", s.auth.Authorize(auth.RoleOwner)(http.HandlerFunc(s.handleTranscribe)))
+	mux.Handle("POST /api/v1/audio/speech", s.auth.Authorize(auth.RoleOwner)(http.HandlerFunc(s.handleSpeech)))
+
 	// Session/model reads and writes are the owner's private history; owner/admin only
 	// (chat uses the same roles). Session metadata in GET /sessions stays readAuth for
 	// any authenticated role, but the message content (and thinking) is owner/admin-only

internal/services/gateway/voice.go

@@ -26,12 +26,14 @@ type VoiceService interface {
 // voiceInputRequest is the decoded payload of a voice_input frame. Audio is a
 // base64 (or data: URI) audio blob; Format is its encoding ("wav"/"mp3"/"ogg");
 // Text overrides STT when present (a client that already transcribed can skip
-// recognition); SessionID defaults to "web".
+// recognition); SessionID defaults to "web"; DeviceID is the client-produced
+// end identifier (design/16 §4.5) threaded into the persisted voice turn.
 type voiceInputRequest struct {
 	Audio     string `json:"audio"`
 	Format    string `json:"format"`
 	Text      string `json:"text"`
 	SessionID string `json:"sessionId"`
+	DeviceID  string `json:"deviceId,omitempty"`
 }
 
 // SetVoice injects the voice STT/TTS service used by the voice_input WS route.
@@ -66,6 +68,9 @@ func (c *wsConn) handleVoiceInput(in wsInbound) {
 // carrying the base64 audio plus the recognized transcript and assistant reply.
 func (h *hub) runVoiceTurn(c *wsConn, req voiceInputRequest) {
 	ctx := context.Background()
+	// Thread the client's deviceID so the persisted voice turn is tagged with the
+	// sending end (design/16 §4.5), mirroring ws_hub.go:240.
+	ctx = withDeviceID(ctx, req.DeviceID)
 	if req.SessionID == "" {
 		req.SessionID = "web"
 	}