ChatVoice / Speakr Voice Workspace

ChatVoice is the ChatArch repository and Python package shell for the Speakr voice workspace: a FastAPI + browser product with realtime meeting transcription, AI notes, TTS, and full-duplex voice conversation. The deployed product uses Speakr as its canonical public service, while ChatVoice is the package, CLI, and repository name.

Public site: https://speakr.public.wzhecnu.cn/

Repository: https://github.com/ChatArch/ChatVoice

PyPI package: https://pypi.org/project/ChatVoice/

Documentation: https://arch.gh.wzhecnu.cn/ChatVoice/

The former qwen-audio-demo.public.wzhecnu.cn entry is retired and returns HTTP 410.

Features

• 语音合成 (TTS): server-side proxy for qwen-audio-3.0-tts-plus, returning playable MP3/WAV audio when the model provider key is configured.
• 本地声音复刻: Voice Studio is one unified panel: choose 系统音色 (built-in TTS voices) or 我的复刻声音 (upload/record an authorized reference audio sample) and share the same text box. The clone path runs a one-shot VoiceClone/IndexTTS-2.5 job through a local sidecar; the browser receives a playable/downloadable result for the current job only. Within a session the reference audio is kept so the cloned voice can be reused for new text until the page is left. No voice profile or generated-audio history is saved. See 声音复刻使用指南.
• 实时对话: 独立的豆包式语音对话页；browser WebSocket -> FastAPI proxy -> Qwen Realtime，支持服务端模型列表、VAD、流式文字、24 kHz PCM 播放、自然打断、对话历史和 Markdown 导出。
• 会议记录首页: mobile-first recording surface with live transcript, waveform, pause/resume, and finish flow. Audio is used for realtime transcription; the product saves text and summaries, not recording files.
• 暂停整理: pausing a recording commits the current ASR window so pending live/rewrite text can be finalized before continuing.
• 标题刷新与新建快捷入口: the recorder header includes direct 刷新标题 and 新建 buttons for full-session title regeneration and faster mobile topic creation.
• 会议标签: the recorder header includes a compact # tag picker. Meetings default to no tags, support multi-select thought / diary presets plus repeated custom tags, and expose tags through web storage and API-token data reads.
• 清空防误触: resetting a meeting with existing text/summary state requires confirmation.
• 原始录音不保存: 当前会议记录功能不提供“保存录音”或“下载录音”。服务器只保存文字、摘要和会议元数据；访客模式只在浏览器保存文字/摘要。详见 录音保存边界。
• 语音转写: the recorder streams microphone PCM16 to the ASR WebSocket and appends normalized final segments to the timeline.
• API-first ASR: production ASR is designed around api-server, where the ChatVoice backend calls either a managed ASR API or a self-hosted GPU ASR server. stub-local remains available for contract smoke, and funasr-gpu / funasr-cpu remain compatibility channels.
• Realtime ASR WebSocket: WS /ws/asr/stream accepts continuous PCM16 microphone frames and returns cumulative revision events. Long recordings transparently roll a bounded context window while confirmed text continues to grow.
• 会议纪要: final transcript segments can be sent to a server-side Qwen-compatible model for summary, action items, risks, and open questions.
• 双模式会议历史: guests keep meeting text and summaries only in browser IndexedDB; signed-in accounts sync records through authenticated server storage.
• 0.1 API 访问: signed-in users can generate one-time-visible API tokens from the web settings panel; chatvoice data ... can then read meetings, summaries, and realtime conversations from a running service.
• 受邀账号登录: public registration is disabled. Accounts are provisioned by chatvoice accounts add; passwords use salted PBKDF2 hashes, sessions use HttpOnly cookies, and record writes require CSRF tokens.

登录后端与前端边界

ChatVoice 依赖 ChatLogin>=0.1.1,<0.2.0 的认证与会话核心，但保留自己的 HTML、CSS、原生 JavaScript 和登录/访客弹窗，不注入 ChatLogin 默认模板。宿主适配层复用现有 accounts 与 auth_sessions 表、原账号 ID、PBKDF2 材料和 Cookie；不新建第二套用户库、不强制改密码。

/api/auth/* 和原 JSON 字段保持兼容。已有账号映射为普通用户，不引入 Web Admin；会议/对话 owner 检查与 API Token scope 仍由 ChatVoice 负责。访客记录继续保存在浏览器 IndexedDB，不因接入登录库而自动上传。部署与生产数据迁移仍是独立操作，发布包不等于重启服务。

Security model

• The browser never receives or stores provider credentials.
• Set provider credentials only in server-side environment/config storage; never expose them to the browser.
• Do not commit real env files, model caches, probe output, audio files, runtime logs, or generated API token values.
• Guest meeting and conversation records never enter the server database. Audio/transcript data still passes through ASR/summary or Realtime services while a request is processed.
• Raw meeting recordings are not saved by the meeting recorder: no backend raw-audio database/object store, no browser-local recording chunks, and no recording download endpoint in the current version.
• Realtime history stores text, model, and voice only; raw conversation audio is never written to history storage.
• API tokens are stored server-side as hashes. Token values are displayed only once on creation.

Quick start from the released package

python -m venv .venv
. .venv/bin/activate
python -m pip install --upgrade pip
python -m pip install "ChatVoice[web]==0.1.15"

chatvoice --tree
chatvoice --tree-brief
chatvoice service plan --ensure-dirs --json
chatvoice serve app --host 127.0.0.1 --port 18087

Open:

http://127.0.0.1:18087/

For a credential-free wiring smoke, use:

export CHATVOICE_ASR_CHANNEL=stub-local
chatvoice serve app --host 127.0.0.1 --port 18087

For a real ASR backend, keep credentials server-side and call an API provider:

export CHATVOICE_ASR_CHANNEL=api-server
export CHATVOICE_ASR_API_URL="https://<asr-service>/v1/transcribe"
# Configure the optional ASR bearer token in server-side config/env storage; do not put it in argv.
chatvoice serve app --host 127.0.0.1 --port 18087

Meeting summary generation is also a server-side model boundary: configure the notes model/provider in server-side environment or config storage, and let the browser/API read only the saved summary text.

Fresh account, browser, token, and data flow

Create one invited account in the same local runtime database used by the service:

read -r -s CHATVOICE_ACCOUNT_LOGIN
export CHATVOICE_ACCOUNT_LOGIN
chatvoice accounts add person@example.com --display-name "Person" --password-env CHATVOICE_ACCOUNT_LOGIN --json
chatvoice accounts list --json

Then:

1. open the web service;
2. log in with the invited account;
3. create or open a meeting and generate its summary;
4. open 识别设置 → API Token → 生成 Token;
5. copy the token immediately; it is only shown once.

The same token lifecycle is available from CLI after the service is running:

chatvoice tokens create --url http://127.0.0.1:18087 --account person@example.com --password-env CHATVOICE_ACCOUNT_LOGIN --name automation --json
chatvoice tokens list --url http://127.0.0.1:18087 --account person@example.com --password-env CHATVOICE_ACCOUNT_LOGIN --json

Use the token with the data API/CLI:

read -r -s CHATVOICE_DATA_READ
export CHATVOICE_DATA_READ
chatvoice data meetings --url http://127.0.0.1:18087 --token-env CHATVOICE_DATA_READ --json
chatvoice data conversations --url http://127.0.0.1:18087 --token-env CHATVOICE_DATA_READ --json

Database and concurrency

The packaged web app stores service data in one SQLite WAL file at:

<chatarch-home>/chatvoice/data/meetings.sqlite3

Use one service process (--workers 1) with SQLite. Back up and move data with the CLI single-file dump/restore commands. There is no DATABASE_URL ChatVoice setting in the packaged storage layer; the active database is just the resolved SQLite file. Future high-concurrency Postgres/MySQL support is a separate storage-layer migration.

运行目录与数据结构

pip install 后代码安装在当前 Python 环境的 site-packages/chatvoice/，CLI 在对应环境的 bin/chatvoice；生产建议使用独立 venv。运行数据不写入源码目录，默认 root 解析顺序是 CHATVOICE_RUNTIME_ROOT、CHATVOICE_HOME、CHATARCH_HOME/chatvoice、~/.chatarch/chatvoice。默认结构：

~/.chatarch/chatvoice/
├── data/meetings.sqlite3
├── logs/
├── run/
├── temp/asr/
└── model-cache/

后端 SQLite meetings.sqlite3 目前包含 accounts、auth_sessions、api_tokens、meeting_records、conversation_records。转写、summary、会议标签、实时对话消息以 JSON 字符串保存；原始音频不进后端数据库。访客模式仍使用浏览器 IndexedDB 保存本地会议文字、标签和摘要，不保存录音分片。当前版本支持 SQLite WAL + 单服务进程；数据备份/迁移使用 CLI 单文件 dump/restore。高并发 Postgres/MySQL 是未来单独 storage-layer migration。详见 运行目录与数据结构 和 录音保存边界。

API surface

GET /api/heartbeat 是轻量服务心跳，用来判断 Web 服务、SQLite 只读探测和 ASR 状态是否正常；asr.status 会返回 ready、processing 或 degraded，并包含 FunASR 模型是否已热、最近一次识别成功/失败、耗时和输出长度。录音 WebSocket 也会发 asr.stream.processing / asr.stream.heartbeat，前端会显示“模型加载中/识别处理中/失败原因”，不再静默录音无文字。

• GET /api/status: redacted backend status, models, ASR channels, sidecar configuration, and route shapes.
• GET /api/heartbeat: lightweight service/database/ASR heartbeat with model warm-up, processing, and recent error/success state.
• POST /api/tts: JSON {text, voice, format} -> audio/mpeg or audio/wav.
• GET /api/voice-clone/status: redacted local VoiceClone sidecar status.
• POST /api/voice-clone/jobs: authenticated multipart {text, lang, duration_factor, reference_audio} -> one-shot local clone job.
• GET /api/voice-clone/jobs/<id>: authenticated job polling with progress/stage/ETA.
• GET /api/voice-clone/jobs/<id>/audio: authenticated generated audio download/preview.
• POST /api/voice-cloning/create: legacy DashScope enrollment endpoint for reusable voice_id creation; not the primary browser Voice Studio flow.
• GET /api/voice-cloning/list: legacy list of server-side voice enrollment ids by prefix.
• GET /api/asr/channels: available ASR channels.
• POST /api/asr: programmatic/smoke multipart upload endpoint with channel=api-server|funasr-gpu|funasr-cpu|stub-local.
• WS /ws/asr/stream: bounded PCM16 stream used by the recorder.
• GET /api/realtime/models: Realtime models currently exposed by the configured account.
• WS /ws/realtime?model=<id>: browser-to-backend Realtime proxy.
• POST /api/meeting-notes/polish: Qwen-compatible chat completion endpoint for transcript polish + realtime summary structure.
• POST /api/auth/register: intentionally returns 403; self-registration is disabled.
• POST /api/auth/login, GET /api/auth/session, POST /api/auth/logout: invited-account session lifecycle.
• GET|PUT|DELETE /api/meetings[/<id>]: authenticated meeting record storage, including optional tags: string[] metadata. Writes require the session CSRF token.
• GET|PUT|DELETE /api/conversations[/<id>]: authenticated text-only Realtime conversation storage. Writes require the session CSRF token.
• GET|POST /api/tokens, DELETE /api/tokens/<id>: signed-in session token management.
• GET /api/data/meetings[/<id>], GET /api/data/conversations[/<id>]: bearer-token data export for meetings, tags, summaries, and realtime conversations.

Verification

python -m pytest -q
PYTHONPATH=src python -m chatvoice.cli --tree
PYTHONPATH=src python -m chatvoice.cli --tree-brief
python -m mkdocs build --strict
python -m build
python -m twine check dist/*