Paxa Labs

com.paxalabs由 paxalabs.com 發布v0.1.0-beta.19更新於 Nov 8, 2026

Thai, English, Mandarin speech AI: TTS with playback, transcription, translation, OCR, extraction

概覽

AI 產生的概覽

Paxa Labs 語音 AI:讓助理朗讀、產生附字幕的配音、轉錄、翻譯,並以 OCR 讀取文件。

可以做什麼
  • 朗讀短句,或把長文排入播放佇列連續播放,並可暫停、繼續、跳過與清空。
  • 把語音存成音訊檔案,包括多角色對話腳本,並產生對應的 srt 或 vtt 字幕。
  • 設計、調整、儲存、重新命名與刪除自訂音色,並列出音色、模型與帳戶餘額。
  • 轉錄本機錄音與影片,支援講者標籤、字詞時間、字幕與停頓清單。
  • 在 15 種語言之間翻譯,可控制正式程度、術語表與風格。
  • 以 OCR 把 PDF 與圖片轉成 Markdown,從文件擷取結構化欄位;也可從麥克風錄下一段口頭回覆。
適用情境
適合讓助理在本機開口說話,例如在 Claude Code 中提示需要你確認,或用來製作配音、字幕與轉錄稿。也適合多語言文件閱讀與翻譯工作。
執行需求
透過 npx 在本機執行,需要 Node.js(麥克風聽寫需 Node 22 或更新版本)。呼叫 API 的工具需要 PAXA_API_KEY 中的 Paxa API 金鑰;選用變數可設定預設音色、輸出目錄、模型、語速、麥克風與詞彙表。播放需要命令列音訊播放器,影片輸入與拼接部分音訊格式需要 ffmpeg。
安裝前請注意
各工具依字元、頁數、分鐘或音色設計消耗 Paxa 點數,會產生費用,可用 get_account 查看餘額。PAXA_API_KEY 是金鑰。音訊、文件與錄音會傳送到 Paxa API,listen 會錄製麥克風。音色工具會在帳戶中建立、重新命名與刪除音色,send_feedback 會向廠商送出回饋。

安裝

在 SourceWeft 中

  1. 開啟 儀表板中的 Paxa Labs,將其新增到工作區。
  2. 為需要使用其工具的對話啟用該服務。

Desktop only,透過 STDIO。 STDIO 服務會啟動本機處理程序,因此需要 SourceWeft 桌面主機。

其他 MCP 客戶端

參照 儲存庫 中的啟動說明。

README

@paxalabs/mcp

[npm] [CI] [node] [license] [Claude Desktop bundle]

[Add to Cursor] [Install in VS Code]

Official MCP server for the Paxa Labs API: Thai, English, and Mandarin Chinese speech AI for your agent, including local audio playback, plus transcription, translation, and document reading across languages.

An agent connected to this server can speak out loud through your machine's speakers, read long content aloud as a managed playback queue, save speech to audio files, listen for a spoken reply, transcribe recordings with subtitles, translate between 15 languages, read PDFs and images with OCR, and extract typed fields from documents.

Beta. The tool set is complete and tested end to end, but tool names and behavior may still change before 1.0 as feedback comes in. Report problems at https://github.com/paxalabs/mcp/issues.

Quick start

Add the server, then ask your agent to sign in to Paxa: your browser opens the Paxa sign-in page, and you approve there. New accounts include free credits. See Sign in for the details, or for using an API key instead.

Claude Code

bash
claude mcp add paxa -- npx -y @paxalabs/mcp

Claude Desktop, Cursor, and other MCP clients

Add to your client's MCP configuration (for Claude Desktop: claude_desktop_config.json):

json
{  "mcpServers": {    "paxa": {      "command": "npx",      "args": ["-y", "@paxalabs/mcp"]    }  }}

One click for Cursor or VS Code: the buttons install the same entry into ~/.cursor/mcp.json or VS Code's MCP settings.

[Add to Cursor] [Install in VS Code]

Sign in

Ask your agent to sign in to Paxa (the sign_in tool), or run npx -y @paxalabs/mcp login in a terminal. Your browser opens the Paxa sign-in page with a code filled in, and you approve there. The sign-in is kept in your user config folder (~/Library/Application Support/paxa/auth.json on macOS, ~/.config/paxa/auth.json on Linux, %APPDATA%\paxa\auth.json on Windows), readable only by you, and every Paxa server and paxa say on the machine use it. It renews itself, and lasts until you sign out or leave it unused for 30 days.

The connection is listed under API keys at paxalabs.com/app/keys, where you can give it a credit limit or disconnect it. npx -y @paxalabs/mcp logout signs this machine out.

To use an API key instead, create one there and set PAXA_API_KEY in the server's configuration, for Claude Code:

bash
claude mcp add paxa -e PAXA_API_KEY=pxa_your_key_here -- npx -y @paxalabs/mcp

A key, when set, is used instead of the sign-in. Keys suit CI, machines without a browser, and a separate spending cap per project.

Tools

ToolWhat it doesCredits
speakSynthesize a short line and play it through the speakers, blocking until done10 per 1000 chars (12.5 in a designed voice)
queue_speechRead long content aloud: auto-chunks, synthesizes ahead while playing, returns immediately10 per 1000 chars (12.5 in a designed voice)
control_playbackControl the shared audio queue: status, pause, resume, skip, clearfree
play_audioPlay a local audio file through the speakersfree
text_to_speechSynthesize speech to an audio file (mp3, opus, wav, raw pcm, or mulaw and alaw for phone lines, with sample_rate for pcm and wav) without playing it: long scripts are split and joined into one file, and subtitles writes a matching srt or vtt10 per 1000 chars (12.5 with subtitles or in a designed voice, 15 with both)
dialogue_to_speechSynthesize a script of lines, each in its own voice, to one audio file with a set gap between lines: returns each line's start and end, and subtitles writes captions that never cross two lines10 per 1000 chars, each line at its voice's rate (12.5 with subtitles or in a designed voice, 15 with both)
design_voiceMake a new voice from a written description and play the line it speaks5 per call
refine_voiceChange a draft voice in words, such as "a little lower and slower", and play its new line5 per call
save_voiceSave a draft as one of your voices, with an id the speech tools acceptfree
rename_voiceRename one of your designed voices; the id and sound stayfree
delete_voiceDelete one of your designed voices and free its slotfree
translateTranslate into any of 15 languages from any of them, with formality, glossary, style examples, alternative renderings, and context controls25 per 1000 chars per rendering
ocr_documentOCR a local PDF, PNG, JPEG, or WebP (Thai, English, Simplified Chinese, and more, mixed pages too) into Markdown or structured blocks6.5 per page
extract_fieldsFill a schema of typed fields (Thai IDs, dates, amounts, banks, line items) from a local PDF or image; every value is printed in the document or null with the reason13 per page (19.5 for schemas over 50 fields)
listenRecord one spoken turn from the microphone and return its transcript; the API detects when the user stops talking12.5 per minute listened
send_feedbackRate an output or report a problem to the Paxa team, tied to the request ids the tools printfree
transcribe_audioTranscribe a local recording or video (Thai, English, Mandarin, other languages, mixed; up to 120 minutes) to text, with optional speaker labels, word timings, srt or vtt subtitles, and a list of pauses to cut8.33 per minute
list_voicesYour designed voices and slots, then the TTS voice roster with character notesfree
list_modelsAvailable models, limits, and pricingfree
sign_inConnect this machine to your Paxa account: opens the sign-in page in your browser for you to approvefree
get_accountCredit balance, plan, and rate limitsfree

All audio flows through one ordered queue, so sounds never overlap: speak lines slip in ahead of queued long-form segments, and queue_speech keeps a book or article flowing gap-free by synthesizing the next segment while the current one plays.

With a streaming-capable player installed (see below), speech starts on the first bytes from the API instead of after the full download: about 0.3 s to the first word regardless of length, versus 0.7 s for a short line and 2.5 s for a long paragraph when buffered.

Voice mode for Claude Code

Three pieces turn Claude Code into something you can walk away from: it talks when it has news, and it calls you when it needs you.

1. Install the server (Quick start above).

2. Tell Claude when to talk. Add this to ~/.claude/CLAUDE.md, or to one project's CLAUDE.md:

markdown
## Voice
I have the Paxa MCP server (tools: speak, queue_speech, control_playback).I am often away from the screen, so use voice like this:
- At the end of a turn where you did real work, call speak with a one or  two sentence summary before writing the final message: what you did,  what is next, and anything you need from me.- When you need a decision from me, speak the question too.- Keep it short and conversational. Never read code, file paths, logs, or  long lists aloud. Those stay in text.- Do not speak for quick back-and-forth or trivial answers.- If I ask to hear something long, use queue_speech.- Speak in the language I write in.

3. Get told when Claude needs you. When Claude Code waits for a permission or an answer, the model is not running, so it cannot call speak. Claude Code fires a hook at those moments instead, and paxa say turns the hook into a spoken phrase such as "Permission needed." Put the paxa command on your PATH:

bash
npm install -g @paxalabs/mcp

Then add to ~/.claude/settings.json:

json
{  "hooks": {    "Notification": [      {        "matcher": "permission_prompt|idle_prompt|agent_needs_input",        "hooks": [{ "type": "command", "command": "paxa say" }]      }    ]  }}

paxa say takes the key from PAXA_API_KEY, or from the paxa entry in ~/.claude.json when that is unset, or else uses the sign-in on this machine, so step 1 is all the setup it needs. A Stop hook configured the same way speaks "Done." at the end of every turn.

The built-in phrases are synthesized once per voice and kept in your user cache directory (~/Library/Caches/paxa/say on macOS, ~/.cache/paxa/say on Linux, %LOCALAPPDATA%\paxa\cache\say on Windows). After that first play, which costs well under one credit, a notification plays from disk: no network round trip and no credits.

To change the words, write your own text in the hook command and add --cache so it gets the same treatment. One entry per event, since the matcher picks the event:

json
{  "hooks": {    "Notification": [      {        "matcher": "permission_prompt",        "hooks": [{ "type": "command", "command": "paxa say --cache \"Hey, need your OK\"" }]      },      {        "matcher": "idle_prompt|agent_needs_input",        "hooks": [{ "type": "command", "command": "paxa say --cache --voice cookie \"Your turn\"" }]      }    ]  }}

Without --cache, nothing you type or pipe into paxa say is written to disk, and messages carried inside a hook payload never are.

On macOS the built-in afplay needs about half a second just to start and stop, which is most of the delay you hear on a short phrase. With brew install mpg123 (or ffmpeg) installed, cached phrases play through that instead: mpg123 starts in about 50 ms, ffplay in about 300 ms.

paxa say also works on its own:

bash
paxa say "Build finished"paxa say --voice cookie "Deploy is live"

If your editor or desktop app was not launched from a terminal, its PATH may not include your node bin directory, and the hook will fail silently. Use the absolute path to paxa in the hook command if that happens.

Speech models

Speech runs on Paxa TTS Flash v2 by default: 50 voices that each speak Thai, English, and Mandarin Chinese, and every voice reads direction tags written into the text, such as [whispering], [laughs], [speaking slowly, close to the microphone], a weighted blend like [happy:0.7+sad:0.3], or <emph> and <whisper> marks. Pause tags add silence where they stand: [short pause], [pause], [long pause], or an exact [pause 300ms] up to 500 ms. Nothing in brackets is spoken, tag characters bill as text, and a tag the model cannot do is refused before any charge. The directions guide lists every sound and rule.

temperature (0 to 2, default 0.5) sets how much a reading varies from one request to the next. Use 0 when a line must match the takes around it, such as one sentence regenerated inside a finished voiceover.

Pass model: "paxa-tts-flash-v1" on a call, or set PAXA_TTS_MODEL, only for v1's legacy features: four Thai voices with regional accents (Isan, Northern, Southern), the fixed emotion tags on khanomchan and kaprao, or one of the eight voices v2 does not carry (donut, toast, padthai, yoyo, somtam, larb, khaosoi, roti). Same price and limits on both.

Design your own voice

When no catalog voice fits, describe the one you want. design_voice turns one or two sentences (who is speaking, how they sound, where it will be heard, and the language they speak) into a voice and plays the line it speaks. refine_voice takes one change at a time, such as "a little lower and slower", and the rest of the voice stays; earlier drafts stay usable, so you can go back to one. save_voice keeps the draft you like under an id (voice_...) that speak, queue_speech, and text_to_speech accept on Paxa TTS Flash v2; set it as PAXA_DEFAULT_VOICE to make it the default. Each design or change costs 5 credits, saving is free, and speech in a designed voice costs 1.25x. Your plan sets how many voices you keep (Free 1, Starter 3, Standard 10, Pro 50); list_voices shows them with the slots in use, rename_voice renames one, and delete_voice frees one. The voice description guide has examples.

Content creation

Four things creators asked for, each the local half of an API call:

  • Transcribe a video. Give transcribe_audio an mp4, mov, mkv, or most other containers; ffmpeg pulls the sound track out locally and the API transcribes it. Add pauses_over: 0.7 and the result lists every silence longer than that with its start and end, the dead air to cut. Ask for save: ["srt"] and the captions come back too.
  • Voiceover with captions that match. text_to_speech with subtitles: "srt" (or "vtt") writes a caption file next to the audio, timed from the synthesized words. Cues break at sentence punctuation, at pauses, and at the line budget, never inside a word, and Thai lines end where the writer put a space. A Chinese character takes two columns of the line budget, so Mandarin lines hold 16 characters by default. Timed synthesis bills 1.25x.
  • Long narration to one file. Scripts past 20,000 characters are split at paragraphs and sentences, synthesized part by part, and joined into a single file, captions offset to match.
  • Conversations in several voices. dialogue_to_speech takes the lines in speaking order, each with its voice, and writes one file with gap_ms of silence between lines (300 by default). The result lists where each line starts and ends, and caption cues never cross from one line to the next. Scripts past 100 lines or 20,000 characters are sent in parts and joined with the same gap. If a request fails partway, the lines that arrived whole are kept and the result names the line to resume from.

ffmpeg is the one requirement for video input and for joining mp3, opus, or wav parts (brew install ffmpeg on macOS). Raw pcm, mulaw, and alaw parts join without it, and everything else works without it too.

Translation

translate writes any of 15 languages from any other: Thai, English, Chinese (Simplified), Japanese, Korean, Arabic, French, German, Hindi, Indonesian, Malay, Portuguese, Russian, Spanish, and Vietnamese. target is required; the source is detected per segment, so mixed-language text works, and source pins it when detection is wrong. Formality, borrowed words, glossaries, and style examples follow the target, and alternatives (up to 5) sets how many renderings come back in total, the best first, each billed as another pass over the text. Send a whole glossary: terms that do not appear in the text are dropped before billing, and the result says how many were sent.

Listening

listen turns the microphone into an answer. Ask a question with speak, call listen, and the user's spoken reply comes back as text as soon as they stop talking: Paxa's realtime transcription detects the end of the turn, so nothing has to guess at silence. The audio goes straight to the API and is never written to disk. A few seconds of listening costs well under one credit (12.5 credits per minute, silence included).

It needs a command-line recorder, which stock macOS and Windows do not ship: brew install ffmpeg or brew install sox on macOS; sox on Windows (or ffmpeg with PAXA_MIC set to the DirectShow device name); ffmpeg, sox, parecord, or arecord on Linux. It also needs Node 22 or newer, and the app that launched the server (Terminal, VS Code, Claude Desktop) must have microphone permission. PAXA_MIC picks a device when the system default is not the one you want. To check recognition without a microphone, node scripts/listen-file.mjs recording.mp3 feeds a file through the same session.

Environment variables

VariableRequiredDefaultPurpose
PAXA_API_KEYnoAn API key from paxalabs.com/app/keys. When set, it is used instead of the sign-in
PAXA_OUTPUT_DIRnoworking directoryWhere text_to_speech and dialogue_to_speech save files
PAXA_TTS_MODELnopaxa-tts-flash-v2Speech model for every call that does not pick one. v2 has 50 voices that each speak Thai, English, and Mandarin, and all read direction tags. paxa-tts-flash-v1 only for regional Thai accents, the fixed emotion tags on khanomchan and kaprao, or one of the eight voices v2 does not carry
PAXA_DEFAULT_VOICEnofoithongVoice used when a tool call does not pick one: a catalog voice from list_voices, or one of your designed voices (voice_...). On v2 every catalog voice speaks all three languages, so pick by character
PAXA_VOCABULARYnoKeyword pinning for transcribe_audio: comma-separated names and terms the transcript should spell as written (product names, people, jargon)
PAXA_DEFAULT_SPEEDno1Speaking rate used when a call does not pick one, 0.5 to 1.5. Pitch is preserved and the price is the same; 1.2 to 1.3 suits long reads
PAXA_MICnosystem defaultMicrophone device for listen: an AVFoundation device name on macOS, a PulseAudio or ALSA device on Linux, a DirectShow device name on Windows
PAXA_VOCABULARY_FILEnoA text file with one term per line (# starts a comment), also pinned on every transcription. Read at call time, so edits apply without a restart
PAXA_BASE_URLnohttps://api.paxalabs.comAPI origin override

To pin a different list per project, set the vocabulary variables in a project-scope server entry (Claude Code's project scope, .cursor/mcp.json, .vscode/mcp.json). On each call the tool takes the call's own terms first, then the configured ones, deduplicated and cut at the API's limit of 50, and reports how many it pinned.

Playback support

PlatformFile playerStreaming playerPause/resume
macOSafplay (built in)mpg123, ffplay, or mpv if installedyes
Linuxffplay, mpv, mpg123, paplay, or aplaympg123, ffplay, or mpvyes
Windowsffplay if installed, else PowerShell (wav)mpg123, ffplay, or mpv if installedno

Streaming needs a player that reads from stdin. On macOS, brew install mpg123 (the quickest to start) or ffmpeg enables it; without one, speech still plays through afplay after the download completes. If no player is found at all, speech tools report it clearly and text_to_speech still works.

Claude Desktop extension

Each release on GitHub ships a .mcpb bundle. Download it, open it with Claude Desktop, and ask Claude to sign in to Paxa (or enter an API key in the extension settings). The bundle carries its own copy of the server and its dependencies, so it works without Node.js or npm on the machine.

Development

bash
pnpm installpnpm build        # compile to dist/pnpm typecheckpnpm mcpb         # build release/paxalabs-mcp-<version>.mcpb for Claude Desktop
# live smoke test (spends a few credits, plays audio out loud)PAXA_API_KEY=pxa_... TEST_OUT_DIR=/tmp/paxa-out node scripts/e2e.mjs

來源:README.md,提交 eb12267

工具

21

speak

Speak out loud

Synthesize a short line with Paxa TTS and play it through this machine's speakers. Blocks until playback finishes. Waits for the currently playing audio but jumps ahead of queued long-form segments. For long content use queue_speech. Costs 10 credits per 1000 characters, 12.5 in a designed voice (voice_...). Direction tags (paxa-tts-flash-v2, every voice): words in square brackets written into the text set the delivery from that point on, in English, Thai, or Chinese, e.g. [whispering], [speaking slowly, close to the microphone], [sad:0.5], or a blend [happy:0.7+sad:0.3]; [neutral] returns to the voice's own delivery. Sounds such as [laughs], [sighs], [gasps] play where written; [um] and [hmm] are spoken; <emph>...</emph> and <whisper>...</whisper> mark a span. [short pause], [pause], and [long pause] add silence after the word before them, and [pause 300ms] an exact length up to 500 ms. Nothing in brackets is spoken, tag characters bill as text, and a tag the model cannot do is refused before any charge. On paxa-tts-flash-v1 only khanomchan and kaprao read tags, from a fixed emotion list, and no voice reads pause tags.

輸入參數text必填modelspeedvoicelanguagetemperature

queue_speech

Queue long-form speech

Read long content aloud (stories, articles, books). Splits the text into segments, synthesizes ahead while playing, and returns at once with the segment count and the estimated credits; segments play in order after anything already queued. There is no handle: control_playback monitors and controls the whole queue. Use speak instead for one short line that should play now and wait, and text_to_speech for a file without playback. Costs 10 credits per 1000 characters, 12.5 in a designed voice (voice_...). Direction tags (paxa-tts-flash-v2, every voice): words in square brackets written into the text set the delivery from that point on, in English, Thai, or Chinese, e.g. [whispering], [speaking slowly, close to the microphone], [sad:0.5], or a blend [happy:0.7+sad:0.3]; [neutral] returns to the voice's own delivery. Sounds such as [laughs], [sighs], [gasps] play where written; [um] and [hmm] are spoken; <emph>...</emph> and <whisper>...</whisper> mark a span. [short pause], [pause], and [long pause] add silence after the word before them, and [pause 300ms] an exact length up to 500 ms. Nothing in brackets is spoken, tag characters bill as text, and a tag the model cannot do is refused before any charge. On paxa-tts-flash-v1 only khanomchan and kaprao read tags, from a fixed emotion list, and no voice reads pause tags.

輸入參數text必填modelspeedvoicelanguagetemperature

control_playback

Playback control

Inspect and control the shared audio queue that speak, queue_speech, play_audio, and the voice design tools play through. status reports what is playing, what is queued, and any synthesis failures. pause holds playback until resume or clear, and audio queued meanwhile waits; pause is not supported on Windows, where skip or clear stops playback instead. skip drops the current segment and is refused when nothing is playing. clear stops playback and discards every queued segment, which must be queued again to be heard; audio files already written stay on disk. Free (no API call).

輸入參數action必填

play_audio

Play an audio file

Play a local audio file through this machine's speakers, for example a file produced earlier by text_to_speech. Waits for the audio playing now, plays ahead of queued long-form speech, and blocks until the file ends; control_playback skip or clear stops it early. Fails without playing when the file is missing, is not a playable format, or no audio player is installed; raw pcm, mulaw, and alaw files have no header and cannot be played. To say new text, use speak instead. Free (no API call).

輸入參數file_path必填

text_to_speech

Text to speech (file only)

Synthesize speech with Paxa TTS and save it as an audio file without playing it: voiceovers, narration, audiobooks, phone prompts. To hear speech instead, use speak (a short line) or queue_speech (long reads); play_audio plays a saved file later. The file goes to output_path, replacing a file already there, or to a new timestamped name in PAXA_OUTPUT_DIR. For several speakers in one file, use dialogue_to_speech. Text beyond 20000 characters is split at paragraphs and sentences and the parts are joined into one file (mp3, opus, and wav need ffmpeg for that). With subtitles set, an srt or vtt caption file that matches the voice word for word is written next to the audio. Costs 10 credits per 1000 characters, 12.5 with subtitles, 12.5 in a designed voice (voice_...), 15 with both; every format and sample rate costs the same. Direction tags (paxa-tts-flash-v2, every voice): words in square brackets written into the text set the delivery from that point on, in English, Thai, or Chinese, e.g. [whispering], [speaking slowly, close to the microphone], [sad:0.5], or a blend [happy:0.7+sad:0.3]; [neutral] returns to the voice's own delivery. Sounds such as [laughs], [sighs], [gasps] play where written; [um] and [hmm] are spoken; <emph>...</emph> and <whisper>...</whisper> mark a span. [short pause], [pause], and [long pause] add silence after the word before them, and [pause 300ms] an exact length up to 500 ms. Nothing in brackets is spoken, tag characters bill as text, and a tag the model cannot do is refused before any charge. On paxa-tts-flash-v1 only khanomchan and kaprao read tags, from a fixed emotion list, and no voice reads pause tags.

輸入參數text必填modelspeedvoiceformatlanguagesubtitlesoutput_pathsample_ratetemperaturesubtitle_line_chars

dialogue_to_speech

Dialogue to speech (file only)

Synthesize a script of lines, each in its own voice, into one audio file without playing it: podcasts, interviews, radio plays, lessons and phone flows with more than one speaker. Pass the lines in speaking order with a voice each; gap_ms of silence goes between lines. For a single voice use text_to_speech; to hear a conversation now, call queue_speech once per line in that line's voice; play_audio plays the saved file. The file goes to output_path, replacing a file already there, or to a new timestamped name in PAXA_OUTPUT_DIR. A script over 100 lines or 20000 characters is sent in parts split between lines and joined with the same gap (mp3, opus, and wav need ffmpeg for that). Returns each line's start and end in the audio. With subtitles set, an srt or vtt caption file is written next to the audio, and no cue spans two lines. If synthesis fails partway, the lines already delivered whole, which were paid for, are saved and the result names the line to resume from. Costs the same as text_to_speech for the characters of every line: 10 credits per 1000, 12.5 with subtitles, 12.5 for a line in a designed voice (voice_...), 15 with both; the gap is free. Direction tags (paxa-tts-flash-v2, every voice): words in square brackets written into the text set the delivery from that point on, in English, Thai, or Chinese, e.g. [whispering], [speaking slowly, close to the microphone], [sad:0.5], or a blend [happy:0.7+sad:0.3]; [neutral] returns to the voice's own delivery. Sounds such as [laughs], [sighs], [gasps] play where written; [um] and [hmm] are spoken; <emph>...</emph> and <whisper>...</whisper> mark a span. [short pause], [pause], and [long pause] add silence after the word before them, and [pause 300ms] an exact length up to 500 ms. Nothing in brackets is spoken, tag characters bill as text, and a tag the model cannot do is refused before any charge. On paxa-tts-flash-v1 only khanomchan and kaprao read tags, from a fixed emotion list, and no voice reads pause tags.

輸入參數lines必填modelspeedformatgap_mslanguagesubtitlesoutput_pathsample_ratetemperaturesubtitle_line_chars

design_voice

Design a new voice

Make a new voice from a written description and play the line it speaks through this machine's speakers. Use it when the user wants a voice the catalog does not have. Describe who is speaking and how they sound in one or two sentences: age, gender, role, pitch and texture, pace and manner, where it will be heard, and the language it speaks (a designed voice speaks one language: the one the description names, otherwise the one it is written in), such as "A calm Bangkok pharmacist in her thirties who explains dosages slowly and kindly, speaking Thai". Leave out real people's names, sound effects, and what the voice should say. The result is a draft, not yet a voice: refine_voice changes it, save_voice keeps it. Costs 5 credits per call, the line included; a retry designs again and is charged again.

輸入參數playdescription必填

refine_voice

Refine a draft voice

Change a draft voice in words and play its new line. The instruction says what to change, such as "a little lower and slower" or "warmer, and older"; the rest of the voice stays, so change one quality at a time. Each call returns a new draft handle and the earlier handles keep their versions, so the user can go back to one. save_voice keeps the draft the user picks. Refining never changes a voice already saved: saving a refined draft makes another voice, and rename_voice changes only a name. Costs 5 credits per call, the line included; a retry refines again and is charged again.

輸入參數playdraft必填instruction必填

save_voice

Save a designed voice

Save a draft from design_voice or refine_voice as one of the account's voices, and get its id (voice_...), which speak, queue_speech, and text_to_speech accept as voice on paxa-tts-flash-v2, for this account only. Save only the draft the user picked. Saving the same draft again returns the same voice and uses no slot. A saved voice's sound never changes: design a new voice for a different sound. The plan sets how many voices the account keeps (Free 1, Starter 3, Standard 10, Pro 50); list_voices shows the slots in use. Free.

輸入參數namedraft必填

rename_voice

Rename a designed voice

Rename one of the account's designed voices, given its voice_ id from list_voices or save_voice. The id and the sound stay the same, so anything that uses the id keeps working. The new name takes 1 to 60 characters, and a catalog voice's name is refused. To change how a voice sounds, refine a draft and save it as a new voice; to remove one, use delete_voice. Free.

輸入參數id必填name必填

delete_voice

Delete a designed voice

Delete one of the account's designed voices and free its slot. The id is the voice_ id list_voices shows; it is looked up first, so an id the account does not have is answered without deleting anything, and catalog voices cannot be deleted. Cannot be undone: every later request in the voice fails, and designing again does not bring the same sound back. Use it only when the user asks to delete that voice; to change only its name, use rename_voice. Free.

輸入參數id必填

translate

Translate text

Translate text into any of 15 languages, from any of them, with Paxa Translate: Thai (th), English (en), Chinese (zh), Japanese (ja), Korean (ko), Arabic (ar), French (fr), German (de), Hindi (hi), Indonesian (id), Malay (ms), Portuguese (pt), Russian (ru), Spanish (es), Vietnamese (vi). The source language is detected per segment, so mixed-language text works; a segment already in the target language comes back unchanged. Accepts one string or up to 200 segments that share context, and returns each translation (numbered for segments) with its detected source language, any review flag, and the credits used. For text in a PDF or image, run ocr_document first. Costs 25 credits per 1000 text characters per rendering (minimum 2 credits), so 1000 characters cost 25 credits, or 50 with alternatives 2; instructions, context, examples, do_not_translate strings, and glossary terms bill at 8 per 1000.

輸入參數text必填formatsourcetarget必填contextexamplesglossaryformalityalternativesinstructionsborrowed_wordsdo_not_translate

ocr_document

OCR a document

Run Paxa OCR on a local PDF, PNG, JPEG, or WebP file and return its content as GitHub-flavored Markdown (or structured blocks). Reads Thai, English, Simplified Chinese, and other languages as printed, pages that mix them included, with no language setting; text set in two columns reads column by column. output "structured" returns typed blocks as JSON instead of Markdown. Use extract_fields instead for specific values (a total, a date, an ID number) checked and typed against a schema. Up to 50 pages and 10 MiB per file. Costs 6.5 credits per page.

輸入參數outputfile_path必填

extract_fields

Extract fields from a document

Extract typed fields from a local PDF, PNG, JPEG, or WebP document (up to 20 pages and 10 MiB) with Paxa Document Extraction. You write a schema of fields; every value returned is text printed in the document, read as the field's type, or null with the reason. Nothing is inferred or looked up. Built for Thai documents (receipts, tax invoices, forms, IDs) and fine on English ones. Use ocr_document instead for a document's full text. Returns a status line, the fields as JSON (null where a value is not printed), then the missing, unverified, truncated, and assumed fields. Costs 13 credits per page for a schema of up to 50 leaf fields and 19.5 per page for 51 to 100; the page reading and the evidence spans cost nothing extra. A schema the API refuses costs nothing and names the field and reason. The schema parameter summarizes the dialect; https://paxalabs.com/docs/extraction has every type and key.

輸入參數schema必填file_path必填output_pathinclude_pagesinclude_evidence

transcribe_audio

Transcribe a recording

Transcribe a local recording or video of speech in Thai, English, Mandarin Chinese, or another language, code-switching included (mp3, wav including mu-law and A-law phone recordings, flac, ogg, m4a, aac, webm; or mp4, mov, mkv, avi and other video files, whose sound track is extracted locally with ffmpeg; up to 120 minutes and 95 MiB) with Paxa STT. The transcript comes back inline, so nothing needs to be read from disk afterwards; optionally it is also saved as txt, json (with word timings and segments), srt, or vtt, named after the recording. Existing files are never overwritten. Detects the language and can label speakers; nothing is translated. For one live answer from the microphone, use listen instead. Terms configured in PAXA_VOCABULARY or PAXA_VOCABULARY_FILE are pinned on every call. Costs 8.33 credits per minute of audio (0.1 minimum); subtitles and word timings cost nothing extra. Long recordings take minutes to process.

輸入參數savestylelanguagefile_path必填conventionoutput_dirtimestampsvocabularydiarizationpauses_oversubtitle_line_chars

listen

Listen to the microphone

Record one spoken turn from this machine's microphone and return its transcript (Thai, English, Mandarin Chinese, or another language, code-switching included). Paxa's realtime transcription detects when the user starts and stops talking, so the call returns as soon as the turn ends; the audio goes straight to the API and nothing is written to disk. Waits for any speech this server is playing to finish first. Use it after speak when you asked the user a question and want their spoken answer; for a recording already on disk, use transcribe_audio instead. Returns an empty transcript when nobody spoke within wait_seconds. Costs 12.5 credits per minute of audio sent, silence included (a few seconds of listening costs well under one credit). Needs a microphone recorder on this machine (ffmpeg or sox; parecord or arecord on Linux) and Node 22 or newer; the host app must have microphone permission.

輸入參數stylelanguageconventionvocabularymax_secondswait_seconds

send_feedback

Send feedback to Paxa

Report a wrong or poor output, an API problem, or a platform issue to the Paxa Labs team, tied to the request ids the other tools print ("Request id: ..."). A rating alone on at least one request id is a complete report. For a wrong output, add an example with the exact input, the output received, and the output expected. Returns the report's id once it is filed. Free, no credits. Use it when the user says an output was wrong, or asks to report a problem; do not file reports the user did not ask for.

輸入參數kind必填ratingexamplemessagerequest_ids

list_voices

List voices

List the Paxa TTS voices in one response (no paging): recommended voices first, the account's designed voices with their slots, then the catalog roster per model with ids, names, gender, language, and character notes. Call it when the user asks what voices exist or wants a different one, and before save_voice to see free slots; pass an id from it as voice to speak, queue_speech, or text_to_speech. The default voice needs no lookup. Free.

無需輸入參數。

list_models

List models

List the available Paxa models (speech, transcription, realtime transcription, translation, OCR, and extraction) with their limits and credit pricing, and the languages translate writes. Call it to check a limit or a price before a large job; voices are in list_voices, and the credit balance in get_account. Free.

無需輸入參數。

sign_in

Sign in to Paxa

Connect this machine to the user's Paxa account when no API key is configured: opens the Paxa sign-in page in the user's browser with a code filled in, and the user approves there (new accounts include free credits). Every Paxa tool then works, in every session on this machine, until the user signs out or leaves it unused for 30 days. Waits up to wait_seconds for the approval; if it has not come yet, call sign_in again to keep waiting on the same code. When no browser opens, it returns the page and the code at once for you to give the user. Answers at once when already signed in or when PAXA_API_KEY is set. Free.

輸入參數wait_seconds

get_account

Account and credits

Show the account behind the API key or sign-in, read from the API on each call: credit balance, spending cap, plan, and rate limits. Call it before a large paid job, or after a call fails for credits or a rate limit; prices per model are in list_models. Free.

無需輸入參數。

版本歷史

10
  1. v0.1.0-beta.19最新Nov 7, 2026
  2. v0.1.0-beta.18Nov 5, 2026
  3. v0.1.0-beta.17Nov 3, 2026
  4. v0.1.0-beta.16Oct 24, 2026
  5. v0.1.0-beta.15Oct 18, 2026
  6. v0.1.0-beta.14Oct 18, 2026
  7. v0.1.0-beta.13Oct 8, 2026
  8. v0.1.0-beta.12Oct 7, 2026
  9. v0.1.0-beta.10Sep 22, 2026
  10. v0.1.0-beta.9Sep 16, 2026

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