Herdsman Skill

This directory is not a single script but an integration package for reuse by other agent platforms, enabling external agents to reliably access the Herdsman local model engine.

Use Cases

• Other agent platforms need to use Herdsman as an OpenAI-compatible backend
• Platforms want to access Herdsman via the Anthropic Messages-compatible interface
• Platforms that support the AG-UI protocol need to connect to /agui
• Need to reliably call text, image, OCR, embedding, and speech capabilities without writing long JSON in shell

Default Connection

• Service address: http://127.0.0.1:8080
• OpenAI root path: http://127.0.0.1:8080/v1
• Anthropic endpoint: http://127.0.0.1:8080/v1/anthropic/messages
• AGUI endpoint: http://127.0.0.1:8080/agui
• API Key: Empty by default; if configured, use Authorization: Bearer \x3Ckey>

Mandatory Rules

1. Do not construct complex curl commands directly

Do not construct complex prompts, tools, base64 images, or long-timeout tasks directly in the shell. Prefer using Python scripts under scripts/, or generate temporary Python files following the same pattern.

2. Run model discovery first

Before calling any model, always run:

python headsman-skill/scripts/check_model.py

If you know the model name, you can also:

python headsman-skill/scripts/check_model.py "\x3Cmodel_id>"

3. Long tasks must explicitly set longer timeouts

• Image generation, editing, img2img: recommended timeout >= 120
• OCR: recommended timeout >= 120
• Speech synthesis, recognition, streaming: recommended timeout >= 120
• Text chat: recommended timeout >= 60

4. Save image results to disk

If results will be reused in subsequent conversations, save them to outputs/ and return the absolute path or cache URL to the user.

Protocol Priority

OpenAI Compatible

Preferred for:

• Chat completions
• Tool calls
• Embeddings
• Rerank
• Image generation / editing / img2img
• OCR text recognition
• Speech recognition / synthesis / streaming

Core endpoints:

• GET /v1/models
• POST /v1/chat/completions
• POST /v1/embeddings
• POST /v1/rerank
• POST /v1/images/generations
• POST /v1/images/edits
• POST /v1/images/img2img
• GET /v1/images/cache/:filename
• POST /v1/ocr
• POST /v1/audio/transcriptions
• GET /v1/audio/transcriptions/stream?model= (WebSocket)
• POST /v1/audio/speech
• GET /v1/audio/speech/stream/:token
• GET /v1/audio/info?model=

Additional parameters for chat completions (OpenAI Chat Completions compatible extensions):

Anthropic Compatible

For platforms that only support the Anthropic Messages style, the endpoint is:

• POST /v1/anthropic/messages

Therefore:

• If the platform supports custom full endpoints, it can connect directly
• If the SDK hardcodes /v1/messages, add a lightweight proxy on the platform side or use raw HTTP requests

AGUI

For platforms supporting the AG-UI protocol event stream:

• POST /agui

AGUI is more suitable for protocol clients or SDKs; raw HTTP is not recommended. In the current state, state should at least provide model, and may optionally include webSearch, tools, task_type, pass_through.

Recommended Scripts

• scripts/herdsman_client.py: General HTTP client wrapper
• scripts/check_model.py: Model discovery and filtering
• scripts/chat_completion.py: OpenAI chat completion (supports reasoning_effort / thinking)
• scripts/generate_image.py: Text-to-image generation with auto-download
• scripts/edit_image.py: Image editing with support for local files, URLs, masks, and additional reference images
• scripts/img2img.py: Image-to-image (style transfer, inpainting)
• scripts/ocr.py: OCR text recognition, supports direct local image recognition
• scripts/transcribe_audio.py: Speech transcription, supports local files, URLs, and data URLs
• scripts/audio_speech.py: Text-to-speech (TTS), supports VoiceDesign, VoiceClone, and streaming
• scripts/anthropic_messages.py: Anthropic Messages compatible invocation

Directory Structure

• references/api-examples.md: Capability-based call examples
• references/platform-integration.md: OpenAI / Anthropic / AGUI integration guide
• references/error-codes.md: Common errors and agent-side handling strategies
• references/model-capabilities.md: Model capabilities and endpoint mapping
• outputs/: Recommended directory for saving generated images

Best Practices

1. Use check_model.py first to get installed models
2. Choose OpenAI, Anthropic, or AGUI based on the platform protocol
3. Use Python scripts instead of shell concatenation for long tasks
4. Save image results as files or cache URLs, avoiding large base64 payloads
5. When encountering model_not_found, model_not_installed, invalid_model_capability, re-run model discovery
6. Speech transcription supports both JSON body (audio field) and multipart/form-data (file field)
7. Before OCR, use check_model.py to confirm paddleocr-ppocrv5-server or another OCR model is installed

Speech Extension: TTS Voice Clone + ASR Standalone Transcription

The following three scripts are advanced speech tools integrated with Herdsman, supporting a full workflow from audio conversion to ASR transcription to voice cloning.

Script Overview

convert_audio.py

Convert audio in any format (MP3/M4A/OGG, etc.) to 16kHz mono WAV. No Herdsman dependency.

uv run python scripts/convert_audio.py \x3Cinput_path> [output_path]

Parameters:

• input_path — Path to the reference audio file
• output_path — Optional, defaults to same directory as input with .wav extension

Examples:

uv run python scripts/convert_audio.py ref.mp3
uv run python scripts/convert_audio.py ref.mp3 ref.wav

transcribe_standalone.py

Standalone ASR transcription script (pure urllib, no dependency on herdsman_client.py). Dynamic model selection, supports absolute output paths.

uv run python scripts/transcribe_standalone.py \x3Caudio_path> --model \x3Cmodel_id> [--language \x3Clanguage>] [--output \x3Cabsolute_path>]

Parameters:

• audio_path — Input audio file path (.wav/.mp3/.m4a, etc.)
• --model — ASR model ID (required, dynamic selection)
• --language — Language code (optional, auto-detect by default)
• --output / -o — Output file absolute path, writes both .txt + .json (optional, prints only if not specified)
• --timeout — Timeout in seconds (default 300)

Tested model recommendations:

Examples:

# Recommended
uv run python scripts/transcribe_standalone.py audio.wav --model sherpa-onnx-paraformer-zh-small --output "D:/result.txt"
# Print only
uv run python scripts/transcribe_standalone.py audio.wav --model whisper-base

tts_voice_clone.py

Voice cloning TTS synthesis using qwen3-tts-voiceclone. Three dynamic parameters: reference audio WAV, original text, target script.

uv run python scripts/tts_voice_clone.py \x3Cref_audio_wav> \x3Cref_text> \x3Ctarget_text> [--output \x3Cpath>]

Parameters:

• ref_audio_wav — 16kHz mono WAV path
• ref_text — Original text corresponding to the reference audio
• target_text — Target text to be synthesized with cloned voice
• --output / -o — Output audio path (default ripple_tts_cloned.wav)
• --timeout — Timeout in seconds (default 180)

Examples:

uv run python scripts/tts_voice_clone.py ref.wav "original text" "target synthesis text" -o output.wav

Full Workflow

# 1. Convert to WAV
uv run python scripts/convert_audio.py source.mp3 ref.wav

# 2. ASR transcription (extract audio text for comparison)
uv run python scripts/transcribe_standalone.py ref.wav --model sherpa-onnx-paraformer-zh-small --output "D:/transcribed.txt"

# 3. Voice clone synthesis
uv run python scripts/tts_voice_clone.py ref.wav "original text" "target synthesis text" -o final.wav

Notes

• Reference audio recommended 10-60 seconds, low background noise, natural speech rate
• The original text must exactly match the audio content, otherwise cloning quality is affected
• ASR transcription supports absolute paths via --output for cross-directory use
• Error messages output to stderr, normal results output to stdout