Flowyaipc Herdsman Skill En

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Integration package for the Herdsman model engine. Used by other agent platforms to call scripts in this directory and protocol specifications when connecting to OpenAI, Anthropic, or AGUI-compatible services.

Install

openclaw skills install flowyaipc-herdsman-skill-1-0-2-en

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 <key>

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 "<model_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):

Parameter	Type	Description
`reasoning_effort`	string	Reasoning level: `low` / `medium` / `high`; local llama.cpp maps to template parameters
`thinking_enabled`	boolean	Enable or disable thinking mode for supported models; local llama.cpp maps to `enable_thinking`
`thinking_tokens`	number	Thinking token budget; local llama.cpp maps to `reasoning_budget`

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

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

Script	Function	External Dependency
`scripts/convert_audio.py`	Audio format conversion (any format to 16kHz WAV)	ffmpeg
`scripts/transcribe_standalone.py`	ASR speech transcription (pure urllib, no herdsman_client dependency)	Herdsman ASR model
`scripts/tts_voice_clone.py`	Voice cloning TTS synthesis	Herdsman qwen3-tts-voiceclone

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 <input_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 <audio_path> --model <model_id> [--language <language>] [--output <absolute_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:

Model	Recommendation	Notes
`sherpa-onnx-paraformer-zh-small`	⭐ Preferred	Simplified Chinese, preserves filler words, ~5s fastest
`whisper-base`	Alternative	General high accuracy, Traditional Chinese output
`funasr`	⚠️	WebSocket streaming only, HTTP not supported
`sherpa-onnx-streaming-zipformer-zh-14m`	⚠️	Streaming only, HTTP does not support full transcription

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 <ref_audio_wav> <ref_text> <target_text> [--output <path>]

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

Flowyaipc Herdsman Skill En

Install

Herdsman Skill

Use Cases

Default Connection

Mandatory Rules

1. Do not construct complex curl commands directly

2. Run model discovery first

3. Long tasks must explicitly set longer timeouts

4. Save image results to disk

Protocol Priority

OpenAI Compatible

Anthropic Compatible

AGUI

Recommended Scripts

Directory Structure

Best Practices

Speech Extension: TTS Voice Clone + ASR Standalone Transcription

Script Overview

convert_audio.py

transcribe_standalone.py

tts_voice_clone.py

Full Workflow

Notes

1. Do not construct complex `curl` commands directly