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Python client for the LiveLLM Server

Project description

LiveLLM Python Client

Python 3.10+ License: MIT

Python client library for the LiveLLM Server - a unified proxy for AI agent, audio, and transcription services.

Features

  • 🚀 Async-first - Built on httpx and websockets for high-performance operations
  • 🔒 Type-safe - Full type hints and Pydantic validation
  • 🎯 Multi-provider - OpenAI, Google, Anthropic, Groq, ElevenLabs
  • 🔄 Streaming - Real-time streaming for agent and audio
  • 🛠️ Flexible API - Use request objects or keyword arguments
  • 🎙️ Audio services - Text-to-speech and transcription
  • 🎤 Real-Time Transcription - WebSocket-based live audio transcription with bidirectional streaming
  • Fallback strategies - Sequential and parallel handling
  • 🧹 Auto cleanup - Context managers and garbage collection

Installation

pip install livellm

Or with development dependencies:

pip install livellm[testing]

Quick Start

import asyncio
from livellm import LivellmClient
from livellm.models import Settings, ProviderKind, TextMessage, MessageRole

async def main():
    # Initialize with automatic provider setup
    async with LivellmClient(
        base_url="http://localhost:8000",
        configs=[
            Settings(
                uid="openai",
                provider=ProviderKind.OPENAI,
                api_key="your-api-key"
            )
        ]
    ) as client:
        # Simple keyword arguments style (gen_config as kwargs)
        response = await client.agent_run(
            provider_uid="openai",
            model="gpt-4",
            messages=[TextMessage(role="user", content="Hello!")],
            temperature=0.7
        )
        print(response.output)

asyncio.run(main())

Configuration

Client Initialization

from livellm import LivellmClient
from livellm.models import Settings, ProviderKind

# Basic
client = LivellmClient(base_url="http://localhost:8000")

# With timeout and pre-configured providers
client = LivellmClient(
    base_url="http://localhost:8000",
    timeout=30.0,
    configs=[
        Settings(
            uid="openai",
            provider=ProviderKind.OPENAI,
            api_key="sk-...",
            base_url="https://api.openai.com/v1"  # Optional
        ),
        Settings(
            uid="anthropic",
            provider=ProviderKind.ANTHROPIC,
            api_key="sk-ant-...",
            blacklist_models=["claude-instant-1"]  # Optional
        )
    ]
)

Supported Providers

OPENAIGOOGLEANTHROPICGROQELEVENLABS

# Add provider dynamically
await client.update_config(Settings(
    uid="my-provider",
    provider=ProviderKind.OPENAI,
    api_key="your-api-key"
))

# List and delete
configs = await client.get_configs()
await client.delete_config("my-provider")

Usage Examples

Agent Services

Two Ways to Call Methods

All methods support two calling styles:

Style 1: Keyword arguments (kwargs become gen_config)

response = await client.agent_run(
    provider_uid="openai",
    model="gpt-4",
    messages=[TextMessage(role="user", content="Hello!")],
    temperature=0.7,
    max_tokens=500
)

Style 2: Request objects

from livellm.models import AgentRequest

response = await client.agent_run(
    AgentRequest(
        provider_uid="openai",
        model="gpt-4",
        messages=[TextMessage(role="user", content="Hello!")],
        gen_config={"temperature": 0.7, "max_tokens": 500}
    )
)

Basic Agent Run

from livellm.models import TextMessage

# Using kwargs (recommended for simplicity)
response = await client.agent_run(
    provider_uid="openai",
    model="gpt-4",
    messages=[
        TextMessage(role="system", content="You are helpful."),
        TextMessage(role="user", content="Explain quantum computing")
    ],
    temperature=0.7,
    max_tokens=500
)
print(f"Output: {response.output}")
print(f"Tokens: {response.usage.input_tokens} in, {response.usage.output_tokens} out")

Streaming Agent Response

# Streaming also supports both styles
stream = client.agent_run_stream(
    provider_uid="openai",
    model="gpt-4",
    messages=[TextMessage(role="user", content="Tell me a story")],
    temperature=0.8
)

async for chunk in stream:
    print(chunk.output, end="", flush=True)

Agent with Vision (Binary Messages)

import base64
from livellm.models import BinaryMessage

with open("image.jpg", "rb") as f:
    image_data = base64.b64encode(f.read()).decode("utf-8")

response = await client.agent_run(
    provider_uid="openai",
    model="gpt-4-vision",
    messages=[
        BinaryMessage(
            role="user",
            content=image_data,
            mime_type="image/jpeg",
            caption="What's in this image?"
        )
    ]
)

Agent with Tools

from livellm.models import WebSearchInput, MCPStreamableServerInput, ToolKind

# Web search tool
response = await client.agent_run(
    provider_uid="openai",
    model="gpt-4",
    messages=[TextMessage(role="user", content="Latest AI news?")],
    tools=[WebSearchInput(
        kind=ToolKind.WEB_SEARCH,
        search_context_size="high"  # low, medium, or high
    )]
)

# MCP server tool
response = await client.agent_run(
    provider_uid="openai",
    model="gpt-4",
    messages=[TextMessage(role="user", content="Run custom tool")],
    tools=[MCPStreamableServerInput(
        kind=ToolKind.MCP_STREAMABLE_SERVER,
        url="http://mcp-server:8080",
        prefix="mcp_",
        timeout=15
    )]
)

Audio Services

Text-to-Speech

from livellm.models import SpeakMimeType

# Non-streaming
audio = await client.speak(
    provider_uid="openai",
    model="tts-1",
    text="Hello, world!",
    voice="alloy",
    mime_type=SpeakMimeType.MP3,
    sample_rate=24000,
    speed=1.0  # kwargs become gen_config
)
with open("output.mp3", "wb") as f:
    f.write(audio)

# Streaming
audio = bytes()
async for chunk in client.speak_stream(
    provider_uid="openai",
    model="tts-1",
    text="Hello, world!",
    voice="alloy",
    mime_type=SpeakMimeType.PCM,
    sample_rate=24000
):
    audio += chunk

# Save PCM as WAV
import wave
with wave.open("output.wav", "wb") as wf:
    wf.setnchannels(1)
    wf.setsampwidth(2)
    wf.setframerate(24000)
    wf.writeframes(audio)

Transcription

# Method 1: Multipart upload (kwargs style)
with open("audio.wav", "rb") as f:
    audio_bytes = f.read()

transcription = await client.transcribe(
    provider_uid="openai",
    file=("audio.wav", audio_bytes, "audio/wav"),
    model="whisper-1",
    language="en",  # Optional
    temperature=0.0  # kwargs become gen_config
)
print(f"Text: {transcription.text}")
print(f"Language: {transcription.language}")

# Method 2: JSON request object (base64-encoded)
import base64
from livellm.models import TranscribeRequest

audio_b64 = base64.b64encode(audio_bytes).decode("utf-8")
transcription = await client.transcribe(
    TranscribeRequest(
        provider_uid="openai",
        file=("audio.wav", audio_b64, "audio/wav"),
        model="whisper-1"
    )
)

Real-Time Transcription (WebSocket)

The realtime transcription API is available either directly via TranscriptionWsClient or through LivellmClient.realtime.transcription.

Using TranscriptionWsClient directly

import asyncio
from livellm import TranscriptionWsClient
from livellm.models import (
    TranscriptionInitWsRequest,
    TranscriptionAudioChunkWsRequest,
    SpeakMimeType,
)

async def transcribe_live_direct():
    base_url = "ws://localhost:8000"  # WebSocket base URL

    async with TranscriptionWsClient(base_url, timeout=30) as client:
        # Define audio source (file, microphone, stream, etc.)
        async def audio_source():
            with open("audio.pcm", "rb") as f:
                while chunk := f.read(4096):
                    yield TranscriptionAudioChunkWsRequest(audio=chunk)
                    await asyncio.sleep(0.1)  # Simulate real-time

        # Initialize transcription session
        init_request = TranscriptionInitWsRequest(
            provider_uid="openai",
            model="gpt-4o-mini-transcribe",
            language="en",  # or "auto" for detection
            input_sample_rate=24000,
            input_audio_format=SpeakMimeType.PCM,
            gen_config={},
        )

        # Stream audio and receive transcriptions
        async for response in client.start_session(init_request, audio_source()):
            print(f"Transcription: {response.transcription}")
            if response.is_end:
                print("Transcription complete!")
                break

asyncio.run(transcribe_live_direct())

Using LivellmClient.realtime.transcription (and running agents while listening)

import asyncio
from livellm import LivellmClient
from livellm.models import (
    TextMessage,
    TranscriptionInitWsRequest,
    TranscriptionAudioChunkWsRequest,
    SpeakMimeType,
)

async def transcribe_and_chat():
    # Central HTTP client; .realtime and .transcription expose WebSocket APIs
    client = LivellmClient(base_url="http://localhost:8000", timeout=30)

    async with client.realtime as realtime:
        async with realtime.transcription as t_client:
            async def audio_source():
                with open("audio.pcm", "rb") as f:
                    while chunk := f.read(4096):
                        yield TranscriptionAudioChunkWsRequest(audio=chunk)
                        await asyncio.sleep(0.1)

            init_request = TranscriptionInitWsRequest(
                provider_uid="openai",
                model="gpt-4o-mini-transcribe",
                language="en",
                input_sample_rate=24000,
                input_audio_format=SpeakMimeType.PCM,
                gen_config={},
            )

            # Listen for transcriptions and, for each chunk, run an agent request
            async for resp in t_client.start_session(init_request, audio_source()):
                print("User said:", resp.transcription)

                # You can call agent_run (or speak, etc.) while the transcription stream is active
                agent_response = await realtime.agent_run(
                    provider_uid="openai",
                    model="gpt-4",
                    messages=[
                        TextMessage(role="user", content=resp.transcription),
                    ],
                    temperature=0.7,
                )
                print("Agent:", agent_response.output)

                if resp.is_end:
                    print("Transcription session complete")
                    break

asyncio.run(transcribe_and_chat())

Supported Audio Formats:

  • PCM: 16-bit uncompressed (recommended)
  • μ-law: 8-bit telephony format (North America/Japan)
  • A-law: 8-bit telephony format (Europe/rest of world)

Use Cases:

  • 🎙️ Voice assistants and chatbots
  • 📝 Live captioning and subtitles
  • 🎤 Meeting transcription
  • 🗣️ Voice commands and control

See also:

Fallback Strategies

Handle failures automatically with sequential or parallel fallback:

from livellm.models import AgentRequest, AgentFallbackRequest, FallbackStrategy, TextMessage

messages = [TextMessage(role="user", content="Hello!")]

# Sequential: try each in order until one succeeds
response = await client.agent_run(
    AgentFallbackRequest(
        strategy=FallbackStrategy.SEQUENTIAL,
        requests=[
            AgentRequest(provider_uid="primary", model="gpt-4", messages=messages, tools=[]),
            AgentRequest(provider_uid="backup", model="claude-3", messages=messages, tools=[])
        ],
        timeout_per_request=30
    )
)

# Parallel: try all simultaneously, use first success
response = await client.agent_run(
    AgentFallbackRequest(
        strategy=FallbackStrategy.PARALLEL,
        requests=[
            AgentRequest(provider_uid="p1", model="gpt-4", messages=messages, tools=[]),
            AgentRequest(provider_uid="p2", model="claude-3", messages=messages, tools=[]),
            AgentRequest(provider_uid="p3", model="gemini-pro", messages=messages, tools=[])
        ],
        timeout_per_request=10
    )
)

# Also works for audio
from livellm.models import AudioFallbackRequest, SpeakRequest

audio = await client.speak(
    AudioFallbackRequest(
        strategy=FallbackStrategy.SEQUENTIAL,
        requests=[
            SpeakRequest(provider_uid="elevenlabs", model="turbo", text="Hi", 
                        voice="rachel", mime_type=SpeakMimeType.MP3, sample_rate=44100),
            SpeakRequest(provider_uid="openai", model="tts-1", text="Hi",
                        voice="alloy", mime_type=SpeakMimeType.MP3, sample_rate=44100)
        ]
    )
)

Resource Management

Recommended: Use context managers for automatic cleanup.

# ✅ Best: Context manager (auto cleanup)
async with LivellmClient(base_url="http://localhost:8000") as client:
    response = await client.ping()
# Configs deleted, connection closed automatically

# ✅ Good: Manual cleanup
client = LivellmClient(base_url="http://localhost:8000")
try:
    response = await client.ping()
finally:
    await client.cleanup()

# ⚠️ OK: Garbage collection (shows warning if configs exist)
client = LivellmClient(base_url="http://localhost:8000")
response = await client.ping()
# Cleaned up when object is destroyed

API Reference

Client Methods

Configuration

  • ping() - Health check
  • update_config(config) / update_configs(configs) - Add/update providers
  • get_configs() - List all configurations
  • delete_config(uid) - Remove provider

Agent

  • agent_run(request | **kwargs) - Run agent (blocking)
  • agent_run_stream(request | **kwargs) - Run agent (streaming)

Audio

  • speak(request | **kwargs) - Text-to-speech (blocking)
  • speak_stream(request | **kwargs) - Text-to-speech (streaming)
  • transcribe(request | **kwargs) - Speech-to-text

Real-Time Transcription (TranscriptionWsClient)

  • connect() - Establish WebSocket connection
  • disconnect() - Close WebSocket connection
  • start_session(init_request, audio_source) - Start bidirectional streaming transcription
  • async with client: - Auto connection management (recommended)

Cleanup

  • cleanup() - Release resources
  • async with client: - Auto cleanup (recommended)

Key Models

Core

  • Settings(uid, provider, api_key, base_url?, blacklist_models?) - Provider config
  • ProviderKind - OPENAI | GOOGLE | ANTHROPIC | GROQ | ELEVENLABS

Messages

  • TextMessage(role, content) - Text message
  • BinaryMessage(role, content, mime_type, caption?) - Image/audio message
  • MessageRole - USER | MODEL | SYSTEM (or use strings: "user", "model", "system")

Requests

  • AgentRequest(provider_uid, model, messages, tools?, gen_config?)
  • SpeakRequest(provider_uid, model, text, voice, mime_type, sample_rate, gen_config?)
  • TranscribeRequest(provider_uid, file, model, language?, gen_config?)
  • TranscriptionInitWsRequest(provider_uid, model, language?, input_sample_rate?, input_audio_format?, gen_config?)
  • TranscriptionAudioChunkWsRequest(audio) - Audio chunk for streaming

Tools

  • WebSearchInput(kind=ToolKind.WEB_SEARCH, search_context_size)
  • MCPStreamableServerInput(kind=ToolKind.MCP_STREAMABLE_SERVER, url, prefix?, timeout?)

Fallback

  • AgentFallbackRequest(strategy, requests, timeout_per_request?)
  • AudioFallbackRequest(strategy, requests, timeout_per_request?)
  • FallbackStrategy - SEQUENTIAL | PARALLEL

Responses

  • AgentResponse(output, usage{input_tokens, output_tokens}, ...)
  • TranscribeResponse(text, language)
  • TranscriptionWsResponse(transcription, is_end) - Real-time transcription result

Error Handling

import httpx

try:
    response = await client.agent_run(
        provider_uid="openai",
        model="gpt-4",
        messages=[TextMessage(role="user", content="Hi")]
    )
except httpx.HTTPStatusError as e:
    print(f"HTTP {e.response.status_code}: {e.response.text}")
except httpx.RequestError as e:
    print(f"Request failed: {e}")

Development

# Install with dev dependencies
pip install -e ".[testing]"

# Run tests
pytest tests/

# Type checking
mypy livellm

Requirements

  • Python 3.10+
  • httpx >= 0.27.0
  • pydantic >= 2.0.0
  • websockets >= 15.0.1

Documentation

Links

License

MIT License - see LICENSE file for details.

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