livekit-streaming-tts 0.1.0__tar.gz

livekit_streaming_tts-0.1.0/LICENSE

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livekit_streaming_tts-0.1.0/PKG-INFO

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+Metadata-Version: 2.4
+Name: livekit-streaming-tts
+Version: 0.1.0
+Summary: Engine-agnostic, low-latency TTS plugin for LiveKit agents. Pluggable backends: OmniVoice, Pocket TTS, Kokoro, KittenTTS, XTTS, Piper, Bark.
+Author: Thanush
+License: Apache-2.0
+Project-URL: Homepage, https://github.com/Thanush101/livekit-streaming-tts
+Project-URL: Issues, https://github.com/Thanush101/livekit-streaming-tts/issues
+Project-URL: Documentation, https://github.com/Thanush101/livekit-streaming-tts#readme
+Keywords: livekit,tts,voice,streaming,websocket,speech-synthesis
+Classifier: Development Status :: 4 - Beta
+Classifier: Intended Audience :: Developers
+Classifier: License :: OSI Approved :: Apache Software License
+Classifier: Programming Language :: Python :: 3
+Classifier: Programming Language :: Python :: 3.10
+Classifier: Programming Language :: Python :: 3.11
+Classifier: Programming Language :: Python :: 3.12
+Classifier: Topic :: Multimedia :: Sound/Audio :: Speech
+Requires-Python: >=3.10
+Description-Content-Type: text/markdown
+License-File: LICENSE
+Requires-Dist: numpy>=1.24
+Requires-Dist: fastapi>=0.110
+Requires-Dist: uvicorn[standard]>=0.27
+Requires-Dist: aiohttp>=3.9
+Requires-Dist: pydantic>=2.0
+Requires-Dist: inflect>=7.0
+Requires-Dist: livekit-agents>=0.10
+Requires-Dist: python-multipart>=0.0.6
+Provides-Extra: hq-resample
+Requires-Dist: soxr>=0.3; extra == "hq-resample"
+Provides-Extra: omnivoice
+Requires-Dist: torch>=2.1; extra == "omnivoice"
+Requires-Dist: soundfile>=0.12; extra == "omnivoice"
+Provides-Extra: pocket
+Requires-Dist: torch>=2.1; extra == "pocket"
+Provides-Extra: kokoro
+Requires-Dist: kokoro>=0.7; extra == "kokoro"
+Requires-Dist: torch>=2.1; extra == "kokoro"
+Provides-Extra: kitten
+Requires-Dist: kittentts>=0.1; extra == "kitten"
+Provides-Extra: xtts
+Requires-Dist: TTS>=0.22; extra == "xtts"
+Requires-Dist: torch>=2.1; extra == "xtts"
+Provides-Extra: piper
+Requires-Dist: piper-tts>=1.2; extra == "piper"
+Provides-Extra: bark
+Requires-Dist: bark>=0.1; extra == "bark"
+Requires-Dist: torch>=2.1; extra == "bark"
+Provides-Extra: all
+Requires-Dist: torch>=2.1; extra == "all"
+Requires-Dist: soundfile>=0.12; extra == "all"
+Requires-Dist: kokoro>=0.7; extra == "all"
+Requires-Dist: kittentts>=0.1; extra == "all"
+Requires-Dist: TTS>=0.22; extra == "all"
+Requires-Dist: piper-tts>=1.2; extra == "all"
+Requires-Dist: bark>=0.1; extra == "all"
+Requires-Dist: soxr>=0.3; extra == "all"
+Provides-Extra: dev
+Requires-Dist: pytest>=8.0; extra == "dev"
+Requires-Dist: pytest-asyncio>=0.23; extra == "dev"
+Requires-Dist: ruff>=0.4; extra == "dev"
+Dynamic: license-file
+
+# livekit-streaming-tts
+
+Engine-agnostic, low-latency, self-hosted TTS for LiveKit voice agents. One WebSocket protocol, one client plugin, swappable backends — write your agent once, switch from OmniVoice to Pocket TTS to Kokoro by changing one env var.
+
+## Why this exists
+
+Most LiveKit TTS plugins are tightly coupled to one provider — change the model and you rewrite the plugin, redo latency tuning, and rediscover the same edge cases (idle-closed sockets, half-words from the LLM, no barge-in support, etc.). This package factors out the parts that are the same for every TTS:
+
+- A LiveKit `tts.TTS` plugin with pooled WebSockets, sentence-aware streaming, and barge-in.
+- A FastAPI server that does text normalization, sentence chunking, voice caching, and audio encoding (PCM/WAV, any sample rate).
+- A small `TTSEngine` ABC — the only thing each backend implements.
+
+## Supported engines
+
+| Engine     | Voice cloning | Streaming gen | Multilingual | Speed | Footprint    |
+|------------|---------------|---------------|--------------|-------|--------------|
+| OmniVoice  | yes           | no            | yes          | yes   | GPU, ~3GB    |
+| Pocket TTS | yes (kvcache) | no            | en           | no    | GPU, ~1GB    |
+| Kokoro     | no (presets)  | no            | en/ja/zh     | yes   | GPU, ~330MB  |
+| KittenTTS  | no (presets)  | no            | en           | no    | CPU, ~50MB   |
+| XTTS v2    | yes (zero-shot)| yes          | 16 langs     | yes   | GPU, ~2GB    |
+| Piper      | no (per-file) | no            | many         | yes   | CPU, ~30MB   |
+| Bark       | no (presets)  | no            | yes          | no    | GPU, ~5GB    |
+
+## Topology
+
+```
+┌────────────────────────────┐                ┌────────────────────────────┐
+│  CLIENT host (CPU)         │     WS         │  SERVER host (GPU)         │
+│  - LiveKit agent           │ ◄─────────────►│  - FastAPI + uvicorn       │
+│  - StreamingTTS plugin     │                │  - TTSEngine (pocket /     │
+│  - microphone, speaker     │                │    omnivoice / kokoro / …) │
+└────────────────────────────┘                └────────────────────────────┘
+```
+
+The **same `pip install`** ships both halves:
+- the **server** entrypoint (`create_app`, `examples/run_server.py`)
+- the **client** plugin (`StreamingTTS` — drop into a LiveKit agent)
+
+You install the package on the server with the engine extra you want, and on
+the client with no extra (it doesn't need torch / engine deps — only
+`aiohttp` + `livekit-agents`).
+
+## Install
+
+```bash
+git clone https://github.com/<you>/livekit-streaming-tts
+cd livekit-streaming-tts
+
+# --- server host (GPU box, where the model runs) ---
+pip install -e .[pocket]              # or [omnivoice], [kokoro], [xtts] ...
+pip install -e .[kitten]              # CPU-only, no GPU needed
+
+# --- client host (where your LiveKit agent runs) ---
+pip install -e .                      # no extras — just the plugin & deps
+```
+
+Once published to PyPI, those become `pip install livekit-streaming-tts[pocket]`
+on the server and `pip install livekit-streaming-tts` on the client.
+
+## Run the server
+
+```bash
+TTS_ENGINE=pocket    TTS_PORT=8001 python examples/run_server.py
+TTS_ENGINE=omnivoice TTS_PORT=8001 python examples/run_server.py
+TTS_ENGINE=kitten    TTS_PORT=8001 python examples/run_server.py
+```
+
+The engine is loaded lazily — only the deps you `pip install`'d are imported.
+
+Verify it's up:
+
+```bash
+curl http://localhost:8001/health
+# {"status":"ok","engine":"pocket","capabilities":{...},"voices":[...]}
+```
+
+## Connect from the client
+
+Two supported clients:
+
+### 1) LiveKit agent (the main use case)
+
+```python
+from livekit.agents import Agent, AgentSession, WorkerOptions, cli
+from livekit.plugins import openai, silero
+from livekit_streaming_tts import StreamingTTS
+
+tts = StreamingTTS(
+    ws_url="ws://your-server:8001/tts/ws",
+    voice="alba",
+    sample_rate=24000,   # 8000 for telephony
+    format="pcm",
+    binary=True,
+)
+tts.prewarm()
+
+session = AgentSession(
+    vad=silero.VAD.load(),
+    stt=openai.STT(),
+    llm=openai.LLM(model="gpt-4o-mini"),
+    tts=tts,                          # ← that's the entire integration
+)
+```
+
+That's the whole client side. The plugin handles WS pooling, sentence
+tokenization, retries, and barge-in.
+
+See `examples/livekit_agent.py` for a runnable version.
+
+### 2) Raw WebSocket (non-LiveKit integrations)
+
+If you're building a telephony bridge, a CLI, or any non-LiveKit pipeline,
+you can speak the WS protocol directly. See `examples/raw_ws_client.py`:
+
+```bash
+python examples/raw_ws_client.py "Hello from the streaming TTS server."
+# wrote 96000 bytes of PCM → output.wav
+```
+
+The protocol is documented inline in that file and in `server.py`.
+
+## Use in a LiveKit agent
+
+```python
+from livekit_streaming_tts import StreamingTTS
+
+tts = StreamingTTS(
+    ws_url="ws://your-tts-host:8001/tts/ws",
+    voice="alba",            # any registered voice / preset / engine default
+    sample_rate=24000,       # 8000 for telephony — see below
+    format="pcm",            # "pcm" recommended; "wav" if you need a header
+    binary=True,             # binary WS frames for PCM (~33% bandwidth saved)
+)
+tts.prewarm()                # opens one WS so the first turn is fast
+
+session = AgentSession(stt=..., llm=..., tts=tts, vad=...)
+```
+
+## PCM vs WAV — which should you use?
+
+**Use PCM with binary frames.** Three reasons:
+
+1. **Smaller wire size.** WAV adds a 44-byte RIFF header per chunk (small) plus base64 inflates payload by 33% (significant on long replies). PCM-binary skips both.
+2. **No decode step on the client.** The PCM bytes go straight into LiveKit's `AudioEmitter`. WAV-base64 requires `base64.b64decode` + WAV parse on every chunk.
+3. **Quality is identical.** PCM and WAV are the same int16 samples — WAV just wraps them in a header. There is no quality difference.
+
+When to use WAV: only if you're consuming the bytes outside a streaming pipeline (saving to disk, posting to an HTTP endpoint that wants a file format).
+
+## Telephony
+
+PSTN/SIP is 8kHz μ-law or A-law. LiveKit's SIP integration accepts 8kHz/16kHz PCM and handles G.711 conversion. For lowest latency on phone calls:
+
+```python
+StreamingTTS(sample_rate=8000, format="pcm", binary=True)
+```
+
+This makes the server resample once (engine native → 8kHz) and ship 1/3 the bytes vs 24kHz. Install `soxr` for higher-quality resampling:
+
+```bash
+pip install livekit-streaming-tts[hq-resample]
+```
+
+## Voice cloning
+
+Engines that support it: **OmniVoice**, **Pocket TTS**, **XTTS v2**.
+
+```bash
+# Upload a reference clip and register under a voice_id
+curl -F "file=@my-voice.wav" \
+     -F "voice_id=alice" \
+     -F "ref_text=transcript of the clip" \
+     http://your-tts-host:8001/v1/voices/upload
+```
+
+Then pass `voice="alice"` to `StreamingTTS`. The server caches the prompt — subsequent generations are fast.
+
+Pocket TTS bonus: voice prompts are exported to safetensors (kvcache). On restart the server reloads voices from disk in milliseconds instead of recomputing.
+
+## Adding a new engine
+
+```python
+# my_engine.py
+from livekit_streaming_tts.base import TTSEngine, GenerationParams, AudioChunk, EngineCapabilities
+import numpy as np
+
+class MyEngine(TTSEngine):
+    name = "myengine"
+    native_sample_rate = 22050
+    capabilities = EngineCapabilities(voice_cloning=False, multilingual=False)
+
+    def __init__(self):
+        # Load your model here.
+        ...
+
+    def generate(self, params: GenerationParams) -> AudioChunk:
+        # Your synthesis logic. Return float32 in [-1, 1].
+        samples = self._model.synth(params.text)
+        return AudioChunk(samples=samples.astype(np.float32),
+                          sample_rate=self.native_sample_rate, is_final=True)
+
+# Register at runtime:
+from livekit_streaming_tts import register_engine
+register_engine("myengine", MyEngine)
+```
+
+That's it. Sentence chunking, normalization, voice caching, WS protocol, audio encoding, the LiveKit plugin — all reused.
+
+## Architecture
+
+```
+LLM tokens (sub-word frags)
+    │
+    │ via livekit input_ch
+    ▼
+SentenceTokenizer (joins frags → words)        ← client side
+    │
+    ▼
+Pooled WebSocket (max 30s session)             ← client side
+    │
+    │ {"type":"text","data":{"text":"word "}}
+    ▼
+text_buffer + split_sentences (unicode)        ← server side
+    │
+    ▼
+normalize_text (numbers, money, units, MD)     ← server side, off the GPU thread
+    │
+    ▼
+GPUWorker queue (serialized engine access)     ← server side
+    │
+    ▼
+TTSEngine.generate(params)                     ← swappable backend
+    │
+    ▼
+np.clip + int16 + (optional) resample          ← server side
+    │
+    │ binary PCM frame (or base64 WAV)
+    ▼
+LiveKit AudioEmitter → WebRTC → user
+```
+
+## License
+
+Apache 2.0.