speech-to-speech 0.2.0__tar.gz

speech_to_speech-0.2.0/LICENSE

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speech_to_speech-0.2.0/MANIFEST.in

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+prune tests
+global-exclude __pycache__
+global-exclude *.py[cod]
+global-exclude .DS_Store

speech_to_speech-0.2.0/PKG-INFO

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+Metadata-Version: 2.4
+Name: speech-to-speech
+Version: 0.2.0
+Summary: Low-latency speech-to-speech pipeline
+Author: Hugging Face
+License-Expression: Apache-2.0
+Project-URL: Homepage, https://github.com/huggingface/speech-to-speech
+Project-URL: Repository, https://github.com/huggingface/speech-to-speech
+Project-URL: Issues, https://github.com/huggingface/speech-to-speech/issues
+Keywords: speech-to-speech,voice-agents,speech-recognition,text-to-speech,openai
+Classifier: Development Status :: 3 - Alpha
+Classifier: Intended Audience :: Developers
+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
+Classifier: Topic :: Scientific/Engineering :: Artificial Intelligence
+Requires-Python: >=3.10
+Description-Content-Type: text/markdown
+License-File: LICENSE
+Requires-Dist: fastapi>=0.115.0
+Requires-Dist: httpx>=0.28.0
+Requires-Dist: nltk==3.9.1
+Requires-Dist: numpy>=1.26.0
+Requires-Dist: openai==2.28.0
+Requires-Dist: pydantic>=2.0
+Requires-Dist: rich>=13.0
+Requires-Dist: scipy>=1.10.0
+Requires-Dist: sounddevice>=0.5.0
+Requires-Dist: soundfile>=0.13.0; platform_system == "Darwin"
+Requires-Dist: torch==2.11.0; platform_system == "Darwin"
+Requires-Dist: torch>=2.4.0; platform_system != "Darwin"
+Requires-Dist: transformers>=4.57.0
+Requires-Dist: uvicorn>=0.30.0
+Requires-Dist: nano-parakeet>=0.2.0; platform_system != "Darwin"
+Requires-Dist: faster-qwen3-tts>=0.2.6; platform_system != "Darwin"
+Requires-Dist: mlx-audio<0.5.0,>=0.4.0; platform_system == "Darwin"
+Requires-Dist: misaki>=0.9.4; platform_system == "Darwin"
+Requires-Dist: espeakng-loader>=0.2.4; platform_system == "Darwin"
+Requires-Dist: spacy>=3.8.4; platform_system == "Darwin"
+Requires-Dist: phonemizer-fork>=3.3.2; platform_system == "Darwin"
+Provides-Extra: chattts
+Requires-Dist: ChatTTS>=0.1.1; extra == "chattts"
+Provides-Extra: facebook-mms
+Requires-Dist: transformers>=4.57.0; extra == "facebook-mms"
+Provides-Extra: faster-whisper
+Requires-Dist: faster-whisper>=1.0.3; extra == "faster-whisper"
+Provides-Extra: kokoro
+Requires-Dist: kokoro>=0.9.2; platform_system != "Darwin" and extra == "kokoro"
+Provides-Extra: language-detection
+Requires-Dist: lingua-language-detector>=2.0.2; extra == "language-detection"
+Provides-Extra: mlx-lm
+Requires-Dist: mlx-lm>=0.14.0; platform_system == "Darwin" and extra == "mlx-lm"
+Requires-Dist: mlx-vlm==0.4.1; platform_system == "Darwin" and extra == "mlx-lm"
+Provides-Extra: paraformer
+Requires-Dist: funasr>=1.1.6; extra == "paraformer"
+Requires-Dist: modelscope>=1.17.1; extra == "paraformer"
+Requires-Dist: onnxruntime<1.24; python_version < "3.11" and extra == "paraformer"
+Provides-Extra: pocket
+Requires-Dist: pocket-tts>=0.1.0; extra == "pocket"
+Provides-Extra: websocket
+Requires-Dist: websockets>=12.0; extra == "websocket"
+Provides-Extra: whisper-mlx
+Requires-Dist: lightning-whisper-mlx>=0.0.10; platform_system == "Darwin" and extra == "whisper-mlx"
+Dynamic: license-file
+
+<div align="center">
+  <div>&nbsp;</div>
+  <img src="https://raw.githubusercontent.com/huggingface/speech-to-speech/main/logo.png" width="600"/>
+</div>
+
+# Speech To Speech: Build local voice agents with open-source models
+
+## 📖 Quick Index
+* [Approach](#approach)
+  - [Structure](#structure)
+  - [Modularity](#modularity)
+* [Setup](#setup)
+* [Usage](#usage)
+  - [Realtime approach](#realtime-approach)
+  - [Server/Client approach](#serverclient-approach)
+  - [WebSocket approach](#websocket-approach)
+  - [Local approach](#local-approach-running-on-mac)
+  - [Docker Server approach](#docker-server)
+* [Command-line usage](#command-line-usage)
+  - [Model parameters](#model-parameters)
+  - [Generation parameters](#generation-parameters)
+  - [Notable parameters](#notable-parameters)
+
+## Approach
+
+### Structure
+This repository implements a speech-to-speech cascaded pipeline consisting of the following parts:
+1. **Voice Activity Detection (VAD)**
+2. **Speech to Text (STT)**
+3. **Language Model (LM)**
+4. **Text to Speech (TTS)**
+
+### Modularity
+The pipeline provides a fully open and modular approach, with a focus on leveraging models available through the Transformers library on the Hugging Face hub. The code is designed for easy modification, and we already support device-specific and external library implementations:
+
+**VAD** 
+- [Silero VAD v5](https://github.com/snakers4/silero-vad)
+
+**STT**
+- Any [Whisper](https://huggingface.co/docs/transformers/en/model_doc/whisper) model checkpoint on the Hugging Face Hub through Transformers 🤗, including [whisper-large-v3](https://huggingface.co/openai/whisper-large-v3) and [distil-large-v3](https://huggingface.co/distil-whisper/distil-large-v3)
+- [Lightning Whisper MLX](https://github.com/mustafaaljadery/lightning-whisper-mlx?tab=readme-ov-file#lightning-whisper-mlx)
+- [MLX Audio Whisper](https://github.com/huggingface/mlx-audio) - Fast Whisper inference on Apple Silicon
+- [Parakeet TDT](https://huggingface.co/nvidia/parakeet-tdt-1.1b) - Real-time streaming STT with sub-100ms latency on Apple Silicon (CUDA/CPU via nano-parakeet, no NeMo)
+- [Paraformer - FunASR](https://github.com/modelscope/FunASR)
+
+**LLM**
+- Any instruction-following model on the [Hugging Face Hub](https://huggingface.co/models?pipeline_tag=text-generation&sort=trending) via Transformers 🤗
+- [mlx-lm](https://github.com/ml-explore/mlx-examples/blob/main/llms/README.md)
+- [OpenAI API](https://platform.openai.com/docs/quickstart)
+
+**TTS**
+- [ChatTTS](https://github.com/2noise/ChatTTS?tab=readme-ov-file)
+- [Pocket TTS](https://github.com/kyutai-labs/pocket-tts) - Streaming TTS with voice cloning from Kyutai Labs
+- [Kokoro-82M](https://huggingface.co/hexgrad/Kokoro-82M) - Fast and high-quality TTS optimized for Apple Silicon
+- [Qwen3-TTS](https://huggingface.co/Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice)
+
+## Setup
+
+Install the default package from PyPI:
+```bash
+pip install speech-to-speech
+```
+
+The default install is scoped to the standard realtime voice-agent path:
+- Parakeet TDT for STT
+- OpenAI-compatible API for the language model
+- Qwen3-TTS for speech output
+- local audio and realtime server modes
+
+Optional backends are installed with extras:
+```bash
+pip install "speech-to-speech[kokoro]"
+pip install "speech-to-speech[pocket]"
+pip install "speech-to-speech[faster-whisper]"
+pip install "speech-to-speech[paraformer]"
+pip install "speech-to-speech[mlx-lm]"
+pip install "speech-to-speech[websocket]"
+```
+
+Deprecated model implementations, including MeloTTS, live in [`archive/`](./archive) and are no longer wired into the CLI.
+
+For development from source:
+```bash
+git clone https://github.com/huggingface/speech-to-speech.git
+cd speech-to-speech
+uv sync
+```
+
+This installs the `speech_to_speech` package in editable mode and makes the `speech-to-speech` CLI command available. The project uses a single `pyproject.toml` with platform markers, so macOS and non-macOS dependencies are resolved automatically from one file.
+
+**Note on DeepFilterNet:** DeepFilterNet (used for optional audio enhancement in VAD) requires `numpy<2` and conflicts with Pocket TTS, which requires `numpy>=2`. Install DeepFilterNet manually only in environments where you are not using Pocket TTS.
+
+
+## Usage
+
+The default CLI is equivalent to a realtime Parakeet + OpenAI-compatible LLM + Qwen3-TTS setup. It uses `OPENAI_API_KEY` from the environment unless `--open_api_api_key` is provided:
+```bash
+speech-to-speech
+```
+
+The pipeline can be run in four ways:
+- **Realtime approach**: Models run locally or on a server, and an OpenAI Realtime-compatible WebSocket API is exposed for another app.
+- **Server/Client approach**: Models run on a server, and audio input/output are streamed from a client using TCP sockets.
+- **WebSocket approach**: Models run on a server, and audio input/output are streamed from a client using WebSockets.
+- **Local approach**: Runs locally.
+
+### Recommended setup 
+
+### Realtime Approach
+
+The default realtime setup uses `--llm open_api`, so it needs an OpenAI API key. Export `OPENAI_API_KEY` before launching, or pass `--open_api_api_key` explicitly. For a deployed OpenAI-compatible LLM, also set `--open_api_base_url`.
+
+```bash
+export OPENAI_API_KEY=...
+```
+
+The default mode starts the OpenAI Realtime-compatible server:
+```bash
+speech-to-speech
+```
+
+This is equivalent to:
+```bash
+speech-to-speech \
+    --thresh 0.6 \
+    --stt parakeet-tdt \
+    --llm open_api \
+    --tts qwen3 \
+    --qwen3_tts_model_name Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
+    --qwen3_tts_speaker Aiden \
+    --qwen3_tts_language auto \
+    --qwen3_tts_non_streaming_mode True \
+    --qwen3_tts_mlx_quantization 6bit \
+    --open_api_model_name gpt-5.4-mini \
+    --open_api_chat_size 30 \
+    --open_api_stream \
+    --enable_live_transcription \
+    --mode realtime
+```
+
+### Server/Client Approach
+
+1. Run the pipeline on the server:
+   ```bash
+   speech-to-speech --recv_host 0.0.0.0 --send_host 0.0.0.0
+   ```
+
+2. Run the client locally to handle microphone input and receive generated audio:
+   ```bash
+   python scripts/listen_and_play.py --host <IP address of your server>
+   ```
+
+### WebSocket Approach
+
+1. Run the pipeline with WebSocket mode:
+   ```bash
+   speech-to-speech --mode websocket --ws_host 0.0.0.0 --ws_port 8765
+   ```
+
+2. Connect to the WebSocket server from your client application at `ws://<server-ip>:8765`. The server handles bidirectional audio streaming:
+   - Send raw audio bytes to the server (16kHz, int16, mono)
+   - Receive generated audio bytes from the server
+
+### Local Approach (Mac)
+
+1. For optimal settings on Mac:
+   ```bash
+   speech-to-speech --local_mac_optimal_settings
+   ```
+
+   You can also specify a particular LLM model:
+   ```bash
+   speech-to-speech \
+       --local_mac_optimal_settings \
+       --lm_model_name mlx-community/Qwen3-4B-Instruct-2507-bf16
+   ```
+
+This setting:
+   - Adds `--device mps` to use MPS for all models.
+   - Sets [Parakeet TDT](https://huggingface.co/nvidia/parakeet-tdt-1.1b) for STT (fast streaming ASR on Apple Silicon)
+   - Sets MLX LM for the language model (uses `--lm_model_name` to specify the model)
+   - Sets Qwen3-TTS for TTS
+   - `--tts pocket` and `--tts kokoro` are also valid TTS options on macOS.
+   - Qwen3 on Apple Silicon uses `mlx-audio` and defaults to the `6bit` MLX variant unless you explicitly select a different quantization or model suffix.
+   - To compare the MLX variants locally, run:
+     ```bash
+     python scripts/benchmark_tts.py \
+         --handlers qwen3 \
+         --iterations 3 \
+         --qwen3_mlx_quantizations bf16 4bit 6bit 8bit
+     ```
+
+### Docker Server
+
+#### Install the NVIDIA Container Toolkit
+
+https://docs.nvidia.com/datacenter/cloud-native/container-toolkit/latest/install-guide.html
+
+#### Start the docker container
+```docker compose up```
+
+
+
+### Recommended usage with Cuda
+
+Leverage Torch Compile for Whisper with Pocket TTS for a simple low-latency setup:
+
+```bash
+speech-to-speech \
+	--lm_model_name microsoft/Phi-3-mini-4k-instruct \
+	--stt_compile_mode reduce-overhead \
+  --tts pocket \
+  --recv_host 0.0.0.0 \
+	--send_host 0.0.0.0 
+```
+
+### Multi-language Support
+
+The pipeline currently supports English, French, Spanish, Chinese, Japanese, and Korean.  
+Two use cases are considered:
+
+- **Single-language conversation**: Enforce the language setting using the `--language` flag, specifying the target language code (default is 'en').
+- **Language switching**: Set `--language` to 'auto'. The STT detects the language of each spoken prompt and forwards it to the LLM. Optionally, opt in with `--lm_enable_lang_prompt` (or `--open_api_enable_lang_prompt` for the OpenAI-compatible backend) to also append a "`Please reply to my message in ...`" instruction so the LLM replies in the detected language. Both flags default to `False` — large LLMs usually pick up the language from context on their own, but the explicit instruction can help smaller models stay in the right language.
+
+Please note that you must use STT and LLM checkpoints compatible with the target language(s). For multilingual TTS, use ChatTTS or another backend that supports the target language.
+
+#### With the server version:
+
+For automatic language detection:
+
+```bash
+speech-to-speech \
+    --stt whisper-mlx \
+    --stt_model_name large-v3 \
+    --language auto \
+    --llm mlx-lm \
+    --lm_model_name mlx-community/Qwen3-4B-Instruct-2507-bf16
+```
+
+Or for one language in particular, chinese in this example
+
+```bash
+speech-to-speech \
+    --stt whisper-mlx \
+    --stt_model_name large-v3 \
+    --language zh \
+    --llm mlx-lm \
+    --lm_model_name mlx-community/Qwen3-4B-Instruct-2507-bf16
+```
+
+#### Local Mac Setup
+
+For automatic language detection (note: `--stt whisper-mlx` overrides the default parakeet-tdt from optimal settings, since Whisper `large-v3` has broader language coverage):
+
+```bash
+speech-to-speech \
+    --local_mac_optimal_settings \
+    --stt whisper-mlx \
+    --stt_model_name large-v3 \
+    --language auto \
+    --lm_model_name mlx-community/Qwen3-4B-Instruct-2507-bf16
+```
+
+Or for one language in particular, chinese in this example
+
+```bash
+speech-to-speech \
+    --local_mac_optimal_settings \
+    --stt whisper-mlx \
+    --stt_model_name large-v3 \
+    --language zh \
+    --lm_model_name mlx-community/Qwen3-4B-Instruct-2507-bf16
+```
+
+### Using Pocket TTS
+
+Pocket TTS from Kyutai Labs provides streaming TTS with voice cloning capabilities. To use it:
+
+```bash
+speech-to-speech \
+    --tts pocket \
+    --pocket_tts_voice jean \
+    --pocket_tts_device cpu
+```
+
+Available voice presets: `alba`, `marius`, `javert`, `jean`, `fantine`, `cosette`, `eponine`, `azelma`. You can also use custom voice files or HuggingFace paths.
+
+## Command-line Usage
+
+> **_NOTE:_** References for all the CLI arguments can be found directly in the [arguments classes](./src/speech_to_speech/arguments_classes) or by running `speech-to-speech -h`.
+
+### Module level Parameters 
+See [ModuleArguments](./src/speech_to_speech/arguments_classes/module_arguments.py) class. Allows to set:
+- a common `--device` (if one wants each part to run on the same device)
+- `--mode` `local` or `server`
+- chosen STT implementation 
+- chosen LM implementation
+- chose TTS implementation
+- logging level
+
+### VAD parameters
+See [VADHandlerArguments](./src/speech_to_speech/arguments_classes/vad_arguments.py) class. Notably:
+- `--thresh`: Threshold value to trigger voice activity detection.
+- `--min_speech_ms`: Minimum duration of detected voice activity to be considered speech.
+- `--min_silence_ms`: Minimum length of silence intervals for segmenting speech, balancing sentence cutting and latency reduction.
+
+
+### STT, LM and TTS parameters
+
+`model_name`, `torch_dtype`, and `device` are exposed for each implementation of the Speech to Text, Language Model, and Text to Speech. Specify the targeted pipeline part with the corresponding prefix (e.g. `stt`, `lm` or `tts`, check the implementations' [arguments classes](./src/speech_to_speech/arguments_classes) for more details).
+
+For example:
+```bash
+--lm_model_name google/gemma-2b-it
+```
+
+### Generation parameters
+
+Other generation parameters of the model's generate method can be set using the part's prefix + `_gen_`, e.g., `--stt_gen_max_new_tokens 128`. These parameters can be added to the pipeline part's arguments class if not already exposed.
+
+## Citations
+
+### Silero VAD
+```bibtex
+@misc{Silero VAD,
+  author = {Silero Team},
+  title = {Silero VAD: pre-trained enterprise-grade Voice Activity Detector (VAD), Number Detector and Language Classifier},
+  year = {2021},
+  publisher = {GitHub},
+  journal = {GitHub repository},
+  howpublished = {\url{https://github.com/snakers4/silero-vad}},
+  commit = {insert_some_commit_here},
+  email = {hello@silero.ai}
+}
+```
+
+### Distil-Whisper
+```bibtex
+@misc{gandhi2023distilwhisper,
+      title={Distil-Whisper: Robust Knowledge Distillation via Large-Scale Pseudo Labelling},
+      author={Sanchit Gandhi and Patrick von Platen and Alexander M. Rush},
+      year={2023},
+      eprint={2311.00430},
+      archivePrefix={arXiv},
+      primaryClass={cs.CL}
+}
+```
+
+### Parler-TTS
+```bibtex
+@misc{lacombe-etal-2024-parler-tts,
+  author = {Yoach Lacombe and Vaibhav Srivastav and Sanchit Gandhi},
+  title = {Parler-TTS},
+  year = {2024},
+  publisher = {GitHub},
+  journal = {GitHub repository},
+  howpublished = {\url{https://github.com/huggingface/parler-tts}}
+}
+```