RealTimeTTS

Name	RealTimeTTS JSON
Version	0.3.44 JSON
	download
home_page	https://github.com/KoljaB/RealTimeTTS
Summary	*Stream text into audio with an easy-to-use, highly configurable library delivering voice output with minimal latency.
upload_time	2024-04-08 00:17:53
maintainer	None
docs_url	None
author	Kolja Beigel
requires_python	<3.12,>=3.9
license	None
keywords	real-time text-to-speech tts streaming audio voice synthesis sentence-segmentation low-latency character-streaming dynamic feedback audio-output text-input tts-engine audio-playback stream-player sentence-fragment audio-feedback interactive python
VCS
bugtrack_url
requirements	No requirements were recorded.
Travis-CI	No Travis.
coveralls test coverage	No coveralls.

            # RealtimeTTS

*Easy to use, low-latency text-to-speech library for realtime applications*

## About the Project

RealtimeTTS is a state-of-the-art text-to-speech (TTS) library designed for real-time applications. It stands out in its ability to convert text streams fast into high-quality auditory output with minimal latency. 

https://github.com/KoljaB/RealtimeTTS/assets/7604638/87dcd9a5-3a4e-4f57-be45-837fc63237e7

## Key Features

- **Low Latency**
  - almost instantaneous text-to-speech conversion
  - compatible with LLM outputs
- **High-Quality Audio**
  - generates clear and natural-sounding speech
- **Multiple TTS Engine Support**
  - supports OpenAI TTS, Elevenlabs, Azure Speech Services, Coqui TTS and System TTS
- **Multilingual**
- **Robust and Reliable**: 
  - ensures continuous operation with a fallback mechanism
  - switches to alternative engines in case of disruptions guaranteeing consistent performance and reliability, which is vital for critical and professional use cases

> **Hint**: *check out [RealtimeSTT](https://github.com/KoljaB/RealtimeSTT), the input counterpart of this library, for speech-to-text capabilities. Together, they form a powerful realtime audio wrapper around large language models.*

## Updates

Latest Version: v0.3.44

See [release history](https://github.com/KoljaB/RealtimeTTS/releases).

## Tech Stack

This library uses:

- **Text-to-Speech Engines**
  - **OpenAIEngine**: OpenAI's TTS system offers 6 natural sounding voices.
  - **CoquiEngine**: High quality local neural TTS.
  - **AzureEngine**: Microsoft's leading TTS technology. 250000 chars free per month.
  - **ElevenlabsEngine**: Offer the best sounding voices available.
  - **SystemEngine**: Native engine for quick setup.

- **Sentence Boundary Detection**
  - **NLTK Sentence Tokenizer**: Uses the Natural Language Toolkit's sentence tokenizer for precise and efficient sentence segmentation.

*By using "industry standard" components RealtimeTTS offers a reliable, high-end technological foundation for developing advanced voice solutions.*

## Installation

Simple installation:

```bash
pip install RealtimeTTS
```

This will install all the necessary dependencies, including a **CPU support only** version of PyTorch (needed for Coqui engine)  

Installation into virtual environment with GPU support:

```bash
python -m venv env_realtimetts
env_realtimetts\Scripts\activate.bat
python.exe -m pip install --upgrade pip
pip install RealtimeTTS
pip install torch==2.2.1+cu118 torchaudio==2.2.1 --index-url https://download.pytorch.org/whl/cu118
```

More information about [CUDA installation](#cuda-installation).

## Engine Requirements

Different engines supported by RealtimeTTS have unique requirements. Ensure you fulfill these requirements based on the engine you choose.

### SystemEngine
The `SystemEngine` works out of the box using your system's built-in TTS capabilities. No additional setup is needed.

### OpenAIEingine
To use the `OpenAIEngine`:
- set environment variable OPENAI_API_KEY
- install ffmpeg (see [CUDA installation](#cuda-installation) point 3)

### AzureEngine
To use the `AzureEngine`, you will need:
- Microsoft Azure Text-to-Speech API key (provided via AzureEngine constructor parameter "speech_key" or in the environment variable AZURE_SPEECH_KEY)
- Microsoft Azure service region.

Make sure you have these credentials available and correctly configured when initializing the `AzureEngine`.

### ElevenlabsEngine
For the `ElevenlabsEngine`, you need:
- Elevenlabs API key (provided via ElevenlabsEngine constructor parameter "api_key" or in the environment variable ELEVENLABS_API_KEY)
- `mpv` installed on your system (essential for streaming mpeg audio, Elevenlabs only delivers mpeg).

  🔹 **Installing `mpv`:**
  - **macOS**:
    ```bash
    brew install mpv
    ```

  - **Linux and Windows**: Visit [mpv.io](https://mpv.io/) for installation instructions.

### CoquiEngine

Delivers high quality, local, neural TTS with voice-cloning.  

Downloads a neural TTS model first. In most cases it be fast enought for Realtime using GPU synthesis. Needs around 4-5 GB VRAM.

- to clone a voice submit the filename of a wave file containing the source voice as "voice" parameter to the CoquiEngine constructor
- in my experience voice cloning works best with a 22050 Hz mono 16bit WAV file containing a short (~10-30 sec) sample 

On most systems GPU support will be needed to run fast enough for realtime, otherwise you will experience stuttering.

## Quick Start

Here's a basic usage example:

```python
from RealtimeTTS import TextToAudioStream, SystemEngine, AzureEngine, ElevenlabsEngine

engine = SystemEngine() # replace with your TTS engine
stream = TextToAudioStream(engine)
stream.feed("Hello world! How are you today?")
stream.play_async()
```

## Feed Text

You can feed individual strings:

```python
stream.feed("Hello, this is a sentence.")
```

Or you can feed generators and character iterators for real-time streaming:

```python
def write(prompt: str):
    for chunk in openai.ChatCompletion.create(
        model="gpt-3.5-turbo",
        messages=[{"role": "user", "content" : prompt}],
        stream=True
    ):
        if (text_chunk := chunk["choices"][0]["delta"].get("content")) is not None:
            yield text_chunk

text_stream = write("A three-sentence relaxing speech.")

stream.feed(text_stream)
```

```python
char_iterator = iter("Streaming this character by character.")
stream.feed(char_iterator)
```

## Playback

Asynchronously:

```python
stream.play_async()
while stream.is_playing():
    time.sleep(0.1)
```

Synchronously:

```python
stream.play()
```

## Testing the Library

The test subdirectory contains a set of scripts to help you evaluate and understand the capabilities of the RealtimeTTS library.

Note that most of the tests still rely on the "old" OpenAI API (<1.0.0). Usage of the new OpenAI API is demonstrated in openai_1.0_test.py.

- **simple_test.py**
    - **Description**: A "hello world" styled demonstration of the library's simplest usage.

- **complex_test.py**
    - **Description**: A comprehensive demonstration showcasing most of the features provided by the library.

- **coqui_test.py**
    - **Description**: Test of local coqui TTS engine.

- **translator.py**
    - **Dependencies**: Run `pip install openai realtimestt`.
    - **Description**: Real-time translations into six different languages.

- **openai_voice_interface.py**
    - **Dependencies**: Run `pip install openai realtimestt`.
    - **Description**: Wake word activated and voice based user interface to the OpenAI API.

- **advanced_talk.py**
    - **Dependencies**: Run `pip install openai keyboard realtimestt`.
    - **Description**: Choose TTS engine and voice before starting AI conversation.

- **minimalistic_talkbot.py**
    - **Dependencies**: Run `pip install openai realtimestt`.
    - **Description**: A basic talkbot in 20 lines of code.
    
- **simple_llm_test.py**
    - **Dependencies**: Run `pip install openai`.
    - **Description**: Simple demonstration how to integrate the library with large language models (LLMs).

- **test_callbacks.py**
    - **Dependencies**: Run `pip install openai`.
    - **Description**: Showcases the callbacks and lets you check the latency times in a real-world application environment.

## Pause, Resume & Stop

Pause the audio stream:

```python
stream.pause()
```

Resume a paused stream:

```python
stream.resume()
```

Stop the stream immediately:

```python
stream.stop()
```

## Requirements Explained

- **Python Version**: 
  - **Required**: Python >= 3.9, < 3.12
  - **Reason**: The library depends on the GitHub library "TTS" from coqui, which requires Python versions in this range.

- **requests (>=2.31.0)**: to send HTTP requests for API calls and voice list retrieval
  
- **PyAudio (>=0.2.13)**: to create an output audio stream
  
- **stream2sentence (>=0.1.1)**: to split the incoming text stream into sentences 

- **pyttsx3 (>=2.90)**: System text-to-speech conversion engine

- **azure-cognitiveservices-speech (>=1.31.0)**: Azure text-to-speech conversion engine
  
- **elevenlabs (>=0.2.24)**: Elevenlabs text-to-speech conversion engine


## Configuration

### Initialization Parameters for `TextToAudioStream`

When you initialize the `TextToAudioStream` class, you have various options to customize its behavior. Here are the available parameters:

#### `engine` (BaseEngine)
- **Type**: BaseEngine
- **Required**: Yes
- **Description**: The underlying engine responsible for text-to-audio synthesis. You must provide an instance of `BaseEngine` or its subclass to enable audio synthesis.

#### `on_text_stream_start` (callable)
- **Type**: Callable function
- **Required**: No
- **Description**: This optional callback function is triggered when the text stream begins. Use it for any setup or logging you may need.

#### `on_text_stream_stop` (callable)
- **Type**: Callable function
- **Required**: No
- **Description**: This optional callback function is activated when the text stream ends. You can use this for cleanup tasks or logging.

#### `on_audio_stream_start` (callable)
- **Type**: Callable function
- **Required**: No
- **Description**: This optional callback function is invoked when the audio stream starts. Useful for UI updates or event logging.

#### `on_audio_stream_stop` (callable)
- **Type**: Callable function
- **Required**: No
- **Description**: This optional callback function is called when the audio stream stops. Ideal for resource cleanup or post-processing tasks.

#### `on_character` (callable)
- **Type**: Callable function
- **Required**: No
- **Description**: This optional callback function is called when a single character is processed.

#### `output_device_index` (int)
- **Type**: Integer
- **Required**: No
- **Default**: None
- **Description**: Specifies the output device index to use. None uses the default device.

#### `tokenizer` (string)
- **Type**: String
- **Required**: No
- **Default**: nltk
- **Description**: Tokenizer to use for sentence splitting (currently "nltk" and "stanza" are supported).

#### `language` (string)
- **Type**: String
- **Required**: No
- **Default**: en
- **Description**: Language to use for sentence splitting.

#### `level` (int)
- **Type**: Integer
- **Required**: No
- **Default**: `logging.WARNING`
- **Description**: Sets the logging level for the internal logger. This can be any integer constant from Python's built-in `logging` module.

#### Example Usage:

```python
engine = YourEngine()  # Substitute with your engine
stream = TextToAudioStream(
    engine=engine,
    on_text_stream_start=my_text_start_func,
    on_text_stream_stop=my_text_stop_func,
    on_audio_stream_start=my_audio_start_func,
    on_audio_stream_stop=my_audio_stop_func,
    level=logging.INFO
)
```

### Methods

#### `play` and `play_async`

These methods are responsible for executing the text-to-audio synthesis and playing the audio stream. The difference is that `play` is a blocking function, while `play_async` runs in a separate thread, allowing other operations to proceed.

##### `fast_sentence_fragment` (bool)
- **Default**: `False`
- **Description**: When set to `True`, the method will prioritize speed, generating and playing sentence fragments faster. This is useful for applications where latency matters.

##### `buffer_threshold_seconds` (float)
- **Default**: `2.0`
- **Description**: Specifies the time in seconds for the buffering threshold, which impacts the smoothness and continuity of audio playback. 

  - **How it Works**: Before synthesizing a new sentence, the system checks if there is more audio material left in the buffer than the time specified by `buffer_threshold_seconds`. If so, it retrieves another sentence from the text generator, assuming that it can fetch and synthesize this new sentence within the time window provided by the remaining audio in the buffer. This process allows the text-to-speech engine to have more context for better synthesis, enhancing the user experience.

  A higher value ensures that there's more pre-buffered audio, reducing the likelihood of silence or gaps during playback. If you experience breaks or pauses, consider increasing this value.

- **Hint**: If you experience silence or breaks between sentences, consider raising this value to ensure smoother playback.

##### `minimum_sentence_length` (int)
- **Default**: `3`
- **Description**: Sets the minimum character length to consider a string as a sentence to be synthesized. This affects how text chunks are processed and played.

##### `log_characters` (bool)
- **Default**: `False`
- **Description**: Enable this to log the individual characters that are being processed for synthesis.

##### `log_synthesized_text` (bool)
- **Default**: `False`
- **Description**: When enabled, logs the text chunks as they are synthesized into audio. Helpful for auditing and debugging.

By understanding and setting these parameters and methods appropriately, you can tailor the `TextToAudioStream` to meet the specific needs of your application.


### CUDA installation

These steps are recommended for those who require **better performance** and have a compatible NVIDIA GPU.

> **Note**: *to check if your NVIDIA GPU supports CUDA, visit the [official CUDA GPUs list](https://developer.nvidia.com/cuda-gpus).*

To use torch with support via CUDA please follow these steps:

> **Note**: *newer pytorch installations [may](https://stackoverflow.com/a/77069523) (unverified) not need Toolkit (and possibly cuDNN) installation anymore.*

1. **Install NVIDIA CUDA Toolkit**:
    For example, to install Toolkit 11.8 please  
    - Visit [NVIDIA CUDA Toolkit Archive](https://developer.nvidia.com/cuda-11-8-0-download-archive).
    - Select version 11.
    - Download and install the software.

2. **Install NVIDIA cuDNN**:
    For example, to install cuDNN 8.7.0 for CUDA 11.x please  
    - Visit [NVIDIA cuDNN Archive](https://developer.nvidia.com/rdp/cudnn-archive).
    - Click on "Download cuDNN v8.7.0 (November 28th, 2022), for CUDA 11.x".
    - Download and install the software.

3. **Install ffmpeg**:

    You can download an installer for your OS from the [ffmpeg Website](https://ffmpeg.org/download.html).  
    
    Or use a package manager:

    - **On Ubuntu or Debian**:
        ```bash
        sudo apt update && sudo apt install ffmpeg
        ```

    - **On Arch Linux**:
        ```bash
        sudo pacman -S ffmpeg
        ```

    - **On MacOS using Homebrew** ([https://brew.sh/](https://brew.sh/)):
        ```bash
        brew install ffmpeg
        ```

    - **On Windows using Chocolatey** ([https://chocolatey.org/](https://chocolatey.org/)):
        ```bash
        choco install ffmpeg
        ```

    - **On Windows using Scoop** ([https://scoop.sh/](https://scoop.sh/)):
        ```bash
        scoop install ffmpeg
        ```    

4. **Install PyTorch with CUDA support**:
    ```bash
    pip install torch==2.1.1+cu118 torchaudio==2.1.1+cu118 --index-url https://download.pytorch.org/whl/cu118
    ```

5. **Fix for to resolve compatility issues**:
    If you run into library compatility issues, try setting these libraries to fixed versions:

    ```bash
    pip install networkx==2.8.8
    pip install typing_extensions==4.8.0
    pip install fsspec==2023.6.0
    pip install imageio==2.31.6
    pip install networkx==2.8.8
    pip install numpy==1.24.3
    pip install requests==2.31.0
    ```

## 💖 Acknowledgements

Huge shoutout to the team behind [Coqui AI](https://coqui.ai/) being the first giving us local high quality synthesis with realtime speed and even a clonable voice!

## Contribution

Contributions are always welcome (e.g. PR to add a new engine).

## License Information

### ❗ Important Note:
While the source of this library is open-source, the usage of many of the engines it depends on are not: External engine providers often restrict commercial use in their free plans. This means the engines can be used for noncommercial projects, but commercial usage requires a paid plan.

### Engine Licenses Summary:

#### CoquiEngine
- **License**: Open-source only for noncommercial projects.
- **Commercial Use**: Requires a paid plan.
- **Details**: [CoquiEngine License](https://coqui.ai/cpml)

#### ElevenlabsEngine
- **License**: Open-source only for noncommercial projects.
- **Commercial Use**: Available with every paid plan.
- **Details**: [ElevenlabsEngine License](https://help.elevenlabs.io/hc/en-us/articles/13313564601361-Can-I-publish-the-content-I-generate-on-the-platform-)

#### AzureEngine
- **License**: Open-source only for noncommercial projects.
- **Commercial Use**: Available from the standard tier upwards.
- **Details**: [AzureEngine License](https://learn.microsoft.com/en-us/answers/questions/1192398/can-i-use-azure-text-to-speech-for-commercial-usag)

#### SystemEngine
- **License**: Mozilla Public License 2.0 and GNU Lesser General Public License (LGPL) version 3.0.
- **Commercial Use**: Allowed under this license.
- **Details**: [SystemEngine License](https://github.com/nateshmbhat/pyttsx3/blob/master/LICENSE)

#### OpenAIEngine
- **License**: please read [OpenAI Terms of Use](https://openai.com/policies/terms-of-use)

**Disclaimer**: This is a summarization of the licenses as understood at the time of writing. It is not legal advice. Please read and respect the licenses of the different engine providers yourself if you plan to use them in a project.

## Author

Kolja Beigel  
Email: kolja.beigel@web.de  
[GitHub](https://github.com/KoljaB/RealtimeTTS)

            

Raw data

            {
    "_id": null,
    "home_page": "https://github.com/KoljaB/RealTimeTTS",
    "name": "RealTimeTTS",
    "maintainer": null,
    "docs_url": null,
    "requires_python": "<3.12,>=3.9",
    "maintainer_email": null,
    "keywords": "real-time, text-to-speech, TTS, streaming, audio, voice, synthesis, sentence-segmentation, low-latency, character-streaming, dynamic feedback, audio-output, text-input, TTS-engine, audio-playback, stream-player, sentence-fragment, audio-feedback, interactive, python",
    "author": "Kolja Beigel",
    "author_email": "kolja.beigel@web.de",
    "download_url": "https://files.pythonhosted.org/packages/ae/0d/5d5bf4e03a44f0ffae4eee39622646343adf6ab0c8f27b52c489fa95cdec/RealTimeTTS-0.3.44.tar.gz",
    "platform": null,
    "description": "# RealtimeTTS\r\n\r\n*Easy to use, low-latency text-to-speech library for realtime applications*\r\n\r\n## About the Project\r\n\r\nRealtimeTTS is a state-of-the-art text-to-speech (TTS) library designed for real-time applications. It stands out in its ability to convert text streams fast into high-quality auditory output with minimal latency. \r\n\r\nhttps://github.com/KoljaB/RealtimeTTS/assets/7604638/87dcd9a5-3a4e-4f57-be45-837fc63237e7\r\n\r\n## Key Features\r\n\r\n- **Low Latency**\r\n  - almost instantaneous text-to-speech conversion\r\n  - compatible with LLM outputs\r\n- **High-Quality Audio**\r\n  - generates clear and natural-sounding speech\r\n- **Multiple TTS Engine Support**\r\n  - supports OpenAI TTS, Elevenlabs, Azure Speech Services, Coqui TTS and System TTS\r\n- **Multilingual**\r\n- **Robust and Reliable**: \r\n  - ensures continuous operation with a fallback mechanism\r\n  - switches to alternative engines in case of disruptions guaranteeing consistent performance and reliability, which is vital for critical and professional use cases\r\n\r\n> **Hint**: *check out [RealtimeSTT](https://github.com/KoljaB/RealtimeSTT), the input counterpart of this library, for speech-to-text capabilities. Together, they form a powerful realtime audio wrapper around large language models.*\r\n\r\n## Updates\r\n\r\nLatest Version: v0.3.44\r\n\r\nSee [release history](https://github.com/KoljaB/RealtimeTTS/releases).\r\n\r\n## Tech Stack\r\n\r\nThis library uses:\r\n\r\n- **Text-to-Speech Engines**\r\n  - **OpenAIEngine**: OpenAI's TTS system offers 6 natural sounding voices.\r\n  - **CoquiEngine**: High quality local neural TTS.\r\n  - **AzureEngine**: Microsoft's leading TTS technology. 250000 chars free per month.\r\n  - **ElevenlabsEngine**: Offer the best sounding voices available.\r\n  - **SystemEngine**: Native engine for quick setup.\r\n\r\n- **Sentence Boundary Detection**\r\n  - **NLTK Sentence Tokenizer**: Uses the Natural Language Toolkit's sentence tokenizer for precise and efficient sentence segmentation.\r\n\r\n*By using \"industry standard\" components RealtimeTTS offers a reliable, high-end technological foundation for developing advanced voice solutions.*\r\n\r\n## Installation\r\n\r\nSimple installation:\r\n\r\n```bash\r\npip install RealtimeTTS\r\n```\r\n\r\nThis will install all the necessary dependencies, including a **CPU support only** version of PyTorch (needed for Coqui engine)  \r\n\r\nInstallation into virtual environment with GPU support:\r\n\r\n```bash\r\npython -m venv env_realtimetts\r\nenv_realtimetts\\Scripts\\activate.bat\r\npython.exe -m pip install --upgrade pip\r\npip install RealtimeTTS\r\npip install torch==2.2.1+cu118 torchaudio==2.2.1 --index-url https://download.pytorch.org/whl/cu118\r\n```\r\n\r\nMore information about [CUDA installation](#cuda-installation).\r\n\r\n## Engine Requirements\r\n\r\nDifferent engines supported by RealtimeTTS have unique requirements. Ensure you fulfill these requirements based on the engine you choose.\r\n\r\n### SystemEngine\r\nThe `SystemEngine` works out of the box using your system's built-in TTS capabilities. No additional setup is needed.\r\n\r\n### OpenAIEingine\r\nTo use the `OpenAIEngine`:\r\n- set environment variable OPENAI_API_KEY\r\n- install ffmpeg (see [CUDA installation](#cuda-installation) point 3)\r\n\r\n### AzureEngine\r\nTo use the `AzureEngine`, you will need:\r\n- Microsoft Azure Text-to-Speech API key (provided via AzureEngine constructor parameter \"speech_key\" or in the environment variable AZURE_SPEECH_KEY)\r\n- Microsoft Azure service region.\r\n\r\nMake sure you have these credentials available and correctly configured when initializing the `AzureEngine`.\r\n\r\n### ElevenlabsEngine\r\nFor the `ElevenlabsEngine`, you need:\r\n- Elevenlabs API key (provided via ElevenlabsEngine constructor parameter \"api_key\" or in the environment variable ELEVENLABS_API_KEY)\r\n- `mpv` installed on your system (essential for streaming mpeg audio, Elevenlabs only delivers mpeg).\r\n\r\n  \ud83d\udd39 **Installing `mpv`:**\r\n  - **macOS**:\r\n    ```bash\r\n    brew install mpv\r\n    ```\r\n\r\n  - **Linux and Windows**: Visit [mpv.io](https://mpv.io/) for installation instructions.\r\n\r\n### CoquiEngine\r\n\r\nDelivers high quality, local, neural TTS with voice-cloning.  \r\n\r\nDownloads a neural TTS model first. In most cases it be fast enought for Realtime using GPU synthesis. Needs around 4-5 GB VRAM.\r\n\r\n- to clone a voice submit the filename of a wave file containing the source voice as \"voice\" parameter to the CoquiEngine constructor\r\n- in my experience voice cloning works best with a 22050 Hz mono 16bit WAV file containing a short (~10-30 sec) sample \r\n\r\nOn most systems GPU support will be needed to run fast enough for realtime, otherwise you will experience stuttering.\r\n\r\n## Quick Start\r\n\r\nHere's a basic usage example:\r\n\r\n```python\r\nfrom RealtimeTTS import TextToAudioStream, SystemEngine, AzureEngine, ElevenlabsEngine\r\n\r\nengine = SystemEngine() # replace with your TTS engine\r\nstream = TextToAudioStream(engine)\r\nstream.feed(\"Hello world! How are you today?\")\r\nstream.play_async()\r\n```\r\n\r\n## Feed Text\r\n\r\nYou can feed individual strings:\r\n\r\n```python\r\nstream.feed(\"Hello, this is a sentence.\")\r\n```\r\n\r\nOr you can feed generators and character iterators for real-time streaming:\r\n\r\n```python\r\ndef write(prompt: str):\r\n    for chunk in openai.ChatCompletion.create(\r\n        model=\"gpt-3.5-turbo\",\r\n        messages=[{\"role\": \"user\", \"content\" : prompt}],\r\n        stream=True\r\n    ):\r\n        if (text_chunk := chunk[\"choices\"][0][\"delta\"].get(\"content\")) is not None:\r\n            yield text_chunk\r\n\r\ntext_stream = write(\"A three-sentence relaxing speech.\")\r\n\r\nstream.feed(text_stream)\r\n```\r\n\r\n```python\r\nchar_iterator = iter(\"Streaming this character by character.\")\r\nstream.feed(char_iterator)\r\n```\r\n\r\n## Playback\r\n\r\nAsynchronously:\r\n\r\n```python\r\nstream.play_async()\r\nwhile stream.is_playing():\r\n    time.sleep(0.1)\r\n```\r\n\r\nSynchronously:\r\n\r\n```python\r\nstream.play()\r\n```\r\n\r\n## Testing the Library\r\n\r\nThe test subdirectory contains a set of scripts to help you evaluate and understand the capabilities of the RealtimeTTS library.\r\n\r\nNote that most of the tests still rely on the \"old\" OpenAI API (<1.0.0). Usage of the new OpenAI API is demonstrated in openai_1.0_test.py.\r\n\r\n- **simple_test.py**\r\n    - **Description**: A \"hello world\" styled demonstration of the library's simplest usage.\r\n\r\n- **complex_test.py**\r\n    - **Description**: A comprehensive demonstration showcasing most of the features provided by the library.\r\n\r\n- **coqui_test.py**\r\n    - **Description**: Test of local coqui TTS engine.\r\n\r\n- **translator.py**\r\n    - **Dependencies**: Run `pip install openai realtimestt`.\r\n    - **Description**: Real-time translations into six different languages.\r\n\r\n- **openai_voice_interface.py**\r\n    - **Dependencies**: Run `pip install openai realtimestt`.\r\n    - **Description**: Wake word activated and voice based user interface to the OpenAI API.\r\n\r\n- **advanced_talk.py**\r\n    - **Dependencies**: Run `pip install openai keyboard realtimestt`.\r\n    - **Description**: Choose TTS engine and voice before starting AI conversation.\r\n\r\n- **minimalistic_talkbot.py**\r\n    - **Dependencies**: Run `pip install openai realtimestt`.\r\n    - **Description**: A basic talkbot in 20 lines of code.\r\n    \r\n- **simple_llm_test.py**\r\n    - **Dependencies**: Run `pip install openai`.\r\n    - **Description**: Simple demonstration how to integrate the library with large language models (LLMs).\r\n\r\n- **test_callbacks.py**\r\n    - **Dependencies**: Run `pip install openai`.\r\n    - **Description**: Showcases the callbacks and lets you check the latency times in a real-world application environment.\r\n\r\n## Pause, Resume & Stop\r\n\r\nPause the audio stream:\r\n\r\n```python\r\nstream.pause()\r\n```\r\n\r\nResume a paused stream:\r\n\r\n```python\r\nstream.resume()\r\n```\r\n\r\nStop the stream immediately:\r\n\r\n```python\r\nstream.stop()\r\n```\r\n\r\n## Requirements Explained\r\n\r\n- **Python Version**: \r\n  - **Required**: Python >= 3.9, < 3.12\r\n  - **Reason**: The library depends on the GitHub library \"TTS\" from coqui, which requires Python versions in this range.\r\n\r\n- **requests (>=2.31.0)**: to send HTTP requests for API calls and voice list retrieval\r\n  \r\n- **PyAudio (>=0.2.13)**: to create an output audio stream\r\n  \r\n- **stream2sentence (>=0.1.1)**: to split the incoming text stream into sentences \r\n\r\n- **pyttsx3 (>=2.90)**: System text-to-speech conversion engine\r\n\r\n- **azure-cognitiveservices-speech (>=1.31.0)**: Azure text-to-speech conversion engine\r\n  \r\n- **elevenlabs (>=0.2.24)**: Elevenlabs text-to-speech conversion engine\r\n\r\n\r\n## Configuration\r\n\r\n### Initialization Parameters for `TextToAudioStream`\r\n\r\nWhen you initialize the `TextToAudioStream` class, you have various options to customize its behavior. Here are the available parameters:\r\n\r\n#### `engine` (BaseEngine)\r\n- **Type**: BaseEngine\r\n- **Required**: Yes\r\n- **Description**: The underlying engine responsible for text-to-audio synthesis. You must provide an instance of `BaseEngine` or its subclass to enable audio synthesis.\r\n\r\n#### `on_text_stream_start` (callable)\r\n- **Type**: Callable function\r\n- **Required**: No\r\n- **Description**: This optional callback function is triggered when the text stream begins. Use it for any setup or logging you may need.\r\n\r\n#### `on_text_stream_stop` (callable)\r\n- **Type**: Callable function\r\n- **Required**: No\r\n- **Description**: This optional callback function is activated when the text stream ends. You can use this for cleanup tasks or logging.\r\n\r\n#### `on_audio_stream_start` (callable)\r\n- **Type**: Callable function\r\n- **Required**: No\r\n- **Description**: This optional callback function is invoked when the audio stream starts. Useful for UI updates or event logging.\r\n\r\n#### `on_audio_stream_stop` (callable)\r\n- **Type**: Callable function\r\n- **Required**: No\r\n- **Description**: This optional callback function is called when the audio stream stops. Ideal for resource cleanup or post-processing tasks.\r\n\r\n#### `on_character` (callable)\r\n- **Type**: Callable function\r\n- **Required**: No\r\n- **Description**: This optional callback function is called when a single character is processed.\r\n\r\n#### `output_device_index` (int)\r\n- **Type**: Integer\r\n- **Required**: No\r\n- **Default**: None\r\n- **Description**: Specifies the output device index to use. None uses the default device.\r\n\r\n#### `tokenizer` (string)\r\n- **Type**: String\r\n- **Required**: No\r\n- **Default**: nltk\r\n- **Description**: Tokenizer to use for sentence splitting (currently \"nltk\" and \"stanza\" are supported).\r\n\r\n#### `language` (string)\r\n- **Type**: String\r\n- **Required**: No\r\n- **Default**: en\r\n- **Description**: Language to use for sentence splitting.\r\n\r\n#### `level` (int)\r\n- **Type**: Integer\r\n- **Required**: No\r\n- **Default**: `logging.WARNING`\r\n- **Description**: Sets the logging level for the internal logger. This can be any integer constant from Python's built-in `logging` module.\r\n\r\n#### Example Usage:\r\n\r\n```python\r\nengine = YourEngine()  # Substitute with your engine\r\nstream = TextToAudioStream(\r\n    engine=engine,\r\n    on_text_stream_start=my_text_start_func,\r\n    on_text_stream_stop=my_text_stop_func,\r\n    on_audio_stream_start=my_audio_start_func,\r\n    on_audio_stream_stop=my_audio_stop_func,\r\n    level=logging.INFO\r\n)\r\n```\r\n\r\n### Methods\r\n\r\n#### `play` and `play_async`\r\n\r\nThese methods are responsible for executing the text-to-audio synthesis and playing the audio stream. The difference is that `play` is a blocking function, while `play_async` runs in a separate thread, allowing other operations to proceed.\r\n\r\n##### `fast_sentence_fragment` (bool)\r\n- **Default**: `False`\r\n- **Description**: When set to `True`, the method will prioritize speed, generating and playing sentence fragments faster. This is useful for applications where latency matters.\r\n\r\n##### `buffer_threshold_seconds` (float)\r\n- **Default**: `2.0`\r\n- **Description**: Specifies the time in seconds for the buffering threshold, which impacts the smoothness and continuity of audio playback. \r\n\r\n  - **How it Works**: Before synthesizing a new sentence, the system checks if there is more audio material left in the buffer than the time specified by `buffer_threshold_seconds`. If so, it retrieves another sentence from the text generator, assuming that it can fetch and synthesize this new sentence within the time window provided by the remaining audio in the buffer. This process allows the text-to-speech engine to have more context for better synthesis, enhancing the user experience.\r\n\r\n  A higher value ensures that there's more pre-buffered audio, reducing the likelihood of silence or gaps during playback. If you experience breaks or pauses, consider increasing this value.\r\n\r\n- **Hint**: If you experience silence or breaks between sentences, consider raising this value to ensure smoother playback.\r\n\r\n##### `minimum_sentence_length` (int)\r\n- **Default**: `3`\r\n- **Description**: Sets the minimum character length to consider a string as a sentence to be synthesized. This affects how text chunks are processed and played.\r\n\r\n##### `log_characters` (bool)\r\n- **Default**: `False`\r\n- **Description**: Enable this to log the individual characters that are being processed for synthesis.\r\n\r\n##### `log_synthesized_text` (bool)\r\n- **Default**: `False`\r\n- **Description**: When enabled, logs the text chunks as they are synthesized into audio. Helpful for auditing and debugging.\r\n\r\nBy understanding and setting these parameters and methods appropriately, you can tailor the `TextToAudioStream` to meet the specific needs of your application.\r\n\r\n\r\n### CUDA installation\r\n\r\nThese steps are recommended for those who require **better performance** and have a compatible NVIDIA GPU.\r\n\r\n> **Note**: *to check if your NVIDIA GPU supports CUDA, visit the [official CUDA GPUs list](https://developer.nvidia.com/cuda-gpus).*\r\n\r\nTo use torch with support via CUDA please follow these steps:\r\n\r\n> **Note**: *newer pytorch installations [may](https://stackoverflow.com/a/77069523) (unverified) not need Toolkit (and possibly cuDNN) installation anymore.*\r\n\r\n1. **Install NVIDIA CUDA Toolkit**:\r\n    For example, to install Toolkit 11.8 please  \r\n    - Visit [NVIDIA CUDA Toolkit Archive](https://developer.nvidia.com/cuda-11-8-0-download-archive).\r\n    - Select version 11.\r\n    - Download and install the software.\r\n\r\n2. **Install NVIDIA cuDNN**:\r\n    For example, to install cuDNN 8.7.0 for CUDA 11.x please  \r\n    - Visit [NVIDIA cuDNN Archive](https://developer.nvidia.com/rdp/cudnn-archive).\r\n    - Click on \"Download cuDNN v8.7.0 (November 28th, 2022), for CUDA 11.x\".\r\n    - Download and install the software.\r\n\r\n3. **Install ffmpeg**:\r\n\r\n    You can download an installer for your OS from the [ffmpeg Website](https://ffmpeg.org/download.html).  \r\n    \r\n    Or use a package manager:\r\n\r\n    - **On Ubuntu or Debian**:\r\n        ```bash\r\n        sudo apt update && sudo apt install ffmpeg\r\n        ```\r\n\r\n    - **On Arch Linux**:\r\n        ```bash\r\n        sudo pacman -S ffmpeg\r\n        ```\r\n\r\n    - **On MacOS using Homebrew** ([https://brew.sh/](https://brew.sh/)):\r\n        ```bash\r\n        brew install ffmpeg\r\n        ```\r\n\r\n    - **On Windows using Chocolatey** ([https://chocolatey.org/](https://chocolatey.org/)):\r\n        ```bash\r\n        choco install ffmpeg\r\n        ```\r\n\r\n    - **On Windows using Scoop** ([https://scoop.sh/](https://scoop.sh/)):\r\n        ```bash\r\n        scoop install ffmpeg\r\n        ```    \r\n\r\n4. **Install PyTorch with CUDA support**:\r\n    ```bash\r\n    pip install torch==2.1.1+cu118 torchaudio==2.1.1+cu118 --index-url https://download.pytorch.org/whl/cu118\r\n    ```\r\n\r\n5. **Fix for to resolve compatility issues**:\r\n    If you run into library compatility issues, try setting these libraries to fixed versions:\r\n\r\n    ```bash\r\n    pip install networkx==2.8.8\r\n    pip install typing_extensions==4.8.0\r\n    pip install fsspec==2023.6.0\r\n    pip install imageio==2.31.6\r\n    pip install networkx==2.8.8\r\n    pip install numpy==1.24.3\r\n    pip install requests==2.31.0\r\n    ```\r\n\r\n## \ud83d\udc96 Acknowledgements\r\n\r\nHuge shoutout to the team behind [Coqui AI](https://coqui.ai/) being the first giving us local high quality synthesis with realtime speed and even a clonable voice!\r\n\r\n## Contribution\r\n\r\nContributions are always welcome (e.g. PR to add a new engine).\r\n\r\n## License Information\r\n\r\n### \u2757 Important Note:\r\nWhile the source of this library is open-source, the usage of many of the engines it depends on are not: External engine providers often restrict commercial use in their free plans. This means the engines can be used for noncommercial projects, but commercial usage requires a paid plan.\r\n\r\n### Engine Licenses Summary:\r\n\r\n#### CoquiEngine\r\n- **License**: Open-source only for noncommercial projects.\r\n- **Commercial Use**: Requires a paid plan.\r\n- **Details**: [CoquiEngine License](https://coqui.ai/cpml)\r\n\r\n#### ElevenlabsEngine\r\n- **License**: Open-source only for noncommercial projects.\r\n- **Commercial Use**: Available with every paid plan.\r\n- **Details**: [ElevenlabsEngine License](https://help.elevenlabs.io/hc/en-us/articles/13313564601361-Can-I-publish-the-content-I-generate-on-the-platform-)\r\n\r\n#### AzureEngine\r\n- **License**: Open-source only for noncommercial projects.\r\n- **Commercial Use**: Available from the standard tier upwards.\r\n- **Details**: [AzureEngine License](https://learn.microsoft.com/en-us/answers/questions/1192398/can-i-use-azure-text-to-speech-for-commercial-usag)\r\n\r\n#### SystemEngine\r\n- **License**: Mozilla Public License 2.0 and GNU Lesser General Public License (LGPL) version 3.0.\r\n- **Commercial Use**: Allowed under this license.\r\n- **Details**: [SystemEngine License](https://github.com/nateshmbhat/pyttsx3/blob/master/LICENSE)\r\n\r\n#### OpenAIEngine\r\n- **License**: please read [OpenAI Terms of Use](https://openai.com/policies/terms-of-use)\r\n\r\n**Disclaimer**: This is a summarization of the licenses as understood at the time of writing. It is not legal advice. Please read and respect the licenses of the different engine providers yourself if you plan to use them in a project.\r\n\r\n## Author\r\n\r\nKolja Beigel  \r\nEmail: kolja.beigel@web.de  \r\n[GitHub](https://github.com/KoljaB/RealtimeTTS)\r\n",
    "bugtrack_url": null,
    "license": null,
    "summary": "*Stream text into audio with an easy-to-use, highly configurable library delivering voice output with minimal latency.",
    "version": "0.3.44",
    "project_urls": {
        "Homepage": "https://github.com/KoljaB/RealTimeTTS"
    },
    "split_keywords": [
        "real-time",
        " text-to-speech",
        " tts",
        " streaming",
        " audio",
        " voice",
        " synthesis",
        " sentence-segmentation",
        " low-latency",
        " character-streaming",
        " dynamic feedback",
        " audio-output",
        " text-input",
        " tts-engine",
        " audio-playback",
        " stream-player",
        " sentence-fragment",
        " audio-feedback",
        " interactive",
        " python"
    ],
    "urls": [
        {
            "comment_text": "",
            "digests": {
                "blake2b_256": "7fed26e66fa8a9b85326ac6164c32f03378cb9bee07511d97c8403fef9dcf4ba",
                "md5": "330347b3300d31a02d9d9384a2e76eaa",
                "sha256": "42831aaec096618f9cc4f2542fef29bf92346521978a6dce005bb62b6cbda71f"
            },
            "downloads": -1,
            "filename": "RealTimeTTS-0.3.44-py3-none-any.whl",
            "has_sig": false,
            "md5_digest": "330347b3300d31a02d9d9384a2e76eaa",
            "packagetype": "bdist_wheel",
            "python_version": "py3",
            "requires_python": "<3.12,>=3.9",
            "size": 467880,
            "upload_time": "2024-04-08T00:17:52",
            "upload_time_iso_8601": "2024-04-08T00:17:52.191106Z",
            "url": "https://files.pythonhosted.org/packages/7f/ed/26e66fa8a9b85326ac6164c32f03378cb9bee07511d97c8403fef9dcf4ba/RealTimeTTS-0.3.44-py3-none-any.whl",
            "yanked": false,
            "yanked_reason": null
        },
        {
            "comment_text": "",
            "digests": {
                "blake2b_256": "ae0d5d5bf4e03a44f0ffae4eee39622646343adf6ab0c8f27b52c489fa95cdec",
                "md5": "9bee022e47653c46d302fed03557ab48",
                "sha256": "bfc9f621c621861075daaeb71deca391086409bdf3c5640f605e3f4435fb00dc"
            },
            "downloads": -1,
            "filename": "RealTimeTTS-0.3.44.tar.gz",
            "has_sig": false,
            "md5_digest": "9bee022e47653c46d302fed03557ab48",
            "packagetype": "sdist",
            "python_version": "source",
            "requires_python": "<3.12,>=3.9",
            "size": 464634,
            "upload_time": "2024-04-08T00:17:53",
            "upload_time_iso_8601": "2024-04-08T00:17:53.987461Z",
            "url": "https://files.pythonhosted.org/packages/ae/0d/5d5bf4e03a44f0ffae4eee39622646343adf6ab0c8f27b52c489fa95cdec/RealTimeTTS-0.3.44.tar.gz",
            "yanked": false,
            "yanked_reason": null
        }
    ],
    "upload_time": "2024-04-08 00:17:53",
    "github": true,
    "gitlab": false,
    "bitbucket": false,
    "codeberg": false,
    "github_user": "KoljaB",
    "github_project": "RealTimeTTS",
    "travis_ci": false,
    "coveralls": false,
    "github_actions": false,
    "requirements": [],
    "lcname": "realtimetts"
}