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Updating XTTS docs
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@ -39,6 +39,10 @@ You can also mail us at info@coqui.ai.
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#### 🐸TTS API
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##### Single reference
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Splits the text into sentences and generates audio for each sentence. The audio files are then concatenated to produce the final audio.
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You can optionally disable sentence splitting for better coherence but more VRAM and possibly hitting models context length limit.
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```python
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from TTS.api import TTS
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tts = TTS("tts_models/multilingual/multi-dataset/xtts_v2", gpu=True)
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@ -47,14 +51,29 @@ tts = TTS("tts_models/multilingual/multi-dataset/xtts_v2", gpu=True)
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tts.tts_to_file(text="It took me quite a long time to develop a voice, and now that I have it I'm not going to be silent.",
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file_path="output.wav",
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speaker_wav=["/path/to/target/speaker.wav"],
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language="en")
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language="en",
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split_sentences=True
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)
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```
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##### Multiple references
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You can pass multiple audio files to the `speaker_wav` argument for better voice cloning.
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```python
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from TTS.api import TTS
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# using the default version set in 🐸TTS
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tts = TTS("tts_models/multilingual/multi-dataset/xtts_v2", gpu=True)
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# using a specific version
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# 👀 see the branch names for versions on https://huggingface.co/coqui/XTTS-v2/tree/main
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# ❗some versions might be incompatible with the API
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tts = TTS("xtts_v2.0.2", gpu=True)
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# getting the latest XTTS_v2
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tts = TTS("xtts", gpu=True)
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# generate speech by cloning a voice using default settings
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tts.tts_to_file(text="It took me quite a long time to develop a voice, and now that I have it I'm not going to be silent.",
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file_path="output.wav",
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@ -62,6 +81,42 @@ tts.tts_to_file(text="It took me quite a long time to develop a voice, and now t
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language="en")
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```
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##### Streaming inference
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XTTS supports streaming inference. This is useful for real-time applications.
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```python
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import os
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import time
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import torch
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import torchaudio
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print("Loading model...")
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tts = TTS("tts_models/multilingual/multi-dataset/xtts_v2", gpu=True)
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model = tts.synthesizer.tts_model
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print("Computing speaker latents...")
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gpt_cond_latent, speaker_embedding = model.get_conditioning_latents(audio_path=["reference.wav"])
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print("Inference...")
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t0 = time.time()
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stream_generator = model.inference_stream(
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"It took me quite a long time to develop a voice and now that I have it I am not going to be silent.",
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"en",
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gpt_cond_latent,
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speaker_embedding
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)
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wav_chuncks = []
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for i, chunk in enumerate(stream_generator):
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if i == 0:
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print(f"Time to first chunck: {time.time() - t0}")
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print(f"Received chunk {i} of audio length {chunk.shape[-1]}")
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wav_chuncks.append(chunk)
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wav = torch.cat(wav_chuncks, dim=0)
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torchaudio.save("xtts_streaming.wav", wav.squeeze().unsqueeze(0).cpu(), 24000)
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```
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#### 🐸TTS Command line
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##### Single reference
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@ -91,10 +146,13 @@ or for all wav files in a directory you can use:
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--use_cuda true
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```
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#### 🐸TTS Model API
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#### model directly
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To use the model API, you need to download the model files and pass config and model file paths manually.
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If you want to be able to run with `use_deepspeed=True` and enjoy the speedup, you need to install deepspeed first.
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##### Calling manually
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If you want to be able to run with `use_deepspeed=True` and **enjoy the speedup**, you need to install deepspeed first.
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```console
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pip install deepspeed==0.10.3
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@ -129,7 +187,7 @@ torchaudio.save("xtts.wav", torch.tensor(out["wav"]).unsqueeze(0), 24000)
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```
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#### streaming inference
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##### Streaming manually
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Here the goal is to stream the audio as it is being generated. This is useful for real-time applications.
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Streaming inference is typically slower than regular inference, but it allows to get a first chunk of audio faster.
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