#!/usr/bin/env python3 # # Copyright (c) 2023 Xiaomi Corporation """ This file demonstrates how to use sherpa-onnx Python API to generate audio from text, i.e., text-to-speech. Different from ./offline-tts-play.py, this file does not play back the generated audio. Usage: Example (1/2) wget https://github.com/k2-fsa/sherpa-onnx/releases/download/tts-models/vits-piper-en_US-amy-low.tar.bz2 tar xf vits-piper-en_US-amy-low.tar.bz2 python3 ./python-api-examples/offline-tts.py \ --vits-model=./vits-piper-en_US-amy-low/en_US-amy-low.onnx \ --vits-tokens=./vits-piper-en_US-amy-low/tokens.txt \ --vits-data-dir=./vits-piper-en_US-amy-low/espeak-ng-data \ --output-filename=./generated.wav \ "Today as always, men fall into two groups: slaves and free men. Whoever does not have two-thirds of his day for himself, is a slave, whatever he may be: a statesman, a businessman, an official, or a scholar." Example (2/2) wget https://github.com/k2-fsa/sherpa-onnx/releases/download/tts-models/vits-zh-aishell3.tar.bz2 tar xvf vits-zh-aishell3.tar.bz2 python3 ./python-api-examples/offline-tts.py \ --vits-model=./vits-aishell3.onnx \ --vits-lexicon=./lexicon.txt \ --vits-tokens=./tokens.txt \ --tts-rule-fsts=./rule.fst \ --sid=21 \ --output-filename=./liubei-21.wav \ "勿以恶小而为之,勿以善小而不为。惟贤惟德,能服于人。122334" You can find more models at https://github.com/k2-fsa/sherpa-onnx/releases/tag/tts-models Please see https://k2-fsa.github.io/sherpa/onnx/tts/index.html for details. """ import argparse import time import sherpa_onnx import soundfile as sf def get_args(): parser = argparse.ArgumentParser( formatter_class=argparse.ArgumentDefaultsHelpFormatter ) parser.add_argument( "--vits-model", type=str, help="Path to vits model.onnx", ) parser.add_argument( "--vits-lexicon", type=str, default="", help="Path to lexicon.txt", ) parser.add_argument( "--vits-tokens", type=str, default="", help="Path to tokens.txt", ) parser.add_argument( "--vits-data-dir", type=str, default="", help="""Path to the dict director of espeak-ng. If it is specified, --vits-lexicon and --vits-tokens are ignored""", ) parser.add_argument( "--tts-rule-fsts", type=str, default="", help="Path to rule.fst", ) parser.add_argument( "--max-num-sentences", type=int, default=2, help="""Max number of sentences in a batch to avoid OOM if the input text is very long. Set it to -1 to process all the sentences in a single batch. A smaller value does not mean it is slower compared to a larger one on CPU. """, ) parser.add_argument( "--output-filename", type=str, default="./generated.wav", help="Path to save generated wave", ) parser.add_argument( "--sid", type=int, default=0, help="""Speaker ID. Used only for multi-speaker models, e.g. models trained using the VCTK dataset. Not used for single-speaker models, e.g., models trained using the LJ speech dataset. """, ) parser.add_argument( "--debug", type=bool, default=False, help="True to show debug messages", ) parser.add_argument( "--provider", type=str, default="cpu", help="valid values: cpu, cuda, coreml", ) parser.add_argument( "--num-threads", type=int, default=1, help="Number of threads for neural network computation", ) parser.add_argument( "--speed", type=float, default=1.0, help="Speech speed. Larger->faster; smaller->slower", ) parser.add_argument( "text", type=str, help="The input text to generate audio for", ) return parser.parse_args() def main(): args = get_args() print(args) tts_config = sherpa_onnx.OfflineTtsConfig( model=sherpa_onnx.OfflineTtsModelConfig( vits=sherpa_onnx.OfflineTtsVitsModelConfig( model=args.vits_model, lexicon=args.vits_lexicon, data_dir=args.vits_data_dir, tokens=args.vits_tokens, ), provider=args.provider, debug=args.debug, num_threads=args.num_threads, ), rule_fsts=args.tts_rule_fsts, max_num_sentences=args.max_num_sentences, ) if not tts_config.validate(): raise ValueError("Please check your config") tts = sherpa_onnx.OfflineTts(tts_config) start = time.time() audio = tts.generate(args.text, sid=args.sid, speed=args.speed) end = time.time() if len(audio.samples) == 0: print("Error in generating audios. Please read previous error messages.") return elapsed_seconds = end - start audio_duration = len(audio.samples) / audio.sample_rate real_time_factor = elapsed_seconds / audio_duration sf.write( args.output_filename, audio.samples, samplerate=audio.sample_rate, subtype="PCM_16", ) print(f"Saved to {args.output_filename}") print(f"The text is '{args.text}'") print(f"Elapsed seconds: {elapsed_seconds:.3f}") print(f"Audio duration in seconds: {audio_duration:.3f}") print(f"RTF: {elapsed_seconds:.3f}/{audio_duration:.3f} = {real_time_factor:.3f}") if __name__ == "__main__": main()