# Copyright (c) 2021 PaddlePaddle Authors. All Rights Reserved. # # Licensed under the Apache License, Version 2.0 (the "License"); # you may not use this file except in compliance with the License. # You may obtain a copy of the License at # # http://www.apache.org/licenses/LICENSE-2.0 # # Unless required by applicable law or agreed to in writing, software # distributed under the License is distributed on an "AS IS" BASIS, # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. # See the License for the specific language governing permissions and # limitations under the License. import argparse from pathlib import Path import numpy import soundfile as sf from paddle import inference from timer import timer from paddlespeech.t2s.exps.syn_utils import get_frontend from paddlespeech.t2s.exps.syn_utils import get_sentences from paddlespeech.t2s.utils import str2bool def get_predictor(args, filed='am'): full_name = '' if filed == 'am': full_name = args.am elif filed == 'voc': full_name = args.voc model_name = full_name[:full_name.rindex('_')] config = inference.Config( str(Path(args.inference_dir) / (full_name + ".pdmodel")), str(Path(args.inference_dir) / (full_name + ".pdiparams"))) if args.device == "gpu": config.enable_use_gpu(100, 0) elif args.device == "cpu": config.disable_gpu() # This line must be commented for fastspeech2, if not, it will OOM if model_name != 'fastspeech2': config.enable_memory_optim() predictor = inference.create_predictor(config) return predictor def get_am_output(args, am_predictor, frontend, merge_sentences, input): am_name = args.am[:args.am.rindex('_')] am_dataset = args.am[args.am.rindex('_') + 1:] am_input_names = am_predictor.get_input_names() get_tone_ids = False get_spk_id = False if am_name == 'speedyspeech': get_tone_ids = True if am_dataset in {"aishell3", "vctk"} and args.speaker_dict: get_spk_id = True spk_id = numpy.array([args.spk_id]) if args.lang == 'zh': input_ids = frontend.get_input_ids( input, merge_sentences=merge_sentences, get_tone_ids=get_tone_ids) phone_ids = input_ids["phone_ids"] elif args.lang == 'en': input_ids = frontend.get_input_ids( input, merge_sentences=merge_sentences) phone_ids = input_ids["phone_ids"] else: print("lang should in {'zh', 'en'}!") if get_tone_ids: tone_ids = input_ids["tone_ids"] tones = tone_ids[0].numpy() tones_handle = am_predictor.get_input_handle(am_input_names[1]) tones_handle.reshape(tones.shape) tones_handle.copy_from_cpu(tones) if get_spk_id: spk_id_handle = am_predictor.get_input_handle(am_input_names[1]) spk_id_handle.reshape(spk_id.shape) spk_id_handle.copy_from_cpu(spk_id) phones = phone_ids[0].numpy() phones_handle = am_predictor.get_input_handle(am_input_names[0]) phones_handle.reshape(phones.shape) phones_handle.copy_from_cpu(phones) am_predictor.run() am_output_names = am_predictor.get_output_names() am_output_handle = am_predictor.get_output_handle(am_output_names[0]) am_output_data = am_output_handle.copy_to_cpu() return am_output_data def get_voc_output(args, voc_predictor, input): voc_input_names = voc_predictor.get_input_names() mel_handle = voc_predictor.get_input_handle(voc_input_names[0]) mel_handle.reshape(input.shape) mel_handle.copy_from_cpu(input) voc_predictor.run() voc_output_names = voc_predictor.get_output_names() voc_output_handle = voc_predictor.get_output_handle(voc_output_names[0]) wav = voc_output_handle.copy_to_cpu() return wav def parse_args(): parser = argparse.ArgumentParser( description="Paddle Infernce with acoustic model & vocoder.") # acoustic model parser.add_argument( '--am', type=str, default='fastspeech2_csmsc', choices=[ 'speedyspeech_csmsc', 'fastspeech2_csmsc', 'fastspeech2_aishell3', 'fastspeech2_vctk', 'tacotron2_csmsc' ], help='Choose acoustic model type of tts task.') parser.add_argument( "--phones_dict", type=str, default=None, help="phone vocabulary file.") parser.add_argument( "--tones_dict", type=str, default=None, help="tone vocabulary file.") parser.add_argument( "--speaker_dict", type=str, default=None, help="speaker id map file.") parser.add_argument( '--spk_id', type=int, default=0, help='spk id for multi speaker acoustic model') # voc parser.add_argument( '--voc', type=str, default='pwgan_csmsc', choices=[ 'pwgan_csmsc', 'mb_melgan_csmsc', 'hifigan_csmsc', 'pwgan_aishell3', 'pwgan_vctk', 'wavernn_csmsc' ], help='Choose vocoder type of tts task.') # other parser.add_argument( '--lang', type=str, default='zh', help='Choose model language. zh or en') parser.add_argument( "--text", type=str, help="text to synthesize, a 'utt_id sentence' pair per line") parser.add_argument( "--inference_dir", type=str, help="dir to save inference models") parser.add_argument("--output_dir", type=str, help="output dir") # inference parser.add_argument( "--use_trt", type=str2bool, default=False, help="Whether to use inference engin TensorRT.", ) parser.add_argument( "--int8", type=str2bool, default=False, help="Whether to use int8 inference.", ) parser.add_argument( "--fp16", type=str2bool, default=False, help="Whether to use float16 inference.", ) parser.add_argument( "--device", default="gpu", choices=["gpu", "cpu"], help="Device selected for inference.", ) args, _ = parser.parse_known_args() return args # only inference for models trained with csmsc now def main(): args = parse_args() # frontend frontend = get_frontend(args) # am_predictor am_predictor = get_predictor(args, filed='am') # model: {model_name}_{dataset} am_dataset = args.am[args.am.rindex('_') + 1:] # voc_predictor voc_predictor = get_predictor(args, filed='voc') output_dir = Path(args.output_dir) output_dir.mkdir(parents=True, exist_ok=True) sentences = get_sentences(args) merge_sentences = True fs = 24000 if am_dataset != 'ljspeech' else 22050 # warmup for utt_id, sentence in sentences[:3]: with timer() as t: am_output_data = get_am_output( args, am_predictor=am_predictor, frontend=frontend, merge_sentences=merge_sentences, input=sentence) wav = get_voc_output( args, voc_predictor=voc_predictor, input=am_output_data) speed = wav.size / t.elapse rtf = fs / speed print( f"{utt_id}, mel: {am_output_data.shape}, wave: {wav.shape}, time: {t.elapse}s, Hz: {speed}, RTF: {rtf}." ) print("warm up done!") N = 0 T = 0 for utt_id, sentence in sentences: with timer() as t: am_output_data = get_am_output( args, am_predictor=am_predictor, frontend=frontend, merge_sentences=merge_sentences, input=sentence) wav = get_voc_output( args, voc_predictor=voc_predictor, input=am_output_data) N += wav.size T += t.elapse speed = wav.size / t.elapse rtf = fs / speed sf.write(output_dir / (utt_id + ".wav"), wav, samplerate=24000) print( f"{utt_id}, mel: {am_output_data.shape}, wave: {wav.shape}, time: {t.elapse}s, Hz: {speed}, RTF: {rtf}." ) print(f"{utt_id} done!") print(f"generation speed: {N / T}Hz, RTF: {fs / (N / T) }") if __name__ == "__main__": main()