# 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 import os import time import paddle from paddleaudio.backends import soundfile_load as load_audio from paddleaudio.compliance.librosa import melspectrogram from yacs.config import CfgNode from paddlespeech.s2t.utils.log import Log from paddlespeech.vector.io.batch import feature_normalize from paddlespeech.vector.models.ecapa_tdnn import EcapaTdnn from paddlespeech.vector.modules.sid_model import SpeakerIdetification from paddlespeech.vector.training.seeding import seed_everything logger = Log(__name__).getlog() def extract_audio_embedding(args, config): # stage 0: set the training device, cpu or gpu paddle.set_device(args.device) # set the random seed, it is a must for multiprocess training seed_everything(config.seed) # stage 1: build the dnn backbone model network ecapa_tdnn = EcapaTdnn(**config.model) # stage4: build the speaker verification train instance with backbone model model = SpeakerIdetification( backbone=ecapa_tdnn, num_class=config.num_speakers) # stage 2: load the pre-trained model args.load_checkpoint = os.path.abspath( os.path.expanduser(args.load_checkpoint)) # load model checkpoint to sid model state_dict = paddle.load( os.path.join(args.load_checkpoint, 'model.pdparams')) model.set_state_dict(state_dict) logger.info(f'Checkpoint loaded from {args.load_checkpoint}') # stage 3: we must set the model to eval mode model.eval() # stage 4: read the audio data and extract the embedding # wavform is one dimension numpy array waveform, sr = load_audio(args.audio_path) # feat type is numpy array, whose shape is [dim, time] # we need convert the audio feat to one-batch shape [batch, dim, time], where the batch is one # so the final shape is [1, dim, time] start_time = time.time() feat = melspectrogram( x=waveform, sr=config.sr, n_mels=config.n_mels, window_size=config.window_size, hop_length=config.hop_size) feat = paddle.to_tensor(feat).unsqueeze(0) # in inference period, the lengths is all one without padding lengths = paddle.ones([1]) feat = feature_normalize(feat, mean_norm=True, std_norm=False) # model backbone network forward the feats and get the embedding embedding = model.backbone( feat, lengths).squeeze().numpy() # (1, emb_size, 1) -> (emb_size) elapsed_time = time.time() - start_time audio_length = waveform.shape[0] / sr # stage 5: do global norm with external mean and std rtf = elapsed_time / audio_length logger.info(f"{args.device} rft={rtf}") return embedding if __name__ == "__main__": # yapf: disable parser = argparse.ArgumentParser(__doc__) parser.add_argument('--device', choices=['cpu', 'gpu'], default="cpu", help="Select which device to train model, defaults to gpu.") parser.add_argument("--config", default=None, type=str, help="configuration file") parser.add_argument("--load-checkpoint", type=str, default='', help="Directory to load model checkpoint to contiune trainning.") parser.add_argument("--audio-path", default="./data/demo.wav", type=str, help="Single audio file path") args = parser.parse_args() # yapf: enable # https://yaml.org/type/float.html config = CfgNode(new_allowed=True) if args.config: config.merge_from_file(args.config) config.freeze() print(config) extract_audio_embedding(args, config)