#!/usr/bin/env python3 # Copyright 2024 Xiaomi Corp. (authors: Xiaoyu Yang) # # See ../../../../LICENSE for clarification regarding multiple authors # # 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. """ This script loads a checkpoint and uses it to decode waves. You can generate the checkpoint with the following command: Note: This is an example for the AudioSet dataset, if you are using different dataset, you should change the argument values according to your dataset. Usage of this script: repo_url=https://huggingface.co/marcoyang/icefall-audio-tagging-audioset-zipformer-2024-03-12 repo=$(basename $repo_url) GIT_LFS_SKIP_SMUDGE=1 git clone $repo_url pushd $repo/exp git lfs pull --include pretrained.pt popd python3 zipformer/pretrained.py \ --checkpoint $repo/exp/pretrained.pt \ --label-dict $repo/data/class_labels_indices.csv \ $repo/test_wavs/1.wav \ $repo/test_wavs/2.wav \ $repo/test_wavs/3.wav \ $repo/test_wavs/4.wav """ import argparse import csv import logging import math from typing import List import kaldifeat import torch import torchaudio from torch.nn.utils.rnn import pad_sequence from train import add_model_arguments, get_model, get_params def get_parser(): parser = argparse.ArgumentParser( formatter_class=argparse.ArgumentDefaultsHelpFormatter ) parser.add_argument( "--checkpoint", type=str, required=True, help="Path to the checkpoint. " "The checkpoint is assumed to be saved by " "icefall.checkpoint.save_checkpoint().", ) parser.add_argument( "--label-dict", type=str, help="""class_labels_indices.csv.""", ) parser.add_argument( "sound_files", type=str, nargs="+", help="The input sound file(s) to transcribe. " "Supported formats are those supported by torchaudio.load(). " "For example, wav and flac are supported. " "The sample rate has to be 16kHz.", ) parser.add_argument( "--sample-rate", type=int, default=16000, help="The sample rate of the input sound file", ) add_model_arguments(parser) return parser def read_sound_files( filenames: List[str], expected_sample_rate: float ) -> List[torch.Tensor]: """Read a list of sound files into a list 1-D float32 torch tensors. Args: filenames: A list of sound filenames. expected_sample_rate: The expected sample rate of the sound files. Returns: Return a list of 1-D float32 torch tensors. """ ans = [] for f in filenames: wave, sample_rate = torchaudio.load(f) assert ( sample_rate == expected_sample_rate ), f"expected sample rate: {expected_sample_rate}. Given: {sample_rate}" # We use only the first channel ans.append(wave[0].contiguous()) return ans @torch.no_grad() def main(): parser = get_parser() args = parser.parse_args() params = get_params() params.update(vars(args)) logging.info(f"{params}") device = torch.device("cpu") if torch.cuda.is_available(): device = torch.device("cuda", 0) logging.info(f"device: {device}") logging.info("Creating model") model = get_model(params) num_param = sum([p.numel() for p in model.parameters()]) logging.info(f"Number of model parameters: {num_param}") checkpoint = torch.load(args.checkpoint, map_location="cpu", weights_only=False) model.load_state_dict(checkpoint["model"], strict=False) model.to(device) model.eval() # get the label dictionary label_dict = {} with open(params.label_dict, "r") as f: reader = csv.reader(f, delimiter=",") for i, row in enumerate(reader): if i == 0: continue label_dict[int(row[0])] = row[2] logging.info("Constructing Fbank computer") opts = kaldifeat.FbankOptions() opts.device = device opts.frame_opts.dither = 0 opts.frame_opts.snip_edges = False opts.frame_opts.samp_freq = params.sample_rate opts.mel_opts.num_bins = params.feature_dim opts.mel_opts.high_freq = -400 fbank = kaldifeat.Fbank(opts) logging.info(f"Reading sound files: {params.sound_files}") waves = read_sound_files( filenames=params.sound_files, expected_sample_rate=params.sample_rate ) waves = [w.to(device) for w in waves] logging.info("Decoding started") features = fbank(waves) feature_lengths = [f.size(0) for f in features] features = pad_sequence(features, batch_first=True, padding_value=math.log(1e-10)) feature_lengths = torch.tensor(feature_lengths, device=device) # model forward and predict the audio events encoder_out, encoder_out_lens = model.forward_encoder(features, feature_lengths) logits = model.forward_audio_tagging(encoder_out, encoder_out_lens) for filename, logit in zip(args.sound_files, logits): topk_prob, topk_index = logit.sigmoid().topk(5) topk_labels = [label_dict[index.item()] for index in topk_index] logging.info( f"{filename}: Top 5 predicted labels are {topk_labels} with " f"probability of {topk_prob.tolist()}" ) logging.info("Done") if __name__ == "__main__": formatter = "%(asctime)s %(levelname)s [%(filename)s:%(lineno)d] %(message)s" logging.basicConfig(format=formatter, level=logging.INFO) main()