From 69dcdbcfc0c21627c40fb8f7c435136c11691574 Mon Sep 17 00:00:00 2001
From: 雾聪 <wucong.lyb@alibaba-inc.com>
Date: 星期二, 27 六月 2023 17:19:49 +0800
Subject: [PATCH] Merge branch 'main' of https://github.com/alibaba-damo-academy/FunASR into main
---
funasr/utils/wav_utils.py | 15 +++++++++++++--
1 files changed, 13 insertions(+), 2 deletions(-)
diff --git a/funasr/utils/wav_utils.py b/funasr/utils/wav_utils.py
index ebb80d2..bd067c2 100644
--- a/funasr/utils/wav_utils.py
+++ b/funasr/utils/wav_utils.py
@@ -11,6 +11,7 @@
import numpy as np
import torch
import torchaudio
+import soundfile
import torchaudio.compliance.kaldi as kaldi
@@ -162,7 +163,13 @@
waveform = torch.from_numpy(waveform.reshape(1, -1))
else:
# load pcm from wav, and resample
- waveform, audio_sr = torchaudio.load(wav_file)
+ try:
+ waveform, audio_sr = torchaudio.load(wav_file)
+ except:
+ waveform, audio_sr = soundfile.read(wav_file, dtype='float32')
+ if waveform.ndim == 2:
+ waveform = waveform[:, 0]
+ waveform = torch.tensor(np.expand_dims(waveform, axis=0))
waveform = waveform * (1 << 15)
waveform = torch_resample(waveform, audio_sr, model_sr)
@@ -181,7 +188,11 @@
def wav2num_frame(wav_path, frontend_conf):
- waveform, sampling_rate = torchaudio.load(wav_path)
+ try:
+ waveform, sampling_rate = torchaudio.load(wav_path)
+ except:
+ waveform, sampling_rate = soundfile.read(wav_path)
+ waveform = torch.tensor(np.expand_dims(waveform, axis=0))
speech_length = (waveform.shape[1] / sampling_rate) * 1000.
n_frames = (waveform.shape[1] * 1000.0) / (sampling_rate * frontend_conf["frame_shift"] * frontend_conf["lfr_n"])
feature_dim = frontend_conf["n_mels"] * frontend_conf["lfr_m"]
--
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