From db33d0f610f7593ad1e61f8c3cc68de4a9b68f99 Mon Sep 17 00:00:00 2001
From: 雾聪 <wucong.lyb@alibaba-inc.com>
Date: 星期四, 12 十月 2023 13:05:26 +0800
Subject: [PATCH] Merge branch 'main' of https://github.com/alibaba-damo-academy/FunASR into main
---
funasr/utils/wav_utils.py | 4 ++--
1 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/funasr/utils/wav_utils.py b/funasr/utils/wav_utils.py
index e523495..bd067c2 100644
--- a/funasr/utils/wav_utils.py
+++ b/funasr/utils/wav_utils.py
@@ -189,9 +189,9 @@
def wav2num_frame(wav_path, frontend_conf):
try:
- waveform, audio_sr = torchaudio.load(wav_file)
+ waveform, sampling_rate = torchaudio.load(wav_path)
except:
- waveform, audio_sr = soundfile.read(wav_file)
+ 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"])
--
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