From 7da5b31e25845905b814dfa6282ebf09ada329d5 Mon Sep 17 00:00:00 2001
From: 游雁 <zhifu.gzf@alibaba-inc.com>
Date: 星期二, 27 六月 2023 16:57:43 +0800
Subject: [PATCH] Merge branch 'main' of github.com:alibaba-damo-academy/FunASR add
---
funasr/utils/wav_utils.py | 8 +++++---
1 files changed, 5 insertions(+), 3 deletions(-)
diff --git a/funasr/utils/wav_utils.py b/funasr/utils/wav_utils.py
index a6e394f..bd067c2 100644
--- a/funasr/utils/wav_utils.py
+++ b/funasr/utils/wav_utils.py
@@ -166,7 +166,9 @@
try:
waveform, audio_sr = torchaudio.load(wav_file)
except:
- waveform, audio_sr = soundfile.read(wav_file)
+ 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)
@@ -187,9 +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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