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This also affects audiofile.samples() and audiofile.duration() as both uses audiofile.convert_to_wav(), which relies on audiofile.read() to get the ground truth for sampling rate:
This could be fixed by relying on audiofile.sampling_rate() inside audiofile.convert_to_wav(), but maybe we should also check, why the opus file behaves like this.
ffmpeg always returns opus files wit a sampling rate of 48000 Hz, if not otherwise specified. This is known and marked as not to be fixed, see https://trac.ffmpeg.org/ticket/5240. Which means in order to fix it we need to provide ffmpeg with the desired sampling rate when converting an opus file to wav.
The GigaSpeech dataset contains audio in opus files stored at 16,000 Hz.
I attached the file
POD0000002525.opus
to this issue.But when reading the file in the usual way, the sampling rate no longer matches:
This also affects
audiofile.samples()
andaudiofile.duration()
as both usesaudiofile.convert_to_wav()
, which relies onaudiofile.read()
to get the ground truth for sampling rate:This could be fixed by relying on
audiofile.sampling_rate()
insideaudiofile.convert_to_wav()
, but maybe we should also check, why the opus file behaves like this.POD0000002525.zip
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