Re: [Kaldi-users] LibriSpeech nnet2 model: training more on a new dataset

[Neil N. <nn...@in...>]

Vassil,

I have always used Lame (Ubuntu Software Center is your friend) to
convert between wav and mp3. It is well regarded. I suggest SOX for
down-sampling. Spek will give a spectral analysis picture for the entire
file. Audacity will give an ongoing spectral analysis but it is not
frequency labeled. Sonic Visualizer may have something. Upgrading to
Ubuntu 14.04 can be tricky in spots but something to consider since the
versions of GCC and all the software tend to be limited to the OS rev.

Neil

On 07/05/2015 02:05 AM, Vassil Panayotov wrote:
> BTW, when preparing LibriSpeech, I've noticed that the quality of MP3
> conversion can vary substantially, depending on the particular tool used.
> For example the output of mpg123(or maybe it was mpg321) was very noisy and
> the ASR WER was 10-15% absolute higher than when alternative MP3 decoders
> were used. When converting to 16kHz .wav ffmpeg cuts off the frequencies
> higher than 7kHz. So eventually I settled for mplayer. It preserves the
> frequency content in the 7-8kHz range and as far as I could tell the audio
> sounded a bit "closer" to the original recording, although I'm not sure if
> there is any measurable difference in ASR performance b/w ffmpeg and
> mplayer produced .wav-s. The versions of the tools I've tried were those
> shipped with Ubuntu 10.04 and 12.04, so the issues may be fixed in the more
> recent releases.
> 
> Vassil

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