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>>> jehoshua |
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>>> elpimous_robot |
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>>> jehoshua |
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>>> elpimous_robot |
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>>> jehoshua
[January 21, 2018, 10:21pm]
Have recently installed Deepspeech and run it against a small WAV file -
55 words slash
275 characters, length - 19.968s
The accuracy is not good at all, with an error rate of 43.63%. If I use
a small script to test the same audio with Google Speech Recognition ,
the error rate is only 20%. Still high but better.
Have even cut that WAV down to 9 seconds, yet no change in accuracy.
Initially used the command as per docs
deepspeech models/output_graph.pb my_audio_file.wav models/alphabet.txt
and have added the files lm.binary and trie to the command, and tests
reveal no change.
The audio is English and very clear. I followed the guidlines re
specifications for the WAV and it is the same as the sample WAV's for
Deepspeech.
In regards to these sample model files:
lm.binary slash
trie slash
output_graph.pb
what bearing do they have on accuracy ? Do I need to create a specific
model to improve the accuracy, or 'add to' (train) the sample models ?
There are hundreds of audio files that we have for just one
speaker/person, and we wish to be able to improve the accuracy
substantially.
Just wondering that because this is for a specific purpose, is it better
to create a model for that purpose only, rather than use the existing
(sample) models ?
[This is an archived TTS discussion thread from discourse.mozilla.org/t/improving-accuracy-by-creating-a-specific-model]
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