You signed in with another tab or window. Reload to refresh your session.You signed out in another tab or window. Reload to refresh your session.You switched accounts on another tab or window. Reload to refresh your session.Dismiss alert
Does the lib allow you to get the .wav or flac encoded data on an arbitrary timeslice ? Im looking to abstract the google speech to text service away from the client so the plan is to create the .wav or the flac on say 4 second intervals and pipe that to a server to proceed with the rest of the workflow. Couldnt see anything in the readme, any pointers would be greatly apreciated.
The text was updated successfully, but these errors were encountered:
basically, you'd probably want to select a block size according to your sampling rate, so that your write-callback can collect encoded frames until your timeslice is met and then emit it
(but keep in mind the explanation of block size in the FLAC format description, i.e. it should not be too large or too small, e.g. I typically see 4096 for auto selected block size (for 44.1 kHz sampling rate))
(and note, that "normal" WAV usually requires a header for each chunk; there are variant formats for streaming e.g. audio/l16, but the receiving end/backend would need to be able to deal with that)
Does the lib allow you to get the .wav or flac encoded data on an arbitrary timeslice ? Im looking to abstract the google speech to text service away from the client so the plan is to create the .wav or the flac on say 4 second intervals and pipe that to a server to proceed with the rest of the workflow. Couldnt see anything in the readme, any pointers would be greatly apreciated.
The text was updated successfully, but these errors were encountered: