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I am using NAudio and Adobe Media Encoder (separately) to convert a 2-hour MP3 file into a WAV file:

The resulting WAV file is not in sync with the source MP3 file:

MP3 second #996 = WAV second #1000

MP3 second #1991 = WAV second #2000

The Google Speech API requires a WAV file to create a time-coded transcript, but the time codes are then not accurate for the MP3 file (which is a podcast).

I have to use MP3 as the source format, since that is what all the podcasts use.

How can I create a WAV file that is in sync with the source MP3 file?

According to Adobe Soundbooth, the Sample Rate of the podcast MP3 is 44.1 kHz, and the converted WAV file is also 44.1 kHz.

According to the Windows properties, the bit rate of the podcast MP3 is 64kbps. When I transferred the podcast MP3 to a new MP3 file with a bit rate of 320 kbps, that was also out of sync with the original 64kbps file.

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Try using Logic, Pro Tools, or Ableton Live to convert to WAV. The time code should remain synced. Let me know otherwise.

I’m guessing that maybe some setting is being changed, like kHz or bit rate. For example, kHz can be rounded off while converting. Can you check the original setting and the converted file setting to see if anything is different?

Perhaps the original file is at 44.056khz, and you are converting it to 44.1khz, but I’m not sure that this is true.

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