ADPCM Timing Issue -Sound Forge - PC - MAGIX Community Forums Archive

Subject:ADPCM Timing Issue Posted by: rdobbs Date:8/30/2012 8:39:20 AM Can anyone explain this? A file in Sound Forge Audio Studio is exactly 1 minute 01:00:00:000 The file is rendered as an ADPCM wav file, 4 bit mono and sent to the client, who is a radio station. The station keeps telling me that the file is long: 01:00:00:018. I have unchecked the Ignore fact chunk when opening compressed WAVE files, and that fixed the extraneous audio issue at the end of the file, and I thought it fixed the timing issue, but every file I send them seems to be long, but the time is exactly 1 minute on my Sound Forge timeline. Status Format is set to Time.
Subject:RE: ADPCM Timing Issue Reply by: roblesinge Date:8/30/2012 9:41:50 AM This is most likely header information that is being perceived as audio information by whatever system the client is using. Typically true dialogic ADPCM files don't have headers, since the system is expected to know what kind of file it is and play it natively. To be honest, I've never had much luck making ADPCM files with Sound Forge. My company uses a program called Vox Studio for most telephony related files. You might try another program to see if the results are the same elsewhere. Just one more thing you might try: When you save the file, do you have the, "Save metadata with file" box checked at the bottom of the save window? If so, try unchecking this and see what happens. Rob.
Subject:RE: ADPCM Timing Issue Reply by: rraud Date:8/30/2012 10:21:42 AM I assume your using the the 'time' format: 00:00:00:000 - Make sure "Snapping" is disabled, it may default to the nearest zero crossing when you are trimming. "Quantize to frames" should be disabled as well. both in the "Options" menu. (If it's like SF Pro) - Try a higher resolution time format, samples per second. 44,100 x 60 = 2,646,000 samples - I'm not very familiar with that format except it uses a form of data compression maybe it is adding time to the file? .. for instance, encoding an MP3 file, inherently adds about 30ms. If that's the case, you could time compress the file by 18ms and re-render. - Is the sample rate being changed somewhere down the line from the original? BTW: Probably a typo but 01:00:00:018 is one hour and eighteen milliseconds

Subject:ADPCM Timing Issue

Posted by: rdobbs

Date:8/30/2012 8:39:20 AM

Can anyone explain this?
A file in Sound Forge Audio Studio is exactly 1 minute 01:00:00:000
The file is rendered as an ADPCM wav file, 4 bit mono and sent to the client, who is a radio station. The station keeps telling me that the file is long: 01:00:00:018. I have unchecked the Ignore fact chunk when opening compressed WAVE files, and that fixed the extraneous audio issue at the end of the file, and I thought it fixed the timing issue, but every file I send them seems to be long, but the time is exactly 1 minute on my Sound Forge timeline. Status Format is set to Time.

Subject:RE: ADPCM Timing Issue

Reply by: roblesinge

Date:8/30/2012 9:41:50 AM

This is most likely header information that is being perceived as audio information by whatever system the client is using. Typically true dialogic ADPCM files don't have headers, since the system is expected to know what kind of file it is and play it natively. To be honest, I've never had much luck making ADPCM files with Sound Forge. My company uses a program called Vox Studio for most telephony related files. You might try another program to see if the results are the same elsewhere.

Just one more thing you might try: When you save the file, do you have the, "Save metadata with file" box checked at the bottom of the save window? If so, try unchecking this and see what happens.

Rob.

Subject:RE: ADPCM Timing Issue

Reply by: rraud

Date:8/30/2012 10:21:42 AM

I assume your using the the 'time' format: 00:00:00:000
- Make sure "Snapping" is disabled, it may default to the nearest zero crossing when you are trimming. "Quantize to frames" should be disabled as well. both in the "Options" menu. (If it's like SF Pro)
- Try a higher resolution time format, samples per second. 44,100 x 60 = 2,646,000 samples
- I'm not very familiar with that format except it uses a form of data compression maybe it is adding time to the file? .. for instance, encoding an MP3 file, inherently adds about 30ms. If that's the case, you could time compress the file by 18ms and re-render.
- Is the sample rate being changed somewhere down the line from the original?

BTW: Probably a typo but 01:00:00:018 is one hour and eighteen milliseconds