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Gear / Technical Help => Post-Processing, Computer / Streaming / Internet Devices & Related Activity => Topic started by: Brian Skalinder on March 10, 2007, 01:46:58 AM
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Edit to add: if you have trouble with a file playing too fast/slow and need help, best bet: start a new thread, you'll get more responses! :)
Keywords: speed correction, sample rate, pitch, time, wav header, too fast, too slow, chipmunk, chipmunks
Mismatched WAV header sample rate and actual data sample rate seems a fairly common problem. The result: the recorded file sounds too fast or too slow. The following table outlines mismatched WAVE header and actual data sample rates and a description of the resulting improper sound:
Header | Data | Result
=============================================
44.1 kHz | 32.0 kHz | Too fast
44.1 kHz | 44.1 kHz | Just right!
44.1 kHz | 48.0 kHz | Too slow
44.1 kHz | 88.2 kHz | Way too slow
44.1 kHz | 96.0 kHz | Way too slow
44.1 kHz | 192.0 kHz | Ridiculously slow
---------------------------------------------
48.0 kHz | 32.0 kHz | Much too fast
48.0 kHz | 44.1 kHz | Too fast
48.0 kHz | 48.0 kHz | Just right!
48.0 kHz | 88.2 kHz | Way too slow
48.0 kHz | 96.0 kHz | Way too slow
48.0 kHz | 192.0 kHz | Ridiculously slow
---------------------------------------------
88.2 kHz | 32.0 kHz | Ridiculously fast
88.2 kHz | 44.1 kHz | Way too fast
88.2 kHz | 48.0 kHz | Way too fast
88.2 kHz | 88.2 kHz | Just right!
88.2 kHz | 96.0 kHz | Too slow
88.2 kHz | 192.0 kHz | Way too slow
---------------------------------------------
96.0 kHz | 32.0 kHz | Ridiculously fast
96.0 kHz | 44.1 kHz | Way too fast
96.0 kHz | 48.0 kHz | Way too fast
96.0 kHz | 88.2 kHz | Too fast
96.0 kHz | 96.0 kHz | Just right!
96.0 kHz | 192.0 kHz | Way too slow
---------------------------------------------
192.0 kHz | 32.0 kHz | Ludicrously fast
192.0 kHz | 44.1 kHz | Ludicrously fast
192.0 kHz | 48.0 kHz | Ludicrously fast
192.0 kHz | 88.2 kHz | Way too fast
192.0 kHz | 96.0 kHz | Way too fast
192.0 kHz | 192.0 kHz | Just right!
When encountering mismatched WAV header sample rate and data sample rate, simply change the WAV header sample rate so it matches the data sample rate. For example, if the WAV header's sample rate is 48 kHz and the data's sample rate is 44.1 kHz (symptom: the recorded file sounds fast), change the WAV header sample rate to 44.1 kHz. Note: this is NOT sample rate conversion (i.e. resampling, typically a somewhat lengthy, and destructive, process), but rather a quick and easy change to the WAV header only - none of the actual data is changed. Different commonly used audio editors provide slightly different ways of accomplishing this goal. Here are some of the more common audio editor workflows for fixing this problem:
Changing the WAV File Header's Sample Rate (not Sample Rate Conversion)
Change Sample Rate (http://www.audiosignal.co.uk/Resources/changerate.zip)
Edit to add: Thanks to Teddy for a master link in another forum that lead me to this tool: Change Sample Rate (http://www.audiosignal.co.uk/Resources/changerate.zip), a utility that changes the WAV header to the user's desired sample rate. It does not perform sample rate conversion.
Audacity
- Import the file
- Set Track Rate to the desired sample rate value (black upside-down triangle dropdown menu next to track name)
- Set the Track Sample Format to the same value as the source file (e.g. if importing a 16-bit file, set the value to 16-bit)
- Set Preferences | Quality | High-quality Dither to None
- Set Preferences | File Formats | Uncompressed Export Format to the same format as the source file (e.g. if importing a 24-bit file, set the value to Other | WAV (Microsoft) and Signed 24 bit PCM
- Export to WAV
WaveLab
- Open the file
- In Edit | Audio Properties select desired sample rate value
- File | Save As and confirm Wave (PCM) attributes are defined properly, mainly: set Sample Rate to the same sample rate as the data and Bit Resolution to the same value as the original file.
- Save the file to the desired filename
Adobe Audition (Syntrillium Cool Edit Pro before re-branding)
For both 16- and 24-bit files:
- Open the file in the Waveform View
- Set Edit | Adjust Sample Rate to the desired value
Then...
...for a 16-bit file if Options | Settings | Data | Auto-convert all data to 32-bit upon opening is not ticked:
- File | Save As to the desired filename
or
...for a 16-bit file if Options | Settings | Data | Auto-convert all data to 32-bit upon opening is ticked:
- In Edit | Convert Sample Type set the Sample Rate to the same sample rate as the data, the Resolution to 16-bit, and un-tick the Enable Dithering checkbox
- File | Save As to the desired filename
or
...for a 24-bit file:
- File | Save As
- Press the Options button
- From the Format 32-bit data as dropdown select 24-bit packed int (type 1, 24-bit) and untick the Enable Dithering checkbox
- Save to the desired filename
MAGIX Samplitude SE v8
- Load Audio file
- Open Playback Options by typing p or selecting Options | Project Properties | Playback Options
- Set the Sample Rate to the desired value
- In the Project Sample Rate Change popup window, untick the Adapt audio objects to the new sample rate and untick Adapt MIDI objects to the new sample rate, click OK and close the Playback Parameters window
- File | Export Audio | Wave and press the Format Settings button
- Confirm values in the Choose Wave Codec popup window: set the Sample Rate to the desired value (should default) and the Format to the same value as the original file (should default)
- Save the file to the desired filename
Sony Sound Forge
Thanks to Genghis Cougar Mellen Kahn for these instructions...
- Open the file
- Process Tab | Resample and select the desired sample rate
- Tick the Set sample rate only (do not resample checkbox
- Save the file
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Nice post. Wanted to add the freeware (for mac) called soundhack (http://www.soundhack.com/freeware.php) that I use for this purpose.
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Nice post. Wanted to add the freeware (for mac) called soundhack (http://www.soundhack.com/freeware.php) that I use for this purpose.
Any chance you're willing/able to post the workflow?
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Well...
Launch Soundhack
open file (apple O) or File, Open
select Header Change under the "Hack" column (apple H)
change the sample rate to the desired rate
click Apply and then Save Info.
The file will now have a new header written in it.
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Well...
Easy! Thanks, cleantone. :coolguy:
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well i used Audacity and the file still says that it's 48 sampling rate, but it plays fine and isn't chipmunked anymore. did i do anythign wrong? ???
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That just means that the file now "knows" what sampling rate it is and plays properly. So it sounds like it is a 48khz file then right? You would need to convert the sample rate to get to 44.1khz if that is your goal.
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That just means that the file now "knows" what sampling rate it is and plays properly. So it sounds like it is a 48khz file then right? You would need to convert the sample rate to get to 44.1khz if that is your goal.
that's the thing, it sounds like it's a 44.1 file and i burned it to CD and it plays fine. just still says it's a 48 file. weird.
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Thanks a bunch Brian! :)
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For Audacity, I believe you also need to change the project rate in the lower left hand corner to the desired sample rate. Before I did so, Audacity acted as though it was going to dither the file even though I had changed that option to none, and was teed up to take quite a long time to convert. After changing the project rate to 44.1 kHz, the conversion took a very short time.
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*nix heads who like commandline tools..... strip the wave file header, and replace it with a good one
Example: I recorded a 16/44.1 file (AD20 > H120)... for some reason header was set to 24000.
1) Strip header: shntool cat -nh broken_input_file.wav > temp.raw
2) Replace header: sox -t raw -w -c 2 -s -r 44100 temp.raw fixed_output_file.wav
Notes: cat -nh means "no header"
sox -t raw means type raw
-w means 16bit words
-c 2 means 2 channels
-s means signed data
-r 44100 means
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Great, I love finding new stuff:
http://sox.sourceforge.net/Main/HomePage
+T