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Old Oct 26th, 2005, 12:24 PM   #1
 
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I recently finished my second show for my podcast. Everything is working fine, except for when I listen to the show via Odeo, PodNova, or PodcastAlley. It's all high-pitched and fast. But if I listen to the MP3 from the source URL it sounds perfect. I don't understand why this is happening. The first show sounds fine on all those web players. Could a different bitrate in show 1 and show 2 be causing this problem? Show 1 has a bitrate of 128 and show 2 has a bitrate of 80.

The feed URL is http://feeds.feedburner.com/dfpod
The site is: http://pod.digitalfusions.net
PodcastAlley listing: http://www.podcastalley.com/podcast_...p?pod_id=12079
Odeo listing: http://odeo.com/channel/34482/view
PodNova listing: http://www.podnova.com/add.srf?url=h...rner.com/dfpod

I have a Yahoo! Podcasts listing but it sound fine in their web player: http://podcasts.yahoo.com/series?s=4...a0df048d6c6ba9

There's a difference with the Yahoo! one though. It has the original feed, and not the FeedBurner one. Also, another thing I should mention is that the file was corrupted yesterday so I had to redo the show. Is there any possibility that might be the problem?
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Old Oct 26th, 2005, 12:31 PM   #2
 
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This is actually a limitation of the Flash Player (used at all those sites you mentioned) and has to do with the bitrate your file is recorded at. here is some infor for you:


Code:
The issue with the "chimpmonk" effect is due to the way that the mp3 was originally encoded. Macromedia Flash can handle most standard mp3 encodings with any bitrate. However, some mp3 encoders use "non-standard" encoding technics that Macromdia Flash can not handle... I have only been ablt to reproduce the "chimpmonk" effect while using musicmatch's mp3PRO setting. The standard mp3 setting in musicmatch works fine, but when a file is encoded with the mp3PRO setting the file plays through wimpy with the "chimpmonk" effect... I have been unable to find documentation on this issue on Macromedia's web site. This issue is not a function of wimpy, but rather an issue with the Flash plugin. The only solution i can offer at the moment is to re-encode your mp3's with a standard mp3 encoder. Wimpy will support VBR encoding and any bitrate. just be sure that you don't use mp3PRO encoding.

A good mp3 encoder is dbPowerAmp, which is what I currently use. It's a small, right-click type utility that makes encoding an mp3 a snap. Usig the standard settings in dbPowerAmp works great with Wimpy. 

"Just an FYI, I encountered the "chipmunk" problem as well when I used the Wimpy Button because I was encoding MP3s at 96kbps using iTunes. An MP3 encoded at 128kbps seemed ok, but anything else (even encoding the files first at one bit rate, and then another) gave me either a faster or slower playing speed."

"I looked at your FAQs and found that this problem has been documented, but not solved."

"I was able to solve my problem by going into iTunes prefs, choosing 'custom' for the MP3 encoding, and then choosing 44.1kHz for the sample rate instead of 'auto'. Apparently when iTunes uses an auto bit rate, the Flash player is unable to adjust its playing speeds to accomodate the optimized MP3 file."

"So, if other customers come to you with this problem, make sure that they have specified the bit rate at 44.1kHz instead of letting iTunes choose."

Flash goes all chipmunk on a LAME encode at 40kbps mono, but works at 32

Jack at Jukebox Alive notes:

For low bitrates (less than 32) I have the option of resampling at:
8 khz.
11.025 khz.
12 khz.
16 khz.
22.05 khz.
24 khz.

Of those, flash seems to only play nice with 11.025 or 22.05, it was defaulting to 24

...A slightly more robust answer to the resampling issue for
users of Lame mp3 encoders, an additional call to -- resample 22.05
will force the sample rate flash seems to need for compact mp3s

One additional note:
It seems as though you should always try and "set" every configuration. Leaving your compression utility to "auto" or "default" is probably not a good idea.
Found Here >> http://wimpyplayer.com/support/faqs/known.html#1
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Old Oct 26th, 2005, 12:35 PM   #3
 
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Quote:
Originally Posted by Vortech
I recently finished my second show for my podcast. Everything is working fine, except for when I listen to the show via Odeo, PodNova, or PodcastAlley. It's all high-pitched and fast. But if I listen to the MP3 from the source URL it sounds perfect. I don't understand why this is happening. The first show sounds fine on all those web players. Could a different bitrate in show 1 and show 2 be causing this problem? Show 1 has a bitrate of 128 and show 2 has a bitrate of 80.
Check your sample rate. There's a bug in Macromedia Flash that causes MP3 files at sample rates other than 22 kHz or 44.1 kHz to play at those rates anyway, and that causes the chipmunk effect. And of course most online MP3 players are going to be Flash-based.
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Old Oct 26th, 2005, 12:39 PM   #4
 
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So what exactly do I need to change? I'm using Audacity and I'd like to export my podcast episodes using a bitrate of 80.
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Old Oct 26th, 2005, 12:42 PM   #5
 
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Quote:
Originally Posted by Vortech
So what exactly do I need to change? I'm using Audacity and I'd like to export my podcast episodes using a bitrate of 80.
Set the sample rate for your project to 44.1 kHz. This is an entirely different number than the bit rate.
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Old Oct 26th, 2005, 12:52 PM   #6
 
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For some reason Audacity doesn't want to use 44.1 kHz as the sample rate. I've checked the properties of the MP3, and it still says 32 kHz.
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Old Oct 26th, 2005, 01:23 PM   #7
 
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Quote:
Originally Posted by Vortech
For some reason Audacity doesn't want to use 44.1 kHz as the sample rate. I've checked the properties of the MP3, and it still says 32 kHz.
1.) Export your Audacity project to a WAV or AIFF file (depending on whether you're using Windows or Mac).

2.) Close that project.

3.) Go into Audacity's preferences and set the default sample rate for a project to 44.1 kHz.

4.) Open a new project.

5.) Import your sound file from Step 1 into the new project.

6.) Export as MP3.

7.) Double-check the MP3.

If that sounds like too much work, you could instead use LAME from the command line to create your MP3, and convert the sample rate when you do it:
Code:
lame -s 44.1 -b 80 whatever.wav whatever.mp3
Either way, it might come out sounding a tiny bit off if you listen closely. Neither Audacity nor LAME are likely to have world-class sample rate converters; for that you need to spend real money. However, it won't sound like a chipmunk.

And in the future, when you create new projects in Audacity, make sure the default project sample rate is set at 44.1.
[/code]
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Old Oct 26th, 2005, 01:24 PM   #8
 
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Actually, I just noticed that iTunes can change the sample rate if I import it to an MP3 that way. I'll just save it as a high-quality AIFF in Audacity and change it to an MP3 with a bitrate of 80kbps and a sample rate of 44.1 kHz in iTunes.
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Old Oct 26th, 2005, 02:16 PM   #9
 
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Old Oct 27th, 2005, 03:37 PM   #10
 
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