I am involved with a project to help community groups in the UK webcast community events (voice) called Democaster - http://democaster.org

We are adding an embedded Flash player to play our "on-demand" versions of our MP3 files as they download in the background (we also use Ogg). We prefer to use open source solutions.

1. Low bitrate issues - we've read about encoding/Flash issues with lower bitrate MP3 playing like the "chipmunks." Our MP3 files will be in the 16 to 32 kbps range. Any advice on encoder settings, sample rates etc. to work with Flash's ability to play MP3 files?

2. Which embedded Flash player do you recommend? Ideally we'd have the ability to fast forward or move around in the webcast like you can with typical client player and a downloaded file.

Thanks,
Steven Clift

One update. I used Audacity, exported to MP3, connected them to EMFF: http://www.marcreichelt.de/spezial/musicplayer/#english to create/play some sample lower bit rate recordings.

They are here: http://www.publicus.net/mp3.html

Any advice on why the 16 kbps, 11KHz file plays like the chipmunks?

The 16kbps, 22KHz works and the 24kbps/11 Khz sounds pretty good. Unlike music oriented audiophiles our goal is to smash as much "accessible" audio into the smallest files listenable. We want a one hour meeting to come in under 10MB an hour.

Also, I bumped into this advice:
http://www.1pixelout.net/code/audio-player-wordpress-plugin/

The "chipmunk" effect
The Macromedia Flash player has a problem playing files that are encoded at a rate that is not a multiple of 11.025 kHz. This effect is sometimes called the “chipmunk” effect: the file is played at double speed. To avoid this, encode MP3’s at 11.025 kHz 22.050 kHz or 44.100 kHz.


Steve

Would a voice codec (i.e. Speex) be of use for this application?

http://wiki.hydrogenaudio.org/index.php?title=Speex

I know little about it and thus have no idea if it works with flash. Just throwing it out there!

I'm pretty sure iTunes has a "speech" encoding setting for podcasts, so you may want to look into that also.

Perhaps there's a way to tune LAME/MP3 to narrow the bandwith to the range of speech frequencies (lowpass/highpass filters), thus saving a ton o' bits (& bitrate). As i'm only a newbie/novice I'll leave this one to the experts.

Good luck regardless!

Had a similar requirement ages ago and, IIRC, this helped a lot. Don't have the exact parameters I used at hand, but I can try and find 'em for you if needed.

HTH.

Alessandro

[Edit]Here it is (maybe the experts will confirm or reject it - Anyway, keep in mind this was around June 2003):
--alt-preset cbr 16 -a --resample 11 --lowpass 5 --athtype 2 -X3