[Icecast] streaming to dialup users gives low quality audio

Greg J. Ogonowski greg at orban.com
Sat Dec 31 02:21:39 UTC 2005

Until aacPlus, it was impossible to get decent sounding dial-up audio 
quality on an Internet audio stream.  aacPlus changes everything.

The reason why broadband audio streams happened was to improve the 
quality, since at the time it too 128kbps of MP3 to do it.  Ogg 
Vorbis reduces this to about 64kbps, but still not good enough for 
dial-up.  Broadband is no long absolutely necessary for 
"entertainment grade" audio.

Here is a 32kbps stereo stream as proof:
Listen using Winamp or foobar2000.


At 17:46 2005-12-30, Geoff Shang wrote:
>Dave wrote:
>>   I've got two streams, one for broadband, one for dialup. Well, 
>> having had occation to use a dialup connection recently i checked 
>> the dialup stream. Although it was streaming what the broadband 
>> stream was, the audio quality was audibly worse. It didn't buffer, 
>> but it didn't sound as clear as the broadband stream.
>This is expected.  If dialup streams sounded as good as broadband 
>streams, everyone would just use dialup streams.
>>I used lame to encode the tracks to mp3 and used it's standard 
>>preset while doing it. In my ices.conf file for the dialup stream i 
>>originally had a samplerate of 22050, two chanels, and a bitrate of 24.
>This is going to sound pretty horrible, since MP3 can't really do 
>22.05kHz at 24kbps stereo.
>>I changed the bitrate up to 56, which resulted in a noticeable 
>>audio increase in quality but the buffering was unacceptable. If 
>>anyone has settings that work i would be interested in hearing about them.
>You're not going to be able to send 56kbps over a modem, as has been 
>stated already.  IMHO, 40kbps would probably be your absolute top 
>for a 56k modem. If you want to be accessible to 33.6/28.8k modems, 
>don't go any higher than 24kbps.
>At this rate, you hit the stereo vs mono argument.  If you want 
>clearer audio, go mono.  But if you want stereo, you'll have poorer audio.
>In my experience, you can get acceptable audio at 24kbps mono with 
>22.05kHz or 24kbps stereo with 11.025kHz.  Note that these aren't 
>the LAME default sampling rates for these bit rates.
>If you want to go 40kbps, some quick informal testing gives the 
>following results:
>Stereo: LAME default is 16kHz, but to my ears, you can get 
>acceptable results at 22.05kHz.
>Mono. LAME default is 24kHz, but 32kHz sounds just fine.  44.1kHz is 
>starting to push it, but since it gets rolled off anyway, there's 
>really no point in going that high.
>At 32kbps, LAME's defaults are 16kHz stereo and 22.05kHz mono, and 
>these are what I'd probably recommend.
>This is just from some quick informal testing, you should probably 
>listen yourself and see what you like.
>For quick and dirty testing, I used:
>lame --quiet -b <bitrate> [-a] [--resample <samplerate>] 
><infile.wav> - |mpg123 -
>Where -a is for mono, and --resample is for changing the sample rate 
>from the default.
>Oh and you should use a relatively recent LAME release like 
>3.96.  Versions prior to 3.93 or so had noticeably poorer audio at 
>lower sampling rates.
>Of course, other codecs will perform better.  I'd advocate for Ogg 
>Vorbis myself.
>Hope this helps,
>Icecast mailing list
>Icecast at xiph.org

Greg J. Ogonowski
VP Product Development
1525 Alvarado St.
San Leandro, CA  94577  USA
TEL +1 510 351-3500
FAX +1 510 351-0500
greg at orban.com

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