Icecast - Dec 2005 - streaming to dialup users gives low quality audio

Hello,
    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. 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. 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.
Thanks.
Dave.

This doesn't specifically answer your question, but is it possible for
you to use an ogg stream?

I've been running a stream using a 48Kbps ogg (i'm sure I could push
it lower too...) and it sounds to me and other people pretty much just
as "nice" as a 128Kbps mp3.  We can't notice it being audibly
worse.

-Daniel
On 12/30/05, Dave <dmehler26@woh.rr.com> wrote:
> Hello,
>     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. 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. 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.
> Thanks.
> Dave.
>
> _______________________________________________
> Icecast mailing list
> Icecast@xiph.org
> http://lists.xiph.org/mailman/listinfo/icecast
>

--
Daniel Ballenger
http://denetron.com
Sr. Systems Administrator - Denetron LLC

This is another reason aacPlus is an advantage.
It is supported using Orban Opticodec-PC Streaming Encoder, Icecast2 
Servers and Winamp.

http://www.orban.com/orban/products/stream/1010_overview.html

Icecast2 Eval Stream here:

http://www.orban.com/orban/products/stream/orban_eval_ice.pls

-greg.



At 11:05 2005-12-30, Dave wrote:
>Hello,
>    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. 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. 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.
>Thanks.
>Dave.
>
>_______________________________________________
>Icecast mailing list
>Icecast@xiph.org
>http://lists.xiph.org/mailman/listinfo/icecast

__________________________________________________________________________
Greg J. Ogonowski
VP Product Development
ORBAN / CRL, Inc.
1525 Alvarado St.
San Leandro, CA  94577  USA
TEL +1 510 351-3500
FAX +1 510 351-0500
greg@orban.com
http://www.orban.com

Hi,
    Currently streaming ogg isn't practical in this situation. That was one 
of the first things i checked into. WHen i looked i didn't see a streamer 
that did both ogg and mp3.
Thanks.
Dave.

----- Original Message ----- 
From: "Daniel Ballenger" <lpmusix@gmail.com>
To: "Dave" <dmehler26@woh.rr.com>
Cc: <icecast@xiph.org>
Sent: Friday, December 30, 2005 2:23 PM
Subject: Re: [Icecast] streaming to dialup users gives low quality audio

> This doesn't specifically answer your question, but is it possible for
> you to use an ogg stream?
>
> I've been running a stream using a 48Kbps ogg (i'm sure I could
push
> it lower too...) and it sounds to me and other people pretty much just
> as "nice" as a 128Kbps mp3.  We can't notice it being audibly
worse.
>
> -Daniel
> On 12/30/05, Dave <dmehler26@woh.rr.com> wrote:
>> Hello,
>>     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. 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.
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.
>> Thanks.
>> Dave.
>>
>> _______________________________________________
>> Icecast mailing list
>> Icecast@xiph.org
>> http://lists.xiph.org/mailman/listinfo/icecast
>>
>
>
> --
> Daniel Ballenger
> http://denetron.com
> Sr. Systems Administrator - Denetron LLC
>

Anybody on this list know how to create dynamic relaying with icecast
servers?

I have an icecast server and I would like to give my users the ability to
stream their content from anywhere they are using an username and password
combination.

I have researched the web and have found references to rtptools, I know I
should be able to set up a port to listen on and reroute that stream to the
icecast server, but when I try it I get nothing.

Any suggestions, hints or examples would be appreciated.

There are two things that caught my attention in this post.

When you say 22050 kHz sampling rate, two channels and bitrate 24. Do you
mean 24 kbps stereo (which would be 12 kbps per channel, i.e. real bad audio
quality), or 48 kpbs stereo (24 kbps for each channel).

When you talk about dial-up, do you mean analog dial-up that is limited to
56 kbit/sec? In that case streaming at 48kbps is pushing it since many
dial-up connections won't reach that speed because of bad line quality. The
safest way to go is to stream in mono and don't push the bitrate beyond
24kbps or 32kbps if you really must.

KJ

-----Oorspronkelijk bericht-----
Van: icecast-bounces@xiph.org [mailto:icecast-bounces@xiph.org]Namens
Dave
Verzonden: vrijdag 30 december 2005 20:05
Aan: icecast@xiph.org
Onderwerp: [Icecast] streaming to dialup users gives low quality audio


Hello,
    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. 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. 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.
Thanks.
Dave.

_______________________________________________
Icecast mailing list
Icecast@xiph.org
http://lists.xiph.org/mailman/listinfo/icecast


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Dave wrote:
> Hello,
>   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,

Geoff.

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:
http://www.orban.com/orban/products/stream/orban_eval_ice.pls
Listen using Winamp or foobar2000.

-greg.



At 17:46 2005-12-30, Geoff Shang wrote:
>Dave wrote:
>
>>Hello,
>>   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,
>
>Geoff.
>
>_______________________________________________
>Icecast mailing list
>Icecast@xiph.org
>http://lists.xiph.org/mailman/listinfo/icecast

__________________________________________________________________________
Greg J. Ogonowski
VP Product Development
ORBAN / CRL, Inc.
1525 Alvarado St.
San Leandro, CA  94577  USA
TEL +1 510 351-3500
FAX +1 510 351-0500
greg@orban.com
http://www.orban.com