on the first dial 0 = 10

I got caught with that as well.

Robert
TLC Radio

On 19 Sep 2006 at 23:16, Geoff Barkman wrote:

> Hi There
> Is anyone familiar with Broadtech LPFM transmitters? I have a friend that has
> one. Adjusting the output frequency consists of 3 dials numbered 0 to 9. First
> is 10's ....second 1's.... and third is the 0.1's. So, to do the low part of
the
> guardband frequencies it is quite logical 88.1 - 88.7 etc.....
>
> Anyone know how to set it, so it can do frequencies above 99.9 MHz? That way
we
> can use the top guard band 106.7 - 107.7 MHz. Thanks in advance. Cheers Geoff
> Barkman 	 ZL4TUX --
>
>
> ---------------------------------------------------------
> LPFM Website: http://au.groups.yahoo.com/group/LPFM_Radio
> Yahoo! Groups Links
>
>
>
>
>
>
>
>
>

----- Original Message -----

From: Geoff Barkman

To: LPFM_Radio@...

Sent: Tuesday, September 19, 2006 11:16 PM

Subject: [LPFM] Broadtech Transmitters

Hi There
Is anyone familiar with Broadtech LPFM transmitters? I have a friend that has
one. Adjusting the output frequency consists of 3 dials numbered 0 to 9.
First is 10's ....second 1's.... and third is the 0.1's. So, to do the low
part of the guardband frequencies it is quite logical 88.1 - 88.7 etc.....

Anyone know how to set it, so it can do frequencies above 99.9 MHz? That way
we can use the top guard band 106.7 - 107.7 MHz.
Thanks in advance.
Cheers Geoff Barkman                  ZL4TUX
--

Does anyone here have a Headphone amp they would like to sell?

Regards,
Ethan

Hi There
Is anyone familiar with Broadtech LPFM transmitters? I have a friend that has
one. Adjusting the output frequency consists of 3 dials numbered 0 to 9.
First is 10's ....second 1's.... and third is the 0.1's. So, to do the low
part of the guardband frequencies it is quite logical 88.1 - 88.7 etc.....

Anyone know how to set it, so it can do frequencies above 99.9 MHz? That way
we can use the top guard band 106.7 - 107.7 MHz.
Thanks in advance.
Cheers Geoff Barkman 	 ZL4TUX
--

Does anyone have a contact number for Phil Grey of Community Radio Hamilton?

If so, can you please contact me off-list.

Hi

Yes we are interested..if the price is right. for historical reasons.

Call us  06-845-3888 for negotiations.

Cheers

Graham J Barclay
graham@...
SOUNDWAVE FM
( Broadcasting since Oct 3rd 1997 )
P O Box 3103
Onekawa
Napier 4142
New Zealand
Ph: 0064-6-845-3888
Cell : 025-206-7191 ( NZ Daytime Only )
http://www.soundwavefm.co.nz

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--- In LPFM_Radio@..., "BROTHER MAJIK" <broluv@s...>
wrote:
>
> HI EVERYONE - A WHILE BACK I AQUIRED THE CART MACHINES, CART TAPES,
CART TAPE CHROME CARASOLE, ELECTRIC ERASER ETC THAT WERE ORIGINALLY
USED BY "RIVERCITY RADIO 2ZW" IN WANGANUI - THEY ARE FOR SALE IF ANY
ONE IS INTERESTED IN BUYING SOME OF OR ALL OF THEM - PRICES ARE
NEGOTIABLE - IF INTERESTED PLEASE GIVE ME A CALL - HAVE A GREAT DAY -
STEVE - WANGANUI - PHONE 0800 625451
>

hi - it's pretty hard to say! - steve
----- Original Message -----
From: "philip_crookes" <philip@...>
To: <LPFM_Radio@...>
Sent: Friday, September 15, 2006 6:48 PM
Subject: [LPFM] Re: STEVE BROTHER MAJIK - BROTHERLUV FM 106.7


> --- In LPFM_Radio@..., "BROTHER MAJIK" <broluv@s...> wrote:
> >
> > HI EVERYONE - A WHILE BACK I AQUIRED THE CART MACHINES, CART TAPES,
> CART TAPE CHROME CARASOLE, ELECTRIC ERASER ETC THAT WERE ORIGINALLY
> USED BY "RIVERCITY RADIO 2ZW" IN WANGANUI - THEY ARE FOR SALE IF ANY
> ONE IS INTERESTED IN BUYING SOME OF OR ALL OF THEM - PRICES ARE
> NEGOTIABLE - IF INTERESTED PLEASE GIVE ME A CALL - HAVE A GREAT DAY -
> STEVE - WANGANUI - PHONE 0800 625451
> >
> Does anyone use cart machines any more?
>
> And is there anything interesting on the carts worth keeping? Jingles,
> announcements, historic stuff that will disappear unless someone takes
> care of it?
>
> Philip
> Primetime 1ZZ
> www.primetimeradio.co.nz
>
>
>
>
>
>
>
> ---------------------------------------------------------
> LPFM Website: http://au.groups.yahoo.com/group/LPFM_Radio
> Yahoo! Groups Links
>
>
>
>
>
>
>

--- In LPFM_Radio@..., "BROTHER MAJIK" <broluv@s...> wrote:
>
> HI EVERYONE - A WHILE BACK I AQUIRED THE CART MACHINES, CART TAPES,
CART TAPE CHROME CARASOLE, ELECTRIC ERASER ETC THAT WERE ORIGINALLY
USED BY "RIVERCITY RADIO 2ZW" IN WANGANUI - THEY ARE FOR SALE IF ANY
ONE IS INTERESTED IN BUYING SOME OF OR ALL OF THEM - PRICES ARE
NEGOTIABLE - IF INTERESTED PLEASE GIVE ME A CALL - HAVE A GREAT DAY -
STEVE - WANGANUI - PHONE 0800 625451
>
Does anyone use cart machines any more?

And is there anything interesting on the carts worth keeping? Jingles,
announcements, historic stuff that will disappear unless someone takes
care of it?

Philip
Primetime 1ZZ
www.primetimeradio.co.nz

Hi Gavin (and others interested)

A dominator is an excellent hardware brickwall limiter. You can drive
it as hard as you want and it will be very transparent. If you
like/prefer this hardware, I highly suggest using a dominator for
peaking management, but a compellor (or even another Dominator) for
compression (or AGC equivalent).

Based on my experience, your first line of processing should be about
levelling the incoming audio. As a starting point, suss your starting
gain on the Compellor (or Dominator) by playing a normalised music
track feeding in. Start with your release time at halfway, compression
at 2:1 (if applicable).

Your incoming audio will appear on the LED, adjust your gain till it
'just' appears. This is a fair indication that your input is coming in
at 0db.

We are looking for levelling, so a thrust of 10 to 14db (or -10 to
-14db if you're looking upside down) is about right for an AGC
equivalent. Have your release at around 2-4 seconds. If your LED is
fully lit, increase your compression ratio to between 2:1 and 2.5:1.
If no visible reduction, reduce it back to 2:1 and decrease your gain.

After this, using a Dominator, you can either lean toward compressing
a few more db, which you can do by setting release to slow and
increase your gain somewhat - but not too much!. This will keep things
dynamic. Or you can lean toward hardcore limiting which requires
faster release, and push the gain to where you feel comfortable. Steer
away from pumping it too much with fast release.

They are both robust pieces of hardware and may take a lot of fiddling
to perfect - but remember to have fun if you plan on using them, as
thats what they'll give you, and be patient. Exercise the "leave it a
day" rule.

Ricky Huntington might even post some of his thoughts if he's asked?

cheers


--- In LPFM_Radio@..., "Gavin Stephens"
<kiwi_rock@x...> wrote:
  if you've got PC's to spare.
>
> Although I was wondering if anyone out there uses both the aphex
compellor/dominator together, how hard they drive the gain reduction
on their dominator. Also who's had more sucess with either the slow or
fast leveling action.
>

----- Original Message -----

From: Ross Levis

To: LPFM_Radio@...

Sent: Wednesday, September 13, 2006 7:50 PM

Subject: Re: [LPFM] Audio Normalising, interesting link saying no to 100%

I tend to normalize to 98% because I heard of something similar a long time ago.  Although I think it was some issue with encoding to MP3 or Ogg which could cause distortion on 0db samples.

----- Original Message -----

From: Gavin Stephens

To: lpfm_radio@...

Sent: Wednesday, September 13, 2006 5:25 PM

Subject: [LPFM] Audio Normalising, interesting link saying no to 100%

This is an interesting article on why not to normalise audio to 100% digital or 0dB after ripping from CD. Because the output can easily be above 0dB once decoded back to analogue...

 

http://www.audioholics.com/techtips/specsformats/0dbfsdigitalplayback.php

 

Cheers,

Gavin.

---

No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date: 9/12/2006

----- Original Message -----

From: Gavin Stephens

To: lpfm_radio@...

Sent: Wednesday, September 13, 2006 5:25 PM

Subject: [LPFM] Audio Normalising, interesting link saying no to 100%

This is an interesting article on why not to normalise audio to 100% digital or 0dB after ripping from CD. Because the output can easily be above 0dB once decoded back to analogue...

 

http://www.audioholics.com/techtips/specsformats/0dbfsdigitalplayback.php

 

Cheers,

Gavin.

----- Original Message -----

From: Gavin Stephens

To: Islay and Brian Gallagher

Cc: lpfm_radio@...

Sent: Wednesday, September 13, 2006 3:10 PM

Subject: [LPFM] Re: Re: streaming broadband a better option

Hi Brian, (cc'd to LPFM since the stuff after the first paragraph may be of
interest, so excuse the first paragraph others).

House hunting is still in action. We've been offered a no pressure private
sale from friends who own an older investment property (state housing)
that's in tip top shape and in a good part of the small town here. Kid's
playground two doors down... Perfect for our little one. The good thing is
the property is still tenanted so we can take as long as we want to get
around to moving. The next best thing, is that the private sale will be
probably under registered valuation for us, but they would sell for much
more if they listed it to someone they don't know. So all is falling in to
place very well. But we're not tied to it if something else comes up. But
we're off to Aussie to visit family in December so will probably now wait
until we get back to move, as house listings are picking up rapidly with
this good weather so I'm sitting back a little more now. Although I do have
other still in my mind out of town, because it has line of sight to all the
surrounding towns! hehehehehehe.


As for metering... I was bought up on VU's in an access radio station in
Invercargill from 15 years old to ummm 19 I think. Certainly not the amount
of experience of your engineering with them.

Then production became LED based. I certainy liked LED metering and I still
do to some degree. As it stops those for example at (they shall go un named)
big commercial radio stations who do presenting but don't understand
engineering much, from killing their digital processors. I've walked in to a
few of these guys owned stations to find led metering a good way to stop
high peak levels, but those with VU's still seem to drive them off the scale
as if the peak light on some VU meters means it's ON AIR.

But because everything is 16bit digital (-96dB maximum noise floor or 6dB
for every bit) I've always taken audio from CD's and left them at close to
0dBFS. I have wanted to normalise audio down to the studio and movie
recordings standard of -20dBFS in reference to 0VU for the past few years.
So that way everything coming out of a PC drives VU metering around 0dB and
doesn't work AGC leveling gear so hard. For example there's such wide
varying audio from say light compressed 1970's stuff to hyper-compressed
2000 stuff. Of course, they both show the same peak level metering on the PC
and on LED based metering audio consoles... which really doesn't tell the
operator much about the average audio energy before it's going in to audio
processing gear. Unless you don't listen off air, but rather directly to the
consoles programme output as you mentioned.

The other reason for switching to VU based in the digital domain instead of
peak based, is that at least then real peaky stuff is allowed to drive an
on-air limiter harder, and low peaky (I should I say lack of dynamic range
hyper compressed) stuff doesn't drive limiters just as hard so layouts for
audio processing gear such as AGC's can be avoided until a later date also.

I know that AGC's and compressors can be used to operate on RMS rather than
just peak, and quite effectively. But it's so much easier setting up
processing gear when everything going in to them is VU or RMS based. That
way AGC's don't need emergency release/attack options in order to go from a
hyper-compressed song to a wide dynamic song without a huge noticable quick
change on-air.

24-bit audio really comes in to play with my thoughts about going VU based.

I remember when DAT's and DigiCarts first came out. As far as the DigiCart
was concerned, the station I worked at - at the time had the production
consoles LED meters at 0dB in order to have the DigiCart peaking at -18dBFS
where it's reference level marked printed on the front laid. This
misunderstanding view was flawed as I've just realised. Because -20/18dBFS
was sopose to be in reference to 0VU output (not peak output), so there was
headroom for peaks. It wasn't sopose to be used as the peak value when
recording on to DigiCart etc... which just mean't 18dB more noise and less
dynamic range. That's a lot when every 3dB is double the enegry - watts.
Unless it's 6dB for double the audio energy since 6dB is double in voltage?
I forget now, but think it's the 3dB theory.

This also explains why my satellite receiver at home, when hooked up
digitally to my PC the audio levels only peak around -6dB, because they are
using the -20/18dBFS = 0VU standard. The same goes for the old IRN - now
Newstalk ZB Affliates Unit FTP downloadble newscheck and sportscheck
bulletins. I used to swear black and blue with their technical department to
normlise their audio to -0dBFS since CD's were like that. But they said on
their metering it's was driving at +10dB to get that result and no tech in
Auckland could figure out what my metering was showing their audio levels so
low. In the end I wrote some computer batch files to download the news off
the net, decode it, normalise it, tag it etc... and import it for use in an
automation system. Much cheaper than buying software that could only do half
the job.

But now that I look back on it, it's because they gear they were using was
probably and still is all VU based, without a limiter in front of it because
there's no need for a limiter if you follow the -20dBFS (note not PPM but
decibels of full scale digital - absolute peak like in cool edit pro) in
referene to 0VU standard.

The biggest downfall is to use this in the 16 bit world, the signal to noise
ratio starts deminishing to 76dB instead of 96dB. At least 24-bit gives you
the same as high sonic analogue gear at around 110dB like some Aphex gear
(or 144dB in the digital world before conversion). So I figure I can convert
CD's to 24 bit, then only after converting them first, reduce them down
to -20dB RMS without reducing the dynamic range.

BTW... the sound blasters are a good card as far as being not so noisey as
you've foundwith around -80dB noise floor. And yes, built in audio is always
terrible! Wow you're getting -36dB noise floor, that's ummmm, I'll keep
those comments to myself! My cheap $40 card does -56dB. Yes external sound
cards are ok. Make sure you have a fast enough USB port to utilise it
though. USB version 1.1 on older PC's including my Pentium 1.5GHz is only
12Mbps. USB 2.0 on new PC's is about 480Mbps. Needless to say there's a big
difference in how it will handle USB audio. Sound Blasters are almost all
5.1 channel cards so require some bandwidth when under hevy load.

Some sound cards are designed to run at the maximum potental at 24-bit or
16-bit 48KHz with older one's instead of 44.1. For example, my el-cheap-o
sound card is noisey at 44.1KHz, but almost quiet when recording at 48KHz.
So I recorded audio at 48KHz any vocal stuff for spots and down converted it
to 44.1KHz before I found the world of S/PDIF. Now audio enters and
perferrably exits my sound card via digital, and the conversion takes place
elsewhere. So any cheap card does a good job now.

Talking more about sound cards...

When I installed an automation system for independent commercial once I had
fun with a visa card and a lets do it right the first time round boss. I got
to install an Antex card which had mp2 decoding on board. But the noise
floor of the cards was just beautiful. The AudioScience entry level wave
card starts around $500US, but they are actually designed better than
domestic cards hands down. Some people say why pay that? Well here's a few
reasons. They have 4 seperate hardware based and buffered sound cards in one
PCI card (even though the low end one's still mix to a single analogue
output). Audio mixing and buffering is all done on the card so the compute
has some breathing space, and software has access to the 4 seperate play
streams as seperate devices so you can fade audio on playback deck for
example, without effecting audio of the other.

But here's the real difference...

Unfortunately today's domestic sound cards, if used with for example windows
nt where old school wave driver are used (not WMD or directsound or
whatever) they actually can't mix two audio streams on the sound card
together. But everyone goes out and feed's the domestic sound cards with
multiple audio sources right to the edge of digital clipping (which they do
when mixed if their levels are both to high at the same time). DirectSound
and domestic driver sound cards mix audio streams in software/drivers before
sending the stream to the hardware/sound card so you can have multiple sound
effects on games etc... coming out of the sound card. When in reality, they
are not designed for multiple audio stream playback without distortion to a
degree. It's like wiring two-three professional audio cables together
peaking at +14dB each. Distortion occurs with them all playing at once.
Only, in the digital world, 0dB is 0dB, not the slightly bit over sounds
nice. Yet we do this without thought a lot of the time in LPFM operations
while accept 0dBFS CD recordings are a good level... which is just the my CD
sounds louder than yours sales marketting world that's got that idea in to
everyone's head.

Some commercial stations who can't afford 3 or 4 stream wave driver driven
sound cards choose to normalise all their audio to -3 or -6dBFS or use a
couple of sound blasters and alternative their audio output in order to
avoid any undesirable audio effects when mixing multiple streams together.
Although for LPFM, most people just have a domestic sound card with no
budget what so ever, been there done that and probably will do it again very
soon. But if you can shell out for a $1,000 ever on a compressor, get a $800
AudioScience wave sound card. Even if you are using MP3's without the
expensive onbaord decoding they are still built to handle it, and what's
more, it's balanced professional audio level output, immediately decreasing
the noise floor of the sound card and studio interference. With domestic
cards, they're -10dB line level. Although half of them 0 is 0 these days.
But 0 is not +4dB or +8dB. So you've got to add gain when driving a balanced
audio mixer or when eventually getting to the transmitter. Which increases
the noise floor.

These are just my own personal opinions and experience with sound cards,
audio level recordings etc... and thoughts about switching to 24 bit. For
the next door neighbour they probably don't care. But if any of this shares
some light on topics and has someone thinking twice about what they
understood about sound cards and levels etc... my job is done, to help
educate in this area.

Although I'm not completely on the play with 16 to 24 bit conversion. I'm
pretty sure though if I normalise a 16 file convereted to 24 bit 20dB lower,
I don't reduce the dynamic range or increase the noise floor. Not til I
reduce it by a further 48dB which I have no desire to do.

Gavin.

PS: Brian, Peavey compressors? I used a few old I think 1950-60 relics, PYE
wideband compressors with very noise capcitors in them for a while. They
used to use them at TVNZ for TV transmitters audio control I think. Nothing
fancy and terrible peak control, but good average compression/leveling. They
even had a noise gate, not silence gate.

---

No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/446 - Release Date: 9/12/2006

Hi Brian, (cc'd to LPFM since the stuff after the first paragraph may be of
interest, so excuse the first paragraph others).

House hunting is still in action. We've been offered a no pressure private
sale from friends who own an older investment property (state housing)
that's in tip top shape and in a good part of the small town here. Kid's
playground two doors down... Perfect for our little one. The good thing is
the property is still tenanted so we can take as long as we want to get
around to moving. The next best thing, is that the private sale will be
probably under registered valuation for us, but they would sell for much
more if they listed it to someone they don't know. So all is falling in to
place very well. But we're not tied to it if something else comes up. But
we're off to Aussie to visit family in December so will probably now wait
until we get back to move, as house listings are picking up rapidly with
this good weather so I'm sitting back a little more now. Although I do have
other still in my mind out of town, because it has line of sight to all the
surrounding towns! hehehehehehe.


As for metering... I was bought up on VU's in an access radio station in
Invercargill from 15 years old to ummm 19 I think. Certainly not the amount
of experience of your engineering with them.

Then production became LED based. I certainy liked LED metering and I still
do to some degree. As it stops those for example at (they shall go un named)
big commercial radio stations who do presenting but don't understand
engineering much, from killing their digital processors. I've walked in to a
few of these guys owned stations to find led metering a good way to stop
high peak levels, but those with VU's still seem to drive them off the scale
as if the peak light on some VU meters means it's ON AIR.

But because everything is 16bit digital (-96dB maximum noise floor or 6dB
for every bit) I've always taken audio from CD's and left them at close to
0dBFS. I have wanted to normalise audio down to the studio and movie
recordings standard of -20dBFS in reference to 0VU for the past few years.
So that way everything coming out of a PC drives VU metering around 0dB and
doesn't work AGC leveling gear so hard. For example there's such wide
varying audio from say light compressed 1970's stuff to hyper-compressed
2000 stuff. Of course, they both show the same peak level metering on the PC
and on LED based metering audio consoles... which really doesn't tell the
operator much about the average audio energy before it's going in to audio
processing gear. Unless you don't listen off air, but rather directly to the
consoles programme output as you mentioned.

The other reason for switching to VU based in the digital domain instead of
peak based, is that at least then real peaky stuff is allowed to drive an
on-air limiter harder, and low peaky (I should I say lack of dynamic range
hyper compressed) stuff doesn't drive limiters just as hard so layouts for
audio processing gear such as AGC's can be avoided until a later date also.

I know that AGC's and compressors can be used to operate on RMS rather than
just peak, and quite effectively. But it's so much easier setting up
processing gear when everything going in to them is VU or RMS based. That
way AGC's don't need emergency release/attack options in order to go from a
hyper-compressed song to a wide dynamic song without a huge noticable quick
change on-air.

24-bit audio really comes in to play with my thoughts about going VU based.

I remember when DAT's and DigiCarts first came out. As far as the DigiCart
was concerned, the station I worked at - at the time had the production
consoles LED meters at 0dB in order to have the DigiCart peaking at -18dBFS
where it's reference level marked printed on the front laid. This
misunderstanding view was flawed as I've just realised. Because -20/18dBFS
was sopose to be in reference to 0VU output (not peak output), so there was
headroom for peaks. It wasn't sopose to be used as the peak value when
recording on to DigiCart etc... which just mean't 18dB more noise and less
dynamic range. That's a lot when every 3dB is double the enegry - watts.
Unless it's 6dB for double the audio energy since 6dB is double in voltage?
I forget now, but think it's the 3dB theory.

This also explains why my satellite receiver at home, when hooked up
digitally to my PC the audio levels only peak around -6dB, because they are
using the -20/18dBFS = 0VU standard. The same goes for the old IRN - now
Newstalk ZB Affliates Unit FTP downloadble newscheck and sportscheck
bulletins. I used to swear black and blue with their technical department to
normlise their audio to -0dBFS since CD's were like that. But they said on
their metering it's was driving at +10dB to get that result and no tech in
Auckland could figure out what my metering was showing their audio levels so
low. In the end I wrote some computer batch files to download the news off
the net, decode it, normalise it, tag it etc... and import it for use in an
automation system. Much cheaper than buying software that could only do half
the job.

But now that I look back on it, it's because they gear they were using was
probably and still is all VU based, without a limiter in front of it because
there's no need for a limiter if you follow the -20dBFS (note not PPM but
decibels of full scale digital - absolute peak like in cool edit pro) in
referene to 0VU standard.

The biggest downfall is to use this in the 16 bit world, the signal to noise
ratio starts deminishing to 76dB instead of 96dB. At least 24-bit gives you
the same as high sonic analogue gear at around 110dB like some Aphex gear
(or 144dB in the digital world before conversion). So I figure I can convert
CD's to 24 bit, then only after converting them first, reduce them down
to -20dB RMS without reducing the dynamic range.

BTW... the sound blasters are a good card as far as being not so noisey as
you've foundwith around -80dB noise floor. And yes, built in audio is always
terrible! Wow you're getting -36dB noise floor, that's ummmm, I'll keep
those comments to myself! My cheap $40 card does -56dB. Yes external sound
cards are ok. Make sure you have a fast enough USB port to utilise it
though. USB version 1.1 on older PC's including my Pentium 1.5GHz is only
12Mbps. USB 2.0 on new PC's is about 480Mbps. Needless to say there's a big
difference in how it will handle USB audio. Sound Blasters are almost all
5.1 channel cards so require some bandwidth when under hevy load.

Some sound cards are designed to run at the maximum potental at 24-bit or
16-bit 48KHz with older one's instead of 44.1. For example, my el-cheap-o
sound card is noisey at 44.1KHz, but almost quiet when recording at 48KHz.
So I recorded audio at 48KHz any vocal stuff for spots and down converted it
to 44.1KHz before I found the world of S/PDIF. Now audio enters and
perferrably exits my sound card via digital, and the conversion takes place
elsewhere. So any cheap card does a good job now.

Talking more about sound cards...

When I installed an automation system for independent commercial once I had
fun with a visa card and a lets do it right the first time round boss. I got
to install an Antex card which had mp2 decoding on board. But the noise
floor of the cards was just beautiful. The AudioScience entry level wave
card starts around $500US, but they are actually designed better than
domestic cards hands down. Some people say why pay that? Well here's a few
reasons. They have 4 seperate hardware based and buffered sound cards in one
PCI card (even though the low end one's still mix to a single analogue
output). Audio mixing and buffering is all done on the card so the compute
has some breathing space, and software has access to the 4 seperate play
streams as seperate devices so you can fade audio on playback deck for
example, without effecting audio of the other.

But here's the real difference...

Unfortunately today's domestic sound cards, if used with for example windows
nt where old school wave driver are used (not WMD or directsound or
whatever) they actually can't mix two audio streams on the sound card
together. But everyone goes out and feed's the domestic sound cards with
multiple audio sources right to the edge of digital clipping (which they do
when mixed if their levels are both to high at the same time). DirectSound
and domestic driver sound cards mix audio streams in software/drivers before
sending the stream to the hardware/sound card so you can have multiple sound
effects on games etc... coming out of the sound card. When in reality, they
are not designed for multiple audio stream playback without distortion to a
degree. It's like wiring two-three professional audio cables together
peaking at +14dB each. Distortion occurs with them all playing at once.
Only, in the digital world, 0dB is 0dB, not the slightly bit over sounds
nice. Yet we do this without thought a lot of the time in LPFM operations
while accept 0dBFS CD recordings are a good level... which is just the my CD
sounds louder than yours sales marketting world that's got that idea in to
everyone's head.

Some commercial stations who can't afford 3 or 4 stream wave driver driven
sound cards choose to normalise all their audio to -3 or -6dBFS or use a
couple of sound blasters and alternative their audio output in order to
avoid any undesirable audio effects when mixing multiple streams together.
Although for LPFM, most people just have a domestic sound card with no
budget what so ever, been there done that and probably will do it again very
soon. But if you can shell out for a $1,000 ever on a compressor, get a $800
AudioScience wave sound card. Even if you are using MP3's without the
expensive onbaord decoding they are still built to handle it, and what's
more, it's balanced professional audio level output, immediately decreasing
the noise floor of the sound card and studio interference. With domestic
cards, they're -10dB line level. Although half of them 0 is 0 these days.
But 0 is not +4dB or +8dB. So you've got to add gain when driving a balanced
audio mixer or when eventually getting to the transmitter. Which increases
the noise floor.

These are just my own personal opinions and experience with sound cards,
audio level recordings etc... and thoughts about switching to 24 bit. For
the next door neighbour they probably don't care. But if any of this shares
some light on topics and has someone thinking twice about what they
understood about sound cards and levels etc... my job is done, to help
educate in this area.

Although I'm not completely on the play with 16 to 24 bit conversion. I'm
pretty sure though if I normalise a 16 file convereted to 24 bit 20dB lower,
I don't reduce the dynamic range or increase the noise floor. Not til I
reduce it by a further 48dB which I have no desire to do.

Gavin.

PS: Brian, Peavey compressors? I used a few old I think 1950-60 relics, PYE
wideband compressors with very noise capcitors in them for a while. They
used to use them at TVNZ for TV transmitters audio control I think. Nothing
fancy and terrible peak control, but good average compression/leveling. They
even had a noise gate, not silence gate.

If you're looking specifically for software which will put songs on a
similar output level (so one doesn't seem louder than the others, or
vice-versa), I've found the open-source 'MP3 Gain
(http://mp3gain.sourceforge.net/) works fantastically. It alters the
levelling of the file, without tampering with the integrity or quality
of the file.

It's free, and definately worth a look see. www.godrock.co.nz uses it
for their internet stream if you want to get an idea of it's
capabilities are.

Cheers,
Herb.

--- In LPFM_Radio@..., "Gavin Stephens"
<kiwi_rock@x...> wrote:
>
> Hi guys,
>
> Age old topic but my favourite, audio processing that is. I still
like www.burnill.co.uk software based processing for the price of it,
free. I also the sound of it and configuration abilities. It's PC
based and requires some horses, but check it out if you haven't done
so yet if you've got PC's to spare.
>
> Although I was wondering if anyone out there uses both the aphex
compellor/dominator together, how hard they drive the gain reduction
on their dominator. Also who's had more sucess with either the slow or
fast leveling action.
>
> I still like the Aphex gear, I just really wish they would get
around to revising the compellor and throw in some sort of emergency
AGC release/attack for some songs that have say a 9dB sudden jump in
gain (yes CD's before they were compressed within a decibel of their
life).
>
> I'm curious though if anyone knows of stereo generators or
transmitters that have their own overshot compensated low pass filters
in them, rather than straight composite clippers/hard limiters like in
most transmitters.
>
> Cheers,
> Gavin.
>

If you're looking specifically for software which will put songs on a
similar output level (so one doesn't seem louder than the others, or
vice-versa), I've found the open-source 'MP3 Gain
(http://mp3gain.sourceforge.net/) works fantastically. It alters the
levelling of the file, without tampering with the integrity or quality
of the file.

It's free, and definately worth a look see. www.godrock.co.nz uses it
for their internet stream if you want to get an idea of it's
capabilities are.

Cheers,
Herb.

--- In LPFM_Radio@..., "Gavin Stephens"
<kiwi_rock@x...> wrote:
>
> Hi guys,
>
> Age old topic but my favourite, audio processing that is. I still
like www.burnill.co.uk software based processing for the price of it,
free. I also the sound of it and configuration abilities. It's PC
based and requires some horses, but check it out if you haven't done
so yet if you've got PC's to spare.
>
> Although I was wondering if anyone out there uses both the aphex
compellor/dominator together, how hard they drive the gain reduction
on their dominator. Also who's had more sucess with either the slow or
fast leveling action.
>
> I still like the Aphex gear, I just really wish they would get
around to revising the compellor and throw in some sort of emergency
AGC release/attack for some songs that have say a 9dB sudden jump in
gain (yes CD's before they were compressed within a decibel of their
life).
>
> I'm curious though if anyone knows of stereo generators or
transmitters that have their own overshot compensated low pass filters
in them, rather than straight composite clippers/hard limiters like in
most transmitters.
>
> Cheers,
> Gavin.
>

----- Original Message -----

From: lancefm881

To: LPFM_Radio@...

Sent: Tuesday, September 12, 2006 3:47 PM

Subject: [LPFM] Audio Processing

I Currently Use sam3 which has all audio processing built into it, e.g.
dual band processing and 5 band processors, gates, bass eq's
www.spacialaudio.com

---

No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.12.3/445 - Release Date: 9/11/2006

The radio automation software package I use (OtsDJ) has a dynamics
processor built in.  THere's an AGC, compressor, and finally the
limiter.  Each is fully customisable (specify attack/release settings,
the compressor ratio, knee, etc).

Currently it operates on the entire spectrum but as I understand it
future releases will have a multiband version.

By the way I was very impressed with the Winamp plugin "Sound SOlution"
- a multiband compressor/limiter with a few extra features (eg
pre-emphasis).  Took me ages to customise it to an acceptable setting,
but once done it works a treat.  Only use it for playing MP3s at home.



Gavin Stephens wrote:
> Hi guys,
>
> Age old topic but my favourite, audio processing that is. I still like
> www.burnill.co.uk <http://www.burnill.co.uk> software based processing
> for the price of it, free. I also the sound of it and configuration
> abilities. It's PC based and requires some horses, but check it out if
> you haven't done so yet if you've got PC's to spare.
>
> Although I was wondering if anyone out there uses both the aphex
> compellor/dominator together, how hard they drive the gain reduction on
> their dominator. Also who's had more sucess with either the slow or fast
> leveling action.
>
> I still like the Aphex gear, I just really wish they would get around to
> revising the compellor and throw in some sort of emergency AGC
> release/attack for some songs that have say a 9dB sudden jump in gain
> (yes CD's before they were compressed within a decibel of their life).
>
> I'm curious though if anyone knows of stereo generators or transmitters
> that have their own overshot compensated low pass filters in them,
> rather than straight composite clippers/hard limiters like in most
> transmitters.
>
> Cheers,
> Gavin.
>

I Currently Use sam3 which has all audio processing built into it, e.g.
dual band processing and 5 band processors, gates, bass eq's
www.spacialaudio.com

Yep,

I'm operating a 'YP' directory which allows streaming servers to
dynamically update the directory with information about what streams are
available, and what tracks are currently playing.

This will be available on my website as soon as I launch it. (real soon now!)



> Matt, did you say something about listing these streaming LPFM
> stations somewhere?
>
> Richard
>
>
>
>

Matt, did you say something about listing these streaming LPFM
stations somewhere?

Richard

Try loading a Direttore generated playlist in Winamp (a full day log -
not an hour by hour one), hit shuffle, then resave the playlist over
itself. Refresh the playlist in Direttore. Your separation rules might
slow the repeating.


--- In LPFM_Radio@..., "arubie2000" <arubie@x...> wrote:
>
> Hi
>
>  Just want to hear from anyone using "Direttore Free" in their
> studio.
> Do you have problems with songs repeating every few hours?
> Any ideas on how to stop this happening?
>
> I have about 4 days worth of music in a catagory called "A" but
> songs are replaying within only a few hours and the rotation does
> not seem to be playing all of the songs in the catagory.
>
> There are several other  catagories that should all have very long
> no repeat intervals, all are behaving the same way.
>
> The no repeat distance is set at 300 (was at 56), anyone know what
> this is? Is it hours? or somthing else?
>
> Would be greatful for any help or advise.
>
> Thanks
> Adam
>
> P.S. I will be replacing Direttore in about six months with a proper
> automation system - just need to save the money.
>

Hi

  Just want to hear from anyone using "Direttore Free" in their
studio.
Do you have problems with songs repeating every few hours?
Any ideas on how to stop this happening?

I have about 4 days worth of music in a catagory called "A" but
songs are replaying within only a few hours and the rotation does
not seem to be playing all of the songs in the catagory.

There are several other  catagories that should all have very long
no repeat intervals, all are behaving the same way.

The no repeat distance is set at 300 (was at 56), anyone know what
this is? Is it hours? or somthing else?

Would be greatful for any help or advise.

Thanks
Adam

P.S. I will be replacing Direttore in about six months with a proper
automation system - just need to save the money.

----- Original Message -----

From: Gavin Stephens

To: Islay and Brian Gallagher

Cc: LPFM_Radio@...

Sent: Saturday, September 02, 2006 4:23 PM

Subject: SPAM-LOW: [LPFM] Re: Re: streaming broadband a better option

I thought I'd reply to your personal email Brian and send this to the lpfm
group incase it's of any interest.

Talking about 96Kbps encoding for streaming etc... someone please correct me
if I'm wrong in my understanding on the following encoding technologies and
quality. It's been yonks since I've used audio compression full stop little
own streaming. Keeping in mnd we're talking streaming audio quality here....

44,100 certainy has wider frequency response and better over all sampling
quality than 32000Hz. But if you're going to use 96Kbps encoding etc.. then
96Kbps at 32000Hz MP3 has similar overall mpeg encoding quality as 128Kbps
44100Hz MP3 by sacraficing a little audio bandwidth from 20KHz down to 15Khz
suitable for FM broadcast. It's a give and take balancing act between
quality of the audio bandwidth being encoded, to the lack of bitrate
available to the encoder to acheive an overall perceptual quality on top
given it's bitrate. That's where viriable bitrate is better at making those
calculations. Although viriable bitrate kinda came and went a bit. There's
still some of it around though.

Of course if you're using a lower bandwidth than that, it doesn't make a
huge difference in my opinion as the audio quality is going to be a lot more
distinguishable as Internet sounding anyway. aacplus to me sounds better
than mp3pro (this is just a personal opinion), but both are good for low
bitrate general Internet streaming. But when you start talking 96Kbps+,
mp3pro and aacplus are a little more out of their general designed use.At
96+ normal MP3 (fruanhofer based codes being the nicest sound when feed with
linear pcm) start to come in to their own as they were designed for those
bitrates more than what mp3 pro was designed for.

Then at even higher bitrates such as 256Kbps (the usual audio library
compresson with automation systems that use compression in the expensive big
boys toys world) MP2 sounds even better (again personal opinion and
experience) than MP3 to me and has more hardware available for it when
dealing with more expensive budgets (outside the realm of lpfm). It's also
worth mentioning, 224Kbps and above encoding in MP2/3 is independent stereo
encoding, not joint channel encoding there for modern produced audio cd's
with phase effects don't turn to underwater sounds at those bitrates as with
128-192 etc... Which is why I still cringe at 128-160Kbps library quality
compression. I can understand though the reason for it, budgets, and back
then when it was the in thing for expensive automation systems, hard drives
were very expensive for 10-20 gigs.

Keep in mind my ears still hear up to 20Khz and hear the real bad side
effects of mp3 compression etc... so I'm a fussy trained golden ear who's
spent countless hours with audio processing software and hardware, infact I
don't like like the IT term hardware for real world audio processing gear
that we know of as compressors/limiters/clippers and multiband processors
etc...

I just perfer S's are S's, not F's on-air at the end of the day and a lot of
head room with audio leveling, multiband compression, limiting, clipping
etc... from the digital noise floor or with mpeg, the noise mask. The simple
re-editing of linear PCM I love and avoids any reason to ever have to go
back and re-do an entire library if Internet radio etc... or other encoding
technologies come along that don't sound good transcoded with mpeg. Of
course though hard drive space and dvd writers these days makes linear pcm
the better choice starting from scratch, those brought up on small hard
drives and mpeg still have their habbits.

Gavin.

I thought I'd reply to your personal email Brian and send this to the lpfm
group incase it's of any interest.

Talking about 96Kbps encoding for streaming etc... someone please correct me
if I'm wrong in my understanding on the following encoding technologies and
quality. It's been yonks since I've used audio compression full stop little
own streaming. Keeping in mnd we're talking streaming audio quality here....

44,100 certainy has wider frequency response and better over all sampling
quality than 32000Hz. But if you're going to use 96Kbps encoding etc.. then
96Kbps at 32000Hz MP3 has similar overall mpeg encoding quality as 128Kbps
44100Hz MP3 by sacraficing a little audio bandwidth from 20KHz down to 15Khz
suitable for FM broadcast. It's a give and take balancing act between
quality of the audio bandwidth being encoded, to the lack of bitrate
available to the encoder to acheive an overall perceptual quality on top
given it's bitrate. That's where viriable bitrate is better at making those
calculations. Although viriable bitrate kinda came and went a bit. There's
still some of it around though.

Of course if you're using a lower bandwidth than that, it doesn't make a
huge difference in my opinion as the audio quality is going to be a lot more
distinguishable as Internet sounding anyway. aacplus to me sounds better
than mp3pro (this is just a personal opinion), but both are good for low
bitrate general Internet streaming. But when you start talking 96Kbps+,
mp3pro and aacplus are a little more out of their general designed use.At
96+ normal MP3 (fruanhofer based codes being the nicest sound when feed with
linear pcm) start to come in to their own as they were designed for those
bitrates more than what mp3 pro was designed for.

Then at even higher bitrates such as 256Kbps (the usual audio library
compresson with automation systems that use compression in the expensive big
boys toys world) MP2 sounds even better (again personal opinion and
experience) than MP3 to me and has more hardware available for it when
dealing with more expensive budgets (outside the realm of lpfm). It's also
worth mentioning, 224Kbps and above encoding in MP2/3 is independent stereo
encoding, not joint channel encoding there for modern produced audio cd's
with phase effects don't turn to underwater sounds at those bitrates as with
128-192 etc... Which is why I still cringe at 128-160Kbps library quality
compression. I can understand though the reason for it, budgets, and back
then when it was the in thing for expensive automation systems, hard drives
were very expensive for 10-20 gigs.

Keep in mind my ears still hear up to 20Khz and hear the real bad side
effects of mp3 compression etc... so I'm a fussy trained golden ear who's
spent countless hours with audio processing software and hardware, infact I
don't like like the IT term hardware for real world audio processing gear
that we know of as compressors/limiters/clippers and multiband processors
etc...

I just perfer S's are S's, not F's on-air at the end of the day and a lot of
head room with audio leveling, multiband compression, limiting, clipping
etc... from the digital noise floor or with mpeg, the noise mask. The simple
re-editing of linear PCM I love and avoids any reason to ever have to go
back and re-do an entire library if Internet radio etc... or other encoding
technologies come along that don't sound good transcoded with mpeg. Of
course though hard drive space and dvd writers these days makes linear pcm
the better choice starting from scratch, those brought up on small hard
drives and mpeg still have their habbits.

Gavin.

They pulled the plug on free peering, instead requiring ISPs to purchase
'Domestic Transit' connectivity.... so as long as your ISP buys that, or
has an arrangement with someone who has a mutual peering policy with
TelstraClear (ie, Telecom), then they'll still see it as local traffic.

I don't know the exact details of Xnet's arrangements with their various
providers, but I can confirm that traffic from telstraclear to an Xnet DSL
does not count towards the datacap. (I use Xnet at home)


> I'm not completely up on the play with where Telstraclear are with peering
> these days. I thought they pulled the plug on free peering in Auckland and
> started considering national traffic as transit? Are Xnet signed up with a
> private peering arragement with Telstraclear or are Telstra bent at the
> knees after they big hoo har over de-peering last year or whenever it was?
>
> I'm glad someone is finally looking at NZ streaming solutions though. Just
> more glad it's someone that's fimiliar with the ISP bizz and hooked up
> with lpfm.
>
> Gavin.
>
>   ----- Original Message -----
>   From: Matt Camp
>   To: LPFM_Radio@...
>   Sent: Thursday, August 31, 2006 9:46 PM
>   Subject: Re: [LPFM] Streaming, Broadband, and a better option.
>
>
>
>   You're right... although one could argue that a suitable level AAC+
>   stream is pretty close to cd quality :)
>
>   The ISP in question is in fact Xnet, however that's just who I am
>   suggesting that my clients use for their own broadband connections. I've
>   worked with their technical staff to ensure that traffic to and from my
>   service isn't counted towards the data-caps.
>
>   My own servers are telehoused around the place, with the primary one
>   being hosted in the skytower right next to the peering exchange.
>
>   Telecom, TelstraClear, Xnet and Quicksilver will all count data to/from
>   it as national.
>
>   I used to work for Ihug, Quicksilver, and TelstraClear doing IP network
>   design, so I'm reasonably familiar with such things :)
>
>   Both Xnet and Quicksilver will do 'unlimited' national traffic (which
>   really means "within reason"), however for various mostly-technical
>   reasons, I would suggest using Xnet.
>
>   Gavin Stephens wrote:
>   > The brainwashing conumer sales tricks have made us say without
>   > thinking twice: "cd quality".
>   >
>   > Unlike the "near cd quality" official title perceptual audio coding
>   > has been able to attain for its self.
>   >
>   > I can't help but mention that. I read far to often people mis-quoting
>   > audio coding technologies for cd quality, when most tests use the more
>   > appropriate term "near" cd quality.
>   >
>   > You've got me interested in the details though. Who's the well known
>   > ISP? Either way, we'll soon know. But I'd like to know who it will be
>   > since it will impact on peering and national charges with some ISP's
>   > depending on who it is.
>   >
>   > Cheers,
>   > Gavin.
>   >
>   >
>   >     ----- Original Message -----
>   >     *From:* Matt Camp <mailto:matt@...>
>   >     *To:* LPFM_Radio@...
>   >     <mailto:LPFM_Radio@...>
>   >     *Sent:* Thursday, August 31, 2006 4:16 PM
>   >     *Subject:* [LPFM] Streaming, Broadband, and a better option.
>   >
>   >
>   >     This recent discussion on streaming is quite interesting, even if
> only
>   >     from the view that it highlights that such things are not that
>   >     simple, and
>   >     all sorts of issues can very quickly make it a major headache.
>   >
>   >     It doesn't need to be that way.
>   >
>   >     I own and operate a small company (totally seperate from any of my
>   >     other
>   >     ventures, such as my radio station) which specialised in digital
> media
>   >     streaming. Streaming is something I've been interested in for a
>   >     number of
>   >     years now, and have significant knowledge of.
>   >
>   >     I'm still in the process of launching the company (we haven't even
>   >     launched our website yet), but at this stage I'm willing to let
>   >     this list
>   >     know about one of our products - Streaming services designed for
> LPFM
>   >     stations.
>   >
>   >     Basically I will provide LPFM operators with the ability to send
> their
>   >     audio streams to my hosted streaming servers. This means that you
> only
>   >     need to send one copy of the audio, and I will take care of
>   >     sending it to
>   >     potentially hundreds of simultaneous listeners.
>   >
>   >     I have also come to an arrangement with a well known ISP that will
>   >     allow
>   >     you to stream a CD quality audio feed 24 hours a day, 365 days a
>   >     year, and
>   >     pay no more than $29.95/month for that broadband connection... no
>   >     data cap
>   >     to worry about!
>   >
>   >     Pricing on my streaming services has not yet been finally
>   >     determined, but
>   >     rest assured it's designed to fit the LPFM type budgets... ie,
>   >     bugger all.
>   >
>   >     And here's the deal for people on this list.... I need people to
>   >     test my
>   >     service.
>   >
>   >     In order to get a good test, I'm looking for people who want a
>   >     months free
>   >     streaming services.... basically you sign up, and I don't charge
>   >     you for
>   >     the first month. No obligation to continue beyond that free month
>   >     either
>   >     if it turns out you don't like the service.
>   >
>   >     I will provide all necessary software, guides, and assistance to
>   >     get the
>   >     stream up and running. You will also receive your streams listed
>   >     on our NZ
>   >     Audio stream directory, and we can help you with things such as
>   >     embedded
>   >     players on your website.  You will receive full stats and logs
>   >     detailling
>   >     how many people are listening, where from, what times, etc, etc.
>   >
>   >     Various format and bitrate options are available and we can
> basically
>   >     cater for anything.
>   >
>   >     So yeah.... anyone want some free streaming?
>   >
>   >     Drop me an email off-list if you are interested and I'll discuss
> the
>   >     details with you.
>   >
>   >
>   >     ------------------------------------------------------------------------
>   >     No virus found in this incoming message.
>   >     Checked by AVG Free Edition.
>   >     Version: 7.1.405 / Virus Database: 268.11.7/434 - Release Date:
>   >     30/08/2006
>   >
>   >
>
>
>
>
>
> ------------------------------------------------------------------------------
>
>
>   No virus found in this incoming message.
>   Checked by AVG Free Edition.
>   Version: 7.1.405 / Virus Database: 268.11.7/434 - Release Date:
> 30/08/2006
>
>

----- Original Message -----

From: Brian Gallagher

To: LPFM_Radio@...

Sent: Friday, September 01, 2006 11:26 AM

Subject: Re: [LPFM] apology from wife of regular lpfm group e-mail user..

I was surprised at the comment but no offence taken/Perhaps we were pushing it a bit but the matter under discussion was of interest to others and doing it in a public forum such as this brought it out into the open and attracted other worthwhile responses.

Be good while hubby is away and we wont tell hom willwe Gav?

Gav have just dropped in for lunch,will reply to yoyr email latter this arvo.

Cheers

Brian

DuffyFamily <DuffyFamily@...> wrote:

I am sorry if I offended anyone while going through some e-mails. My husband is away for a few days and I thought I was doing him a favour by sorting out the long list of e-mails. I didn't realise that this was a group contact system. Hope I haven't gotten him in touble with you guys.SORRY.

----- Original Message -----

From: Gavin Stephens

To: LPFM_Radio@...

Sent: Friday, September 01, 2006 8:44 AM

Subject: Re: [LPFM] Streaming, Broadband, and a better option.

I'm not completely up on the play with where Telstraclear are with peering these days. I thought they pulled the plug on free peering in Auckland and started considering national traffic as transit? Are Xnet signed up with a private peering arragement with Telstraclear or are Telstra bent at the knees after they big hoo har over de-peering last year or whenever it was?

 

I'm glad someone is finally looking at NZ streaming solutions though. Just more glad it's someone that's fimiliar with the ISP bizz and hooked up with lpfm.

 

Gavin.

 

----- Original Message -----

From: Matt Camp

To: LPFM_Radio@...

Sent: Thursday, August 31, 2006 9:46 PM

Subject: Re: [LPFM] Streaming, Broadband, and a better option.

You're right... although one could argue that a suitable level AAC+
stream is pretty close to cd quality :)

The ISP in question is in fact Xnet, however that's just who I am
suggesting that my clients use for their own broadband connections. I've
worked with their technical staff to ensure that traffic to and from my
service isn't counted towards the data-caps.

My own servers are telehoused around the place, with the primary one
being hosted in the skytower right next to the peering exchange.

Telecom, TelstraClear, Xnet and Quicksilver will all count data to/from
it as national.

I used to work for Ihug, Quicksilver, and TelstraClear doing IP network
design, so I'm reasonably familiar with such things :)

Both Xnet and Quicksilver will do 'unlimited' national traffic (which
really means "within reason"), however for various mostly-technical
reasons, I would suggest using Xnet.

Gavin Stephens wrote:
> The brainwashing conumer sales tricks have made us say without
> thinking twice: "cd quality".
> 
> Unlike the "near cd quality" official title perceptual audio coding
> has been able to attain for its self.
> 
> I can't help but mention that. I read far to often people mis-quoting
> audio coding technologies for cd quality, when most tests use the more
> appropriate term "near" cd quality.
> 
> You've got me interested in the details though. Who's the well known
> ISP? Either way, we'll soon know. But I'd like to know who it will be
> since it will impact on peering and national charges with some ISP's
> depending on who it is.
> 
> Cheers,
> Gavin.
> 
>
>     ----- Original Message -----
>     *From:* Matt Camp <mailto:matt@...>
>     *To:* LPFM_Radio@...
>     <mailto:LPFM_Radio@...>
>     *Sent:* Thursday, August 31, 2006 4:16 PM
>     *Subject:* [LPFM] Streaming, Broadband, and a better option.
>
>
>     This recent discussion on streaming is quite interesting, even if only
>     from the view that it highlights that such things are not that
>     simple, and
>     all sorts of issues can very quickly make it a major headache.
>
>     It doesn't need to be that way.
>
>     I own and operate a small company (totally seperate from any of my
>     other
>     ventures, such as my radio station) which specialised in digital media
>     streaming. Streaming is something I've been interested in for a
>     number of
>     years now, and have significant knowledge of.
>
>     I'm still in the process of launching the company (we haven't even
>     launched our website yet), but at this stage I'm willing to let
>     this list
>     know about one of our products - Streaming services designed for LPFM
>     stations.
>
>     Basically I will provide LPFM operators with the ability to send their
>     audio streams to my hosted streaming servers. This means that you only
>     need to send one copy of the audio, and I will take care of
>     sending it to
>     potentially hundreds of simultaneous listeners.
>
>     I have also come to an arrangement with a well known ISP that will
>     allow
>     you to stream a CD quality audio feed 24 hours a day, 365 days a
>     year, and
>     pay no more than $29.95/month for that broadband connection... no
>     data cap
>     to worry about!
>
>     Pricing on my streaming services has not yet been finally
>     determined, but
>     rest assured it's designed to fit the LPFM type budgets... ie,
>     bugger all.
>
>     And here's the deal for people on this list.... I need people to
>     test my
>     service.
>
>     In order to get a good test, I'm looking for people who want a
>     months free
>     streaming services.... basically you sign up, and I don't charge
>     you for
>     the first month. No obligation to continue beyond that free month
>     either
>     if it turns out you don't like the service.
>
>     I will provide all necessary software, guides, and assistance to
>     get the
>     stream up and running. You will also receive your streams listed
>     on our NZ
>     Audio stream directory, and we can help you with things such as
>     embedded
>     players on your website.  You will receive full stats and logs
>     detailling
>     how many people are listening, where from, what times, etc, etc.
>
>     Various format and bitrate options are available and we can basically
>     cater for anything.
>
>     So yeah.... anyone want some free streaming?
>
>     Drop me an email off-list if you are interested and I'll discuss the
>     details with you.
>
>
>     ------------------------------------------------------------------------
>     No virus found in this incoming message.
>     Checked by AVG Free Edition.
>     Version: 7.1.405 / Virus Database: 268.11.7/434 - Release Date:
>     30/08/2006
>
> 

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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.11.7/434 - Release Date: 30/08/2006

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