Lpc2000

Hello,

For the analog and audio smarts out there I have some questions.

I need to generate a good quality audio using LPC chips. The audio 
will be a 44.1Khz 16-bit WAV file. I can take care of 
loading/parsing data in RAM. Now, we need to convert the digital 
data to analog signal.

- Would you use SPI or parallel bus to access the DAC chip?
- What DAC chip you guys recomend?
- What is the diference between codec and DAC?!?


Thanks,

Gus

I suppose I'd first ask how good does the audio actually have to 
be?  Full CD quality 44khz 16bit is extreme overkill if it's just 
going to drive a small speaker.  There is also the issue of where is 
all this audio going to be stored?

A simple solution would be a LPC213x micro & use the 10 bit d/a 
that's built it.  If 60db s/n is all you need, that would work 
pretty well.  Some of the TI AIC chips are pretty good for this sort 
of thing as they contain a few etra bits and pieces to simplify the 
hardware.


The fancier SPI port on the LPC213x looks like it would work well 
with a serial DAC or codec.  You just need to pick one that likes 
TDM style signalling (i.e. with a frame sync) rather than  I2S.  DAC 
is output only, CODEC is a DAC + ADC.


- jeff



--- In lpc2000@yahoogroups.com, "Gus" <gus_is_working@y...> wrote:

> 
> Hello,
> 
> For the analog and audio smarts out there I have some questions.
> 
> I need to generate a good quality audio using LPC chips. The audio 
> will be a 44.1Khz 16-bit WAV file. I can take care of 
> loading/parsing data in RAM. Now, we need to convert the digital 
> data to analog signal.
> 
> - Would you use SPI or parallel bus to access the DAC chip?
> - What DAC chip you guys recomend?
> - What is the diference between codec and DAC?!?
> 
> 
> Thanks,
> 
> Gus

--- In lpc2000@yahoogroups.com, "Gus" <gus_is_working@y...> wrote:
> 
> Hello,
> 
> For the analog and audio smarts out there I have some questions.
> 
> I need to generate a good quality audio using LPC chips. The audio 
> will be a 44.1Khz 16-bit WAV file. I can take care of 
> loading/parsing data in RAM. Now, we need to convert the digital 
> data to analog signal.
> 
> - Would you use SPI or parallel bus to access the DAC chip?
> - What DAC chip you guys recomend?
> - What is the diference between codec and DAC?!?

Codec is short for coder/decoder which means ADC and DAC.  These days
you can get a sigma-delta stereo Codec for under $5.  So unless you
are really trying to cut every penny from your design, a CD quality
codec will be easy to work with.  Sigma delta is a way of using a very
high sample rate combined with a very simple circuit to produce an ADC
or DAC which samples the input at a very high rate and only needs a
very simple anti-alias filter.   

I don't know that they are typically SPI compatible since SPI is a
specific protocol.  Most of the codecs I have worked with are
compatible with the Sony/Philips serial interface, IIRC it is called
SPIF or SPDF or SPDI.   A google search should pull up some info.  The
bottom line is this is a simple interface which allows for up to 24
bits per sample and up to 48 kHz sample rate.  

Check out chips from Crystal, AKM and Philips.  I found a couple from
AKM which are very simple to use, low cost and very small.

> A simple solution would be a LPC213x micro & use the 10 bit d/a 
that's built it.

What audio frequency can you do with the internal DAC?
Can someone explain more what is xxdb s/n? I am assuming s/n means 
signal/noise! So when we say 60db s/n, what does this mean? good 
audio? bad? acceptable?

Thanks,

Gus

--- In lpc2000@yahoogroups.com, "jdw07675" 
<jdwilson@c...> wrote:
> 
> I suppose I'd first ask how good does the audio actually have to 
> be?  Full CD quality 44khz 16bit is extreme overkill if it's just 
> going to drive a small speaker.  There is also the issue of where 
is 
> all this audio going to be stored?
> 
> A simple solution would be a LPC213x micro & use the 10 bit d/a 
> that's built it.  If 60db s/n is all you need, that would work 
> pretty well.  Some of the TI AIC chips are pretty good for this 
sort 
> of thing as they contain a few etra bits and pieces to simplify 
the 
> hardware.
> 
> 
> The fancier SPI port on the LPC213x looks like it would work well 
> with a serial DAC or codec.  You just need to pick one that likes 
> TDM style signalling (i.e. with a frame sync) rather than  I2S.  
DAC 
> is output only, CODEC is a DAC + ADC.
> 
> 
> - jeff
> 
> 
> 
> --- In lpc2000@yahoogroups.com, "Gus" <gus_is_working@y...> wrote:
> > 
> > Hello,
> > 
> > For the analog and audio smarts out there I have some questions.
> > 
> > I need to generate a good quality audio using LPC chips. The 
audio 

> > will be a 44.1Khz 16-bit WAV file. I can take care of 
> > loading/parsing data in RAM. Now, we need to convert the digital 
> > data to analog signal.
> > 
> > - Would you use SPI or parallel bus to access the DAC chip?
> > - What DAC chip you guys recomend?
> > - What is the diference between codec and DAC?!?
> > 
> > 
> > Thanks,
> > 
> > Gus

--- In lpc2000@yahoogroups.com, "Gus" <gus_is_working@y...> wrote:
> 
> > A simple solution would be a LPC213x micro & use the 10 bit d/a 
> that's built it.
> 
> What audio frequency can you do with the internal DAC?
> Can someone explain more what is xxdb s/n? I am assuming s/n means 
> signal/noise! So when we say 60db s/n, what does this mean? good 
> audio? bad? acceptable?

Good vs. bad depends on your application.  60 dB is just a way of
describing how much noise the DAC will add to the signal.  20 dB is a
factor of 10x in voltage, 60 dB is a factor of 1000.  So a 1 volt
signal will have 1 mV of noise.  

You need to tell us what quality level you require.  Are you shooting
for CD quality (meaning nearly no audible noise or distortion) or more
like telephone quality (some noise and audible distortion)?

60db is okay. now, how fast can I update the analog output on LPC 
chip?

I looked on philips website and there is only data sheet with 
minimum information!! where is the manual? I can't find it.

Gus

--- In lpc2000@yahoogroups.com, "Rick Collins" <gnuarm@a...> wrote:
> 
> --- In lpc2000@yahoogroups.com, "Gus" <gus_is_working@y...> wrote:
> > 
> > > A simple solution would be a LPC213x micro & use the 10 bit 
d/a 
> > that's built it.
> > 
> > What audio frequency can you do with the internal DAC?
> > Can someone explain more what is xxdb s/n? I am assuming s/n 
means 
> > signal/noise! So when we say 60db s/n, what does this mean? good 
> > audio? bad? acceptable?
> 
> Good vs. bad depends on your application.  60 dB is just a way of
> describing how much noise the DAC will add to the signal.  20 dB 
is a
> factor of 10x in voltage, 60 dB is a factor of 1000.  So a 1 volt
> signal will have 1 mV of noise.  
> 
> You need to tell us what quality level you require.  Are you 
shooting
> for CD quality (meaning nearly no audible noise or distortion) or 
more
> like telephone quality (some noise and audible distortion)?

--- In lpc2000@yahoogroups.com, "Gus" <gus_is_working@y...> wrote:
> 
> 60db is okay. now, how fast can I update the analog output on LPC 
> chip?
> 
> I looked on philips website and there is only data sheet with 
> minimum information!! where is the manual? I can't find it.

Don't assume 60 dB.  That number will have to come from the data sheet
as well.  A theoretical maximum for a DAC is 6 dB per bit, but that
level is never achieved as there are always multiple sources of noise
in a DAC.  But it should not be a lot worse, maybe 55 dB or so.  

If this info is not in the data sheet, contact Philips directly.

I've got a preliminary data sheet from Philips for the LPC213X that 
states the ADC is capable conversions at 400 Ksamples/sec, 10 bit
result. The data sheet also says that the input measurement range is 0
- 3.3V

Regards
Dennis

--- In lpc2000@yahoogroups.com, "Rick Collins" <gnuarm@a...> wrote:
> 

> --- In lpc2000@yahoogroups.com, "Gus" <gus_is_working@y...> wrote:
> > 
> > 60db is okay. now, how fast can I update the analog output on LPC 
> > chip?
> > 
> > I looked on philips website and there is only data sheet with 
> > minimum information!! where is the manual? I can't find it.
> 
> Don't assume 60 dB.  That number will have to come from the data
sheet
> as well.  A theoretical maximum for a DAC is 6 dB per bit, but that
> level is never achieved as there are always multiple sources of
noise

> in a DAC.  But it should not be a lot worse, maybe 55 dB or so.  
> 
> If this info is not in the data sheet, contact Philips directly.

I  have dropped into this discussion 1/2 through but felt like contributing

Don't forget that the 6dB/bit applies to the A/D or D/A REGARDLESS of 
sample rate. This factor can be used to improve accuarcy in both 
conversions by increasing the sample rate.

If you sample a 10 bit A/D at 400Kz, then down sample to 44Khz (probably 
using a linear phase  finite impulse reponse filter{ FIR }) then you get 
a noise reduction because you filter off the higher sample noise 
frequencies. You will achieve an extra 3 bits - i.e equivalent to a 13 
Bit A/D.

The same applies to D/A - take the digital samples at 44Kz increase 
frequency to 400Kz - apply an FIR in the digital domain then send to 
your D/Q. The advantage of this is that the analoge filter on the output 
is very simple as it operates at 400Kz.

Some of the early CD players worked on this principal. They had a high 
freq 14 bit D/A. The advantage of this is the improved phase response of 
the system. Jo public couldn't understand how a 14 bit D/A was giving 16 
bit performance so I think they stopped doing it.

I can give FIR details if anyone is interested.

DSP rules ! Yay !  I havent done this stuff since the 80's

Owen Mooney

Owen Mooney wrote:

> Some of the early CD players worked on this principal. They had a high 
> freq 14 bit D/A. The advantage of this is the improved phase response of 
> the system. Jo public couldn't understand how a 14 bit D/A was giving 16 
> bit performance so I think they stopped doing it.

No, I believe the idea transformed to the extreme with MASH 1 bit 
converters. Sample fast enough and you only need 1 bit (a single high 
speed comparator). Isn't this what sigma-delta codecs use ?

-- 
Cheers,
Bruce
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For audio, a signal to noise plus distortion spec is common.  That's 
what 6 dB per bit (with an ideal converter) is all about.  Human hearing 
is complex, and most ears aren't very sensitive to many types of 
distortion.  Harmonic distortion, for example, really isn't noticable 
(to most people) at 0.1%, which is only -40 dB.  But noise at -40 dB is 
quite noticable and very annoying, especially if it is tonal (eg, 1 kHz 
sine wave).

All DACs output a signal that is a ratio of some reference.  Most 
likely, the single largest factor that will impact the perceived sound 
quality from the internal DAC is the quality of its reference.  If the 
reference voltage varies, and those changes have any spectral energy in 
the audio range, the output will probably seem to have "wierd sounds".

Even though the DAC's best theoretical perfornance is 60 dB S/N+D, as 
long as it's monotonic and the reference is good, most of that will 
likely be less noticable distortion , rather than annoying noise.  A 
quick check is to simply output a DC signal (just write the same value 
to the DAC over and over), and measure the noise... or run that "signal" 
into an amplifier and listen to it in a quiet room or via headphones.  
The "hiss" associated analog cassette tapes is about -50 to 60 dB, for 
the sake of comparison.


There are techniques to increase resolution using oversampling.  
Oversampling generally assumes there is noise, and by collecting more 
samples than necessary, you can low pass filter (or average together) 
many samples to get one more accurate from many lower accuracy ones.

In the context of using the 10 bit DAC, you could (in theory) output 
twice times as many samples per second as necessary.  You'd attempt to 
output 11 bits by driving the DAC first with all but the lower bit, and 
then increment it during the second sample if the least significant bit 
is a 1.  Of course, you'll need an analog low pass filter.  Don't forget 
to use low-noise analog parts to avoid adding noise, and capacitors with 
highly linear voltage/charge (capacitance) to avoid adding distortion 
(eg, NPO ceramic or certain poly films).

In practice, this probably won't necessarily work well, because the 
linearity of the DAC output steps isn't perfect.  This is often called 
differential nonlinearity


Sigma-delta converters use an analog trick to "shape" their 
noise+distortion, and they run very fast so there's lots of unnecessary 
spectrum for all that noise and distortion to get filtered away.   It's 
easist and most practical with just 1 bit, because a 1 bit converter has 
only two possible output voltages.... which always fits a perfect 
straight (linear) line.  So it's easy to scale up to very high speed 
(oversampling ratio) without the problem where consecutive conversions 
don't add together as expected.  Together with the noise shaping, the 
massive noise+distortion of using just 1 bit is easy to filter away.  
The important thing about sigma-delta is the noise shaping, which (on 
average) the errors to be concentrated in the higher frequencies... so 
the filtering improves the resolution much more than simply filtering 
oversampled evenly distributed white noise.


If you really need near CD audio quality, your best approach is to use a 
quality DAC chip.  Saddly, these all take high speed serial bitstreams, 
and Philips doesn't seem to have any LPC chips that are designed to 
output the common formats (eg, I2S).  These DACs use phase locked loops 
or other techniques that require a steady clocked bitstream.  It would 
be quite difficult to emulate such a thing from the SPI port, for example.

There is another vendor with ARM7 chips now sampling that does include a 
peripheral that outputs these types of bitstreams, and they also provide 
a DMA-like engine so you can set up a block transfer (avoid burning lots 
of CPU with 44100 interrutps/sec) and a second pending block transfer 
(avoid needing worst case interrupt latency of 22.6 us).  Sorry Philips...

It's possible to build a similar thing in logic using an FPGA... accept 
data into a fifo via the external bus and stream it out to the DAC as 
the clock requires, and interrupt the processor at a low watermark.  But 
that's a lot harder and more expensive than it's probably worth.


Paul

>
>
> > Some of the early CD players worked on this principal. They had a high
> > freq 14 bit D/A. The advantage of this is the improved phase 
> response of
> > the system. Jo public couldn't understand how a 14 bit D/A was 
> giving 16
> > bit performance so I think they stopped doing it.
>
> No, I believe the idea transformed to the extreme with MASH 1 bit
> converters. Sample fast enough and you only need 1 bit (a single high
> speed comparator). Isn't this what sigma-delta codecs use ?
>
> -- 
> Cheers,
> Bruce

Yes, that is one way of describing the core of a sigma-delta codec.

I'd agree with Owen on his observation about using DSP algorithms to 
reduce hardware requirements. We take this to the extreme, 
inplementing a full-blown software modem on the LPC2000. We've had 
customers who have replaced older 8-bit systems with 32-bit RISCs 
like the LPC2000 and actually reducing the system BOM cost, as they 
no longer need a modem IC. My point is that as well as having a much 
better programming environment (full ANSI 'C' etc.), more memory etc. 
you have a lot more processing power than any 8-bit, so why not use 
it?

To get back on-topic: does anyone know of a good (i.e. cheap!) way to 
interface an audio codec to any of the LPC2000 range? We've managed 
to do it without additional hardware for some voice-band (i.e. 8KHz 
sampling) codecs with a few tricks, but not for something with higher 
sampling rates (such as 44KHz and beyond). They tend to have 
something like I2S on their digital side, which doesn;t sit to well 
with the available interfaces.

Regards
Brendan

******************************************************************
Brendan Murphy, Chief Technical Officer (CTO)
Innovada, invent, DCU, Dublin 9, Ireland
Phone (DDI): +353 1 700 7522 Fax: +353  1 700 7545 Mobile: +353 87 
648 1741
E-Mail: brendan.murphy@...
Web: http://www.innovada.com
*******************************************************************



--- In lpc2000@yahoogroups.com, Bruce Paterson <bruce@t...> wrote:
> Owen Mooney wrote:
> 
> > Some of the early CD players worked on this principal. They had a 
high 
> > freq 14 bit D/A. The advantage of this is the improved phase 
response of 
> > the system. Jo public couldn't understand how a 14 bit D/A was 
giving 16 
> > bit performance so I think they stopped doing it.
> 
> No, I believe the idea transformed to the extreme with MASH 1 bit 
> converters. Sample fast enough and you only need 1 bit (a single 
high 

> speed comparator). Isn't this what sigma-delta codecs use ?
> 
> -- 
> Cheers,
> Bruce
> -------------------------------------------------------------------
> This email and any files transmitted with it are confidential and
> intended solely for the use of the individual or entity to whom
> they are addressed. If you have received this email in error please
> notify the system manager.
> 
>      /\\\/\\\/\\\    /   /      Bruce Paterson
>     /  \\\ \\\ \\\  /   /    Senior Design Engineer
>    /   /\\\/\\\/\\\/   /   8 Anzed Court, Mulgrave, Vic, 3170
>   /   /  \\\ \\\ \\\  /  PO Box 4112, Mulgrave, Vic, 3170, Australia
> /   /    \\\/\\\ \\\/   Ph: +61 3 8561 4232   Fax: +61 3 9560 9055
>        Tele-IP Ltd.      Email: bruce@t...    Icq: #32015991
>                          WWW:   http://www.tele-ip.com       VK3TJN
> -------------------------------------------------------------------

--- In lpc2000@yahoogroups.com, Paul Stoffregen <Paul@P...> wrote:
> There is another vendor with ARM7 chips now sampling that does 
include a 
> peripheral that outputs these types of bitstreams, and they also 
provide 
> a DMA-like engine so you can set up a block transfer (avoid burning 
lots 
> of CPU with 44100 interrutps/sec) and a second pending block 
transfer 
> (avoid needing worst case interrupt latency of 22.6 us).  Sorry 
Philips...
> 

Paul,

I don't suppose you'd like to share with us the name of this vendor?

Brendan

Most of them that I'm aware of have an AC97 interface. I haven't 
explored this on the LPC2xxx yet, but plan to. I have a funny feeling 
somewhwre that the LPc has this, but may eb wrong, it's a vague 
recollection of a response in this group a couple of weeks ago.

Al

brendanmurphy37 wrote:

> 
> 
> Yes, that is one way of describing the core of a sigma-delta codec.
> 
> I'd agree with Owen on his observation about using DSP algorithms to
> reduce hardware requirements. We take this to the extreme,
> inplementing a full-blown software modem on the LPC2000. We've had
> customers who have replaced older 8-bit systems with 32-bit RISCs
> like the LPC2000 and actually reducing the system BOM cost, as they
> no longer need a modem IC. My point is that as well as having a much
> better programming environment (full ANSI 'C' etc.), more memory etc.
> you have a lot more processing power than any 8-bit, so why not use
> it?
> 
> To get back on-topic: does anyone know of a good (i.e. cheap!) way to
> interface an audio codec to any of the LPC2000 range? We've managed
> to do it without additional hardware for some voice-band (i.e. 8KHz
> sampling) codecs with a few tricks, but not for something with higher
> sampling rates (such as 44KHz and beyond). They tend to have
> something like I2S on their digital side, which doesn;t sit to well
> with the available interfaces.
> 
> Regards
> Brendan
> 
> ******************************************************************
> Brendan Murphy, Chief Technical Officer (CTO)
> Innovada, invent, DCU, Dublin 9, Ireland
> Phone (DDI): +353 1 700 7522 Fax: +353  1 700 7545 Mobile: +353 87
> 648 1741
> E-Mail: brendan.murphy@...
> Web: http://www.innovada.com
> *******************************************************************
> 
> 
> 
> --- In lpc2000@yahoogroups.com, Bruce Paterson <bruce@t...> wrote:
>  > Owen Mooney wrote:
>  >
>  > > Some of the early CD players worked on this principal. They had a
> high
>  > > freq 14 bit D/A. The advantage of this is the improved phase
> response of
>  > > the system. Jo public couldn't understand how a 14 bit D/A was
> giving 16
>  > > bit performance so I think they stopped doing it.
>  >
>  > No, I believe the idea transformed to the extreme with MASH 1 bit
>  > converters. Sample fast enough and you only need 1 bit (a single
> high
>  > speed comparator). Isn't this what sigma-delta codecs use ?
>  >
>  > --
>  > Cheers,
>  > Bruce
>  > -------------------------------------------------------------------
>  > This email and any files transmitted with it are confidential and
>  > intended solely for the use of the individual or entity to whom
>  > they are addressed. If you have received this email in error please
>  > notify the system manager.
>  >
>  >      /\\\/\\\/\\\    /   /      Bruce Paterson
>  >     /  \\\ \\\ \\\  /   /    Senior Design Engineer
>  >    /   /\\\/\\\/\\\/   /   8 Anzed Court, Mulgrave, Vic, 3170
>  >   /   /  \\\ \\\ \\\  /  PO Box 4112, Mulgrave, Vic, 3170, Australia
>  > /   /    \\\/\\\ \\\/   Ph: +61 3 8561 4232   Fax: +61 3 9560 9055
>  >        Tele-IP Ltd.      Email: bruce@t...    Icq: #32015991
>  >                          WWW:   http://www.tele-ip.com       VK3TJN
>  > -------------------------------------------------------------------
> 
> 
> 
> 
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On Thu, 2005-01-20 at 15:40 -0800, Paul Stoffregen wrote:

> If you really need near CD audio quality, your best approach is to use a 
> quality DAC chip.  Saddly, these all take high speed serial bitstreams, 
> and Philips doesn't seem to have any LPC chips that are designed to 
> output the common formats (eg, I2S).  These DACs use phase locked loops 
> or other techniques that require a steady clocked bitstream.  It would 
> be quite difficult to emulate such a thing from the SPI port, for example.

Higher rate audio codecs also seem to be power-hogs, at least in
relation to the miserly power that ARM7 is typically associated with.

Does anyone have any good experience with lower power, speech quality
codecs, or even higher rate (CD quality) ones that don't suck (power)
and are suitable for battery powered-operation? Would you recommend
going only with the onboard ADC for 8 khz audio for best low-power
performance?


-- 
Brett Delmage <BDelmage@...>
JSI Telecom

> I don't suppose you'd like to share with us the name of this 
vendor?

That vendor and other vendors please! What ARM chips are out there 
with I2S bus?

Gus

--- In lpc2000@yahoogroups.com, "brendanmurphy37" 
<brendan.murphy@i...> wrote:
> 
> --- In lpc2000@yahoogroups.com, Paul Stoffregen <Paul@P...> wrote:
> > There is another vendor with ARM7 chips now sampling that does 
> include a 
> > peripheral that outputs these types of bitstreams, and they also 
> provide 
> > a DMA-like engine so you can set up a block transfer (avoid 
burning 
> lots 
> > of CPU with 44100 interrutps/sec) and a second pending block 
> transfer 
> > (avoid needing worst case interrupt latency of 22.6 us).  Sorry 
> Philips...
> > 
> 
> Paul,
> 
> I don't suppose you'd like to share with us the name of this 
vendor?
> 
> Brendan

Ok, since 2 people asked...

> That vendor and other vendors please! What ARM chips are out there
> with I2S bus?

It's Atmel.  These "SAM" parts have come up in this group over and over 
again.  Except for a few folks with early ES silicon, there's been a lot 
more hype and vapor than samples!

The 64k flash part is sampling now, but not in production yet.  
Production _might_ be in April, maybe.  Whatever they promise, remember 
they've been very late many times before (anyone remember waiting 2 
years for the AVR?)

http://www.atmel.com/dyn/products/product_card.asp?part_id=3521

If you click on the eval kit link, then click on "Avent" at the bottom, 
maybe it'll take you to a page with more info.  At least it did when I 
tried it a couple days ago, but this morning Avent's website only gives 
an error message.  Probably best to just call your local Avent office.  
They had about 30 of the eval boards in stock earlier this week.  Don't 
forget the part number: AT91SAM7S64-IAR.  It's $258 for pretty much just 
a board with the chip and IAR's USB-JTAG cable.

I can confirm this thing is finally for real.  I called up Avent about a 
week ago, and one of the eval boards is sitting right here.  Ordered 
with my personal credit card.  Didn't need to be a special/important 
customer to get it.  Just showed up yesterday.  Haven't hooked it up yet....

Also, I believe the Cirrus ARM parts have included I2S interfaces for 
years, but I'm not really familiar with those parts.  Years ago, they 
required external DRAM.

About the Atmel "SAM" parts, there's a lot of pretty impressive stuff.  
Lots of nice peripherals.  But one thing to pay attention to is the 
flash speed.  This opinion is only from reading the datasheet and not 
real experience with the part (yet), but it appears you can only really 
execute thumb from the flash at the full clock speed.  Also, if you use 
the USB port, you have to clock at 48 MHz (using the PLL, so at least a 
normal crystal is still ok).


The eval kit comes with some examples, but not really any app notes or 
other easy-to-understand docs yet.  If you're good at figuring out how 
to use peripherals from only datasheets, you're probably fine.  But if 
not, you may find much of the datasheet terse and difficult.  For 
example, try reading the USB section!  The I2S output is part of a 
pretty generic high-speed serial peripheral that's very configurable.  
So far, there's only minimal explaination and diagrams that show what 
each option really does.  Atmel does have an app note about I2S using 
what looks like an earlier version of this peripheral in one of their 
other chips, but this new one appears to have expanded some of the 
options.  My point is, at this early stage, be mentally prepared to need 
to do some fiddling and experimenting to figure out exactly how to 
really make this complex chip work the way you want.

But it finally is sampling and eval boards are available off the shelf 
to anyone (well, probably in the US only).  It does have I2S, and 
there's DMA-like buffering that makes these higher speed transfers 
reasonable without burning lots of CPU or needing really low interrupt 
latency.


Paul

--- In lpc2000@yahoogroups.com, Brett Delmage <BDelmage@J...> wrote:
> Higher rate audio codecs also seem to be power-hogs, at least in
> relation to the miserly power that ARM7 is typically associated with.
> 
> Does anyone have any good experience with lower power, speech quality
> codecs, or even higher rate (CD quality) ones that don't suck (power)
> and are suitable for battery powered-operation? Would you recommend
> going only with the onboard ADC for 8 khz audio for best low-power
> performance?

Take a look at the AK4550 from AKM.  At 10 mA from 3.3 volts, this is
low power as well as small size.  The last time I got a price quote,
it was around $4 a chip.  

There should be tons of low power codecs or at least DACs since all
the battery powered CD and MP3 players use them.  I think you need to
look a bit harder.  You are likely finding only the high end devices
that are targeted to 5.1 channel systems and such.  Look for simple
stereo chips.

Indeed, low power codecs aren't hard to find.  Here's an example...

Go to www.ti.com (Texas Instruments), and type "low power codec" into 
the keyword search.

The first result is PCM3501... which is a voice quality codec (16 bits 
but lower sample rates), uses 9 mA at 3.3 volts.  In stock at Digikey.

The second result is PCM3500... very similar to PCM3501.

The third result is TLV320AIC26... which is a CD quality DAC (24 bits, 
48 kHz, 97 dB S/N) and voice quality A/D, plus an amplifier and other 
goodies.  Looks like the DAC needs about 4.6 mA, plus maybe another 2.3 
mA if you use the PLL.  Lots of options with this chip, read the 
datasheet.  That part number is an eval board, but the actual chip is in 
stock at Digikey too.

Lots more similar chips, but some others appear on that first page of 25.

If you only need a DAC, well, try some more searching and you'll find 
'em.  Cirrus Logic, Analog Devices, National Semiconductor, Maxim and 
Linear Tech might be other companies to try besides Texas Instruments.


Paul