Why there are so many bits in sigma-delta audio codec?

Another reason to keep those bits is that there is often a digital volume control in the Codec, so you would like to be able to turn down the amplitude by 20 db or so without losing any of your dynamic range. Analog volume controls are becoming a thing of the past.


Bob

Very good audio converters have 20 to 21 bits of performance. I seem to 
remember that 24 bit performance in a 20kHz bandwidth is equivalent to the 
thermal (johnson) noise of a 3 ohm resistor at room temperature.

Converters will never have 24 bit performance in an audio circuit, so 
clearly there are more than a few marketing bits in most audio converters.

Noise performance of actual converters is also a function of the buffer 
amplifier circuits, anti-imaging or anti-aliasing filters, power supply and 
pcb layout in addition to the converter itself. Sometimes the resistors in 
the supporting circuits contribute more noise than the converters. 

There have been 20 bit converters. The AC97 codec used for a long time in 
PCs used a 20 bit data word. Today, with the exception of low power or 
mixed signal devices, most audio converters have more than 16 bits of 
dynamic range. Many are better than 20 bits, especially DACs, so 24 bits is 
reasonable. These bits are usually expressed in twos complement and are 
often formatted into a 32 bit word (before any floating point conversion).

Al Clark
www.danvillesignal.com 








"Vladimir Vassilevsky" <nospam@nowhere.com> wrote in
news:pPCdnYn30IpEhh3NnZ2dnUVZ5hGdnZ2d@giganews.com: 

> 
> "Les Cargill" <lcargill99@comcast.com> wrote in message 
> news:k5osd0$abh$1@dont-email.me...
> 
>> It's nice to have the option of 24 bits, but there are vanishingly
>> few cases where you actually get even 96 full dB (16 bits ) of clean
>> input into an ADC.
> 
> Good quality audio ADC have dynamic range over 120dB. Instrumental ADCs 
> reach over 140dB.
> Despite of those high parameters, ADCs are always a bottleneck and you
> have to switch gears and do all kinds of analog signal conditioning
> before the ADC.
> 
>> SFAIK, fab process and the general ... economics of electronics
>> dictate a standardization to 24 bit ADCs but being in a place
>> to exploit them fully is rare. 2^24 is one big ole
>> number... 16,777,216, aka 144 dB.
> 
> If you want to measure something to a resolution of 1%, then 24 bits is
> mere 100dB of dynamic range at best.
> 
> Vladimir Vassilevsky
> DSP and Mixed Signal Consultant
> www.abvolt.com
> 
> 
> 

Al Clark <aclark@danvillesignal.com> wrote:
> Very good audio converters have 20 to 21 bits of performance. I seem to 
> remember that 24 bit performance in a 20kHz bandwidth is equivalent to the 
> thermal (johnson) noise of a 3 ohm resistor at room temperature.

> Converters will never have 24 bit performance in an audio circuit, so 
> clearly there are more than a few marketing bits in most audio converters.

There is an additional advantage of more bits, in that it allows
more to keep the record level a little lower, reducing the probability
of hitting full scale on unexpected peaks, then adjusting the level
when converting to 16 bits.

> Noise performance of actual converters is also a function of the buffer 
> amplifier circuits, anti-imaging or anti-aliasing filters, power supply and 
> pcb layout in addition to the converter itself. Sometimes the resistors in 
> the supporting circuits contribute more noise than the converters. 

> There have been 20 bit converters. The AC97 codec used for a long time in 
> PCs used a 20 bit data word. Today, with the exception of low power or 
> mixed signal devices, most audio converters have more than 16 bits of 
> dynamic range. Many are better than 20 bits, especially DACs, so 24 bits is 
> reasonable. These bits are usually expressed in twos complement and are 
> often formatted into a 32 bit word (before any floating point conversion).

Why convert to floating point?  32 bit fixed point should have enough
dynamic range for anything you would want to do to it.

-- glen

"Al Clark" <aclark@danvillesignal.com> wrote in message 
news:XnsA0F08FA545CCBaclarkdanvillesignal@69.16.185.247...

> Very good audio converters have 20 to 21 bits of performance.

Digital resolution should have a headroom of at least 2 bits over ENOB of 
converter; so the quantization error would not influence the performance.

> I seem to
> remember that 24 bit performance in a 20kHz bandwidth is equivalent to the
> thermal (johnson) noise of a 3 ohm resistor at room temperature.

Compared to what full scale voltage?
Let's say full scale is 1.5Vrms. Then -144dB makes ~90nVrms; or 
0.6nV/root(Hz) in the bandwidth of 20kHz. This is the noise of ~24 Ohm 
resistor at room temperature.

> Converters will never have 24 bit performance in an audio circuit, so
> clearly there are more than a few marketing bits in most audio converters.

True 24 bit range is attainable with today's technology; it's just the 
matter of cost.
To get 144dB range, you need to connect about 256 x 120dB ADCs in parallel. 
Big expensive PCB; however quite doable.

> Noise performance of actual converters is also a function of the buffer
> amplifier circuits, anti-imaging or anti-aliasing filters, power supply 
> and
> pcb layout in addition to the converter itself.

The clock jitter is important also.

> Sometimes the resistors in
> the supporting circuits contribute more noise than the converters.
>
> There have been 20 bit converters. The AC97 codec used for a long time in
> PCs used a 20 bit data word.

In I2S, 20 bits is one of standard options; supported everywhere.

> Today, with the exception of low power or
> mixed signal devices, most audio converters have more than 16 bits of
> dynamic range.

Consumer class A/Ds and D/As typically top at the magic number of 100dB. 
Parts with performance above 100dB are available, however they are much more 
expensive. Power consumption is also higher by number of times.


Vladimir Vassilevsky
DSP and Mixed Signal Consultant
www.abvolt.com

>
>
> "Vladimir Vassilevsky" <nospam@nowhere.com> wrote in
> news:pPCdnYn30IpEhh3NnZ2dnUVZ5hGdnZ2d@giganews.com:
>
>>
>> "Les Cargill" <lcargill99@comcast.com> wrote in message
>> news:k5osd0$abh$1@dont-email.me...
>>
>>> It's nice to have the option of 24 bits, but there are vanishingly
>>> few cases where you actually get even 96 full dB (16 bits ) of clean
>>> input into an ADC.
>>
>> Good quality audio ADC have dynamic range over 120dB. Instrumental ADCs
>> reach over 140dB.
>> Despite of those high parameters, ADCs are always a bottleneck and you
>> have to switch gears and do all kinds of analog signal conditioning
>> before the ADC.
>>
>>> SFAIK, fab process and the general ... economics of electronics
>>> dictate a standardization to 24 bit ADCs but being in a place
>>> to exploit them fully is rare. 2^24 is one big ole
>>> number... 16,777,216, aka 144 dB.
>>
>> If you want to measure something to a resolution of 1%, then 24 bits is
>> mere 100dB of dynamic range at best.
>>
>> Vladimir Vassilevsky
>> DSP and Mixed Signal Consultant
>> www.abvolt.com
>>
>>
>>
> 

Robert Adams <robert.adams@analog.com> wrote:

> Another reason to keep those bits is that there is often a digital 
> volume control in the Codec, so you would like to be able to turn 
> down the amplitude by 20 db or so without losing any of your 
> dynamic range. 

For me, it is the ability to change the level after recording.

> Analog volume controls are becoming a thing of the past.

I wondered about that. I used to use a MiniDisk recorder with a digital
record level control. I now use the Tascam DR-1, which does do 24 bit,
and has an analog record level. (I believe the playback level control
is digital.)

Are digital level controls, that is amplifiers with digital gain
selection, noisier than analog controls? Seems to me that they
might be.

-- glen

"glen herrmannsfeldt" <gah@ugcs.caltech.edu> wrote:

> Why convert to floating point?  32 bit fixed point should have enough
> dynamic range for anything you would want to do to it.

Floating point makes perfect sense for audio; as relative noise level 
matters rather the the absolute level.
As for 32 bit fixed,  I routinely do practical projects where 32 bit range 
is not enough for raw data representation.
There is analog gear switching; of course; I have to switch gears in digital 
processing, too.

Vladimir Vassilevsky
DSP and Mixed Signal Consultant
www.abvolt.com

Floating point can be very useful in the signal processing chain.

A few examples are FFTs (eliminates scaling between butterflys), 
compressors, agcs, limiters.

If a processor supports floating point operations natively, it can make 
programming much easier without a computation burden. In fact, in may be 
faster depending on the function.

In a SHARC DSP, it takes one instruction to convert from float to fixed or 
fixed to float. SHARCs and most floating point processors are usually very 
good at fixed point as well. They tend to support long fixed point words, 
which makes them better than most fixed point only processors for audio.

I think it is almost always folly to use floating point emulation with a 
fixed point DSP. It can take a 100 instructions depending on the floating 
point format. Floating point DSPs tend to do fixed and floating point  
operations equally well.

I am not saying that fixed point is inferior. The are many cases where 
fixed point is the better choice. I prefer to have both choices available.


Al Clark
www.danvillesignal.com



glen herrmannsfeldt <gah@ugcs.caltech.edu> wrote in
news:k5pmck$80d$1@speranza.aioe.org: 

>> There have been 20 bit converters. The AC97 codec used for a long time
>> in PCs used a 20 bit data word. Today, with the exception of low power
>> or mixed signal devices, most audio converters have more than 16 bits
>> of dynamic range. Many are better than 20 bits, especially DACs, so 24
>> bits is reasonable. These bits are usually expressed in twos complement
>> and are often formatted into a 32 bit word (before any floating point
>> conversion). 
> 
> Why convert to floating point?  32 bit fixed point should have enough
> dynamic range for anything you would want to do to it.
> 
> -- glen
> 

"Vladimir Vassilevsky" <nospam@nowhere.com> wrote in
news:b6CdnbJBfJBlwB3NnZ2dnUVZ5tGdnZ2d@giganews.com: 

> 
> "Al Clark" <aclark@danvillesignal.com> wrote in message 
> news:XnsA0F08FA545CCBaclarkdanvillesignal@69.16.185.247...
> 
>> Very good audio converters have 20 to 21 bits of performance.
> 
> Digital resolution should have a headroom of at least 2 bits over ENOB
> of converter; so the quantization error would not influence the
> performance. 
> 
>> I seem to
>> remember that 24 bit performance in a 20kHz bandwidth is equivalent to
>> the thermal (johnson) noise of a 3 ohm resistor at room temperature.
> 
> Compared to what full scale voltage?
> Let's say full scale is 1.5Vrms. Then -144dB makes ~90nVrms; or 
> 0.6nV/root(Hz) in the bandwidth of 20kHz. This is the noise of ~24 Ohm 
> resistor at room temperature.

I'm not arguing. I stated this from memory. I think someone at Cirrus Logic 
wrote a paper discussing this.

24 ohms is not much.


> 
>> Converters will never have 24 bit performance in an audio circuit, so
>> clearly there are more than a few marketing bits in most audio
>> converters. 
> 
> True 24 bit range is attainable with today's technology; it's just the 
> matter of cost.
> To get 144dB range, you need to connect about 256 x 120dB ADCs in
> parallel. Big expensive PCB; however quite doable.

Assuming that all the various noise sources are orthogonal. Good luck with 
that.


> 
>> Noise performance of actual converters is also a function of the buffer
>> amplifier circuits, anti-imaging or anti-aliasing filters, power supply
>> and
>> pcb layout in addition to the converter itself.


> 
> The clock jitter is important also.

Absolutely!

The right data at the wrong time is the wrong data.



Al Clark
www.danvillesignal.com

On 10/18/2012 8:04 AM, Randy Yates wrote:
> Robert Adams<robert.adams@analog.com>  writes:
>
>> In a sigma delta ADC,
>
> and by the way, it's "delta sigma"!


As long as you want to nitpick, (or is it nitpik?) the last time I 
looked it up (I can't keep the order straight) they said it was 
acceptable either way.  I don't recall who that was, but I don't think 
it was Wikipedia, rather someone a bit more authoritative.

Rick

On 10/18/2012 3:55 PM, glen herrmannsfeldt wrote:
> Robert Adams<robert.adams@analog.com>  wrote:
>
>> Another reason to keep those bits is that there is often a digital
>> volume control in the Codec, so you would like to be able to turn
>> down the amplitude by 20 db or so without losing any of your
>> dynamic range.
>
> For me, it is the ability to change the level after recording.
>
>> Analog volume controls are becoming a thing of the past.
>
> I wondered about that. I used to use a MiniDisk recorder with a digital
> record level control. I now use the Tascam DR-1, which does do 24 bit,
> and has an analog record level. (I believe the playback level control
> is digital.)
>
> Are digital level controls, that is amplifiers with digital gain
> selection, noisier than analog controls? Seems to me that they
> might be.
>
> -- glen

I'm not sure what value a digital control is on record.  To optimize 
noise levels you want the analog signal to be as large as possible 
without clipping.  That should match full scale on the digital side.  So 
why would it be a good thing to reduce the signal level on the digital 
side during record?  The level on the analog side is what you care about.

After that the only adjustment should be on playback which might just as 
well be on the analog side, but I guess it would be easier in digital 
and shouldn't impact anything unless you have an overly large gain in 
the amp.

Is there something going on in the recording industry so that this makes 
no sense?

Rick