Digital amplification

[funberry]

Does anyone know if there is an IC that accepts a digital audio input (binary 12-24 bit, serial or parallel), and converts it to a PWM output?

[bocka]

I remember reading something like this at Zetex. Unfortunately I don't know the part number

[AudioWizard]

I think this thread would be more at home in the "Class D" forum...

Anyway, Texas Instruments makes integrated class D solutions (TPA series), not sure if you could use those to drive a power MOSFET stage or something...

You could also use an Analog Devices Blackfin DSP to implement that. Go see here . Or any decent DSP actually...

There may be other solutions, see what Philips, Maxim and Cirrus make these days...

[funberry]

I think this thread would be more at home in the "Class D" forum...

I know, but that forum specifically mentions "power amplification/ power D/A".

That kind of seems to exclude low level signal processing...

Thanks.

[AudioWizard]

Ok, I think you'll like the TI TAS5xxx series:

for instance here

[Steven]

A noise shaper for bitstream audio comes to mind, although the output frequency may be too high for what you want... The TDA1307 from Philips is an example, but I think it has been discontinued. The TDA1307 was developed for the single bit DAC TDA1547.

Steven

[funberry]

Some background on why I'm asking:

I would like to implement a Subharmonic Synthesizer (SHS) in DSP.

In case you're wondering, an SHS creates a wave identical to a musical program wave, exactly one octave below the original.

Many people buy/make a subwoofer, to discover most program content doesn't contain much material below 50Hz. It's hard to enjoy a sub when it has nothing to reproduce.

To implement an SHS, you low-pass filter frequencies, say, below 80Hz (with a sharp knee). You then divide the frequency of this content by 2 (some implementations use a flip-flop). You sample the envelope of the original wave, and shape the divided wave to the same envelope.
You've got a copy of the original sounds, one octave below.
Then you either mix back the sub-harmonic into the main program, or you output it separately to a subwoofer.

The subharmonic blends in very naturally with the original sound, because waves exactly one octave apart do not sound dissonant.
The ear doen not perceive it as an artifact.

A binary implementation of this would be to take each sample from input in a processor, and present it to the output at half the clock frequency. When the input has completed two full cycles of a wave, the output has completed just one. The processor then discards the extra wave from the input and proceeds to cycle 3. The discard is necessary so the copy's envelope does not last longer than the original's. The processor needs to implement a buffer (queue) for storing the binary values, since the input is running faster than the output.

For this purpose, it's useful to be able to work in binary, and convert to PWM only later.

Adrian

[AudioWizard]

As I suggested first, if you're going to use a DSP anyway, you can generate the PWM directly from the DSP (provided that you have enough power)!

The PDF file from AD I posted, literature on Class D in the Class D forum and on the web should get you started...

Other than that, the chip from TI I talked about seems to be doing just what you asked for. The THD figures are not overly impressive, but I doubt you can do better than that directly on the DSP or via another integrated solution...

[sparky87]

You can also look at Wolfsonmicro.

[soongsc]

funberry,

Take a look at this site and see if the chips are what you are looking for. They take in serial and output pwm.

http://www.advanic.com.tw/Full.htm