What I dont get is this: For a couple bucks I get a standard 13-bit codec.
13 bits is *sort of* standard resolution for telephony when using a linear
codec.
So to equal this with PWM i need to run at 2^13 * 8000 = 65.5MHz. Then I
hear that as a rule of thumb you want to be 10x faster! This would mean i
require a clock of 1/2 a gigahertz! Is this 10x really necessary?
And say you do bump the frequency up 10x, do you need to upsample (digitally
filter) the audio signal you are putting out? Or just keep putting out the
same value for 10 samples...which seems pointless?
I am worried about how hardcore the filtering needs to be after the PWM
output. I dont have weeks to design the greatest filter ever....Are audible
artifacts of all this switching easily removed?
It appears you may have the wrong idea about how the PWM module would
typically be used. You don't need a 500MHz microcontroller or any kind of
fancy digital or analog filters.
Depending upon what your audio format is in to begin with, all you likely
actually need is a single RC low pass analog filter. You hook the resistor
up to the output of the PWM module. On the other side of the resistor you
hook up a capacitor to ground. You get your analog output signal off the
top of that capacitor.
The PWM module is typically quite easy to use, you simply configure a couple
of registers and it starts oscillating at some fixed frequency at some set
duty cycle. You get to set the frequency and duty cycle by playing with the
registers, but normally you just set it and it goes by itself. Normally,
for DAC use, what you would do is simply configure it to output the maximum
practical frequency available (which ideally should be at least 10X the
*sample rate* (not sample bits) of the audio you are trying to output, the
higher the frequency the better). So for instance, if you wanted to output
8-kilosamples of audio per second, you would ideally configure your PWM
module to start oscillating at least at 80kHz. This should be no sweat,
just set up the proper registers and it will be good to go.
Once you have the PWM module configured, you are ready to start outputting
the audio samples. Suppose you have a 10-bit PWM module available (meaning
that you can adjust the duty cycle of the output frequency anywhere from
100% to 0% in 1024 discrete steps). This would allow you to output at
maximum 10-bit audio samples. For each audio sample, you adjust the PWM
duty cycle (but leave the frequency the same) corresponding to what the
audio sample actually is. When you low pass filter the digital square wave
coming out of the PWM output, you get an analog DC voltage. For example if
you had say a 5V CMOS microcontroller, presumably the output of the PWM
module would go from 0V to 5V. Suppose you output a "512" audio sample,
which would correspond to 50% duty cycle on the PWM output. This would
cause the low pass filter to produce an average DC voltage of 2.5V (half on
at 5V, half off at 0V, average 2.5V when low pass filtered). If you
outputted a "256" audio sample, or a 25% duty cycle output, the DC voltage
on the low pass filter capacitor would drop to 1.25V. Etc. This is how you
get your analog output voltage. Typically you would then AC couple this
analog DC voltage through a capacitor to an op-amp buffer (although you can
simply AC couple it to earphones and listen to it albeit with very weak
volume).
Since the output of the PWM module is a digital square wave instead of a
true analog voltage, you want to very heavily low pass filter it, lest the
square lumps of the PWM main frequency will appear in your analog output.
This means you want to place the low pass filter breakpoint at very low
frequency, as low as possible without unreasonably attenuating the higher
frequencies of your real audio signal. If you were outputting 8-kilosamples
of audio per second, then it would seem reasonable to place the low pass
filter breakpoint somewhere very vaguely around 8 kilohertz. Of course, if
your main PWM frequency is a mere 80kHz, that is only one decade higher, so
you will only get 20dBs of attenuation of the 80kHz content... That is not
that great. The higher your PWM frequency, the better the attenuation will
be, so set it for the maximum practical that your microcontroller will
produce (subject to the constraint that the duty cycle resolution will
probably decrease at high PWM frequencies, so there may be a compromise here
as to what maximum frequency is optimal).
The speed of your microcontroller really only needs to be as fast as you can
handle the audio samples to make them appear in the PWM duty cycle control
register(s). Only a modest microcontroller would be adequate for modest
quality audio, although that depends on how clever and efficient a
programmer you are.
