Nicholas A. Sinnott-Armstrong wrote:
--- James Richard Tyrer wrote:
No, we are starting with 24bit PCM. If you look at this SVGZ file:
http://home.earthlink.net/~tyrerj/files/OG/pcm2pwm-03A.svgz
the red lines show the possible liner interpolation of the PCM signal.
---
OK, that makes more sense.
--- continue quote ---
Doesn't _any_ digital (PWM) system introduce distortion of the analog signal?
Yes, but it is all supposed to be above 1/2 the sample rate.
---
I understand that we can't just oversample more to reduce noise, but why
doesn't the noise-shaping trick of sigma-delta ADCs (integrator) work here?
We can just use a wide BPF, or even use the aforementioned brick wall filter
our ears have (if the frequency is high enough, which it should be with 192Khz
audio).
Yes, a Sigma-Delta DAC does work and it is used by some commercially
available chips.
One small issue. A Sigma-Delta PCM to PWM converter is actually a
Digital to Analog converter that contains analog integrators. I presume
that high end audio equipment would try to have the full 138.5 dB
dynamic range.
IAC, is it going to be able to generate a pulse width of 2.5 pico-seconds?
--- continue quote ---
Another possibility is that it might be possible to use delay lines
rather than a fast clock to do this. The Xilinx chip has DCMs rather
than PLLs and I don't fully understand how they work except they use
delay lines.
--- end of quote ---
I have always been afraid of using delay lines and such to increase the apparent
frequency, and this is especially true now that OGP has had issues with
producing correct video waveforms using those DCMs.
IIUC, the problem is that ground bounce is screwing up the DCM. If you
consider the delay lines to be analog and provide separate analog power
and ground, this problem goes away. IAC, this could probably be
generated with a sawtooth oscillator and comparators and flipflops.
--
JRT
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