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DACless audio
Marco:
--- Quote from: tom66 on September 14, 2022, 03:13:34 pm ---This is because to get e.g. 16-bit PWM at 48kHz you would need an PWM clock of ca. 3GHz
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The effective PWM clock, physically it can be a bunch of delay shifted lower speed clocks.
langwadt:
--- Quote from: tom66 on September 14, 2022, 07:35:36 pm ---
--- Quote from: tszaboo on September 14, 2022, 03:47:31 pm ---But audio doesn't work that way. You don't need to reconstruct all the samples at 48khz, because the sound was recorded with ~10khz input bandwidth. Just look at DSD encoding. That's approximately CD quality at 2.8MHz. Similar, there are digital filters that convert it from PCM to PWM and the internal frequency of these signals are not going to be that high. They just move it out of the audible range, somewhere in a few MHz range, where it doesn't jam radios.
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Yes, that's delta-sigma encoding, but it's my understanding that's reserved for higher end class-D chips (if it's used at all).
Something like e.g. TAS2552 (I2S input class-D amplifier) is doing the PWM in analog, for a few reasons:
* Power FETs have a minimum on/off time which a simple delta-sigma modulator may not respect (requires additional care in the design of the modulator) - this isn't a problem if you're just driving logic into e.g. an LC filter network which most of those CD players are effectively doing
* EMC - a lot easier to jitter a centre carrier to spread the energy than to guarantee performance of a wide-spectrum noise generator (which is what delta sigma is) - EMC is of course an issue for many so-called filterless class D amps
* Doing it in analog is cheaper than in pure digital logic, esp. if you only need THD ~0.1% or so.
I haven't seen many class D DACs with digital inputs that could be called truly DAC-less. In terms of modulation done digitally the closest you might get are the Tripath ones, no longer manufactured, though they don't disclose how the modulator works and the devices only had analog inputs. I've had a play with modulation techniques, and I've got an FPGA design running at 600MHz to do some modulation of audio and there are a lot of difficult headaches that come when driving power transistors this way.
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yeh, doing it digitally you need very fast transistors and the PSRR is basically zero
Benta:
I remember that back in the late 1980s, Philips presented the 16-bit digital loudspeaker at some audio show/symposium.
The speaker element consisted of a small central diaphragm, surrounded by 15 concentric circular diaphragms of varying width; every diaphragm individually driven..
The idea was, that each diaphragm be assigned to a bit in the 16-bit audio signal: the central (smallest) LSB, the outer (largest) MSB.
Strict on/off control of each diaphragm at 44.1 kHz, volume control by varying the suppy voltage to the 16 output stages.
Never caught on, of course (far too expensive, though that's not really an issue for an audio buff), but it never got past the exhibition stage.
Crazy days back then...
jonpaul:
check out the oversampling ADC and DAC, available from AKM, Crystal/Cirrus, TI, since 1986.
Use a 1 bit high frequency sampler and digital filters and DSP decimation
Jon
tooki:
--- Quote from: tszaboo on September 14, 2022, 03:47:31 pm ---But audio doesn't work that way. You don't need to reconstruct all the samples at 48khz, because the sound was recorded with ~10khz input bandwidth.
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Sorry, what?!? 44.1kHz and 48kHz sample rates are used in order to cleanly cover the 20Hz-20kHz frequency range without needing brutally steep low pass filters.
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