Now how much of its improved SQ over rivals three or more times the price is going to be down to the active bass section offloading the most PSU-draining signals from the driving poweramp? The comments about the scale of the soundstage do reflect the kinds of improvements I've been getting by reducing LF noise in my DAC, so the reduced LF noise from the poweramp from having a more benign load to drive could indeed be key. 92dB efficiency certainly helps a lot in reducing poweramp PSU stress.
That article says this speaker is 'sure to shake up the industry'. Really? What do you guys think?
Instead of pulling the output current through R1, we add an npn pass transistor, Q1. The output current now "passes" through the transistor, while the Zener diode still regulates the output voltage by being connected to the transistor base.
The output impedance falls to a few ohms, but the ripple rejection improves only slightly.
This is a useful basic circuit block for audio, but the ripple rejection can easily improved by the addition of a couple of additional components, as we'll see shortly.
Posted 15th January 2014 at 11:34 PM byrjm Updated 16th January 2014 at 04:40 AM byrjm
This is the first of a series, where I will be investigating the output impedance and ripple rejection of various voltage regulator circuits using LTSpice.
Today, for the first "lesson" (I'm teaching myself, as much as anything) we will look at the very simple zener voltage regulator.
The load is 1 kohm, and the Zener breakdown voltage is 12 V. The load current is about 10 mA, and to avoid gross inefficiency we will limit the current flowing through the Zener to about 5 mA, by adjusting R1 accordingly. The input voltage is fixed at 18 V.
To measure the ripple rejection, we perform an AC analysis with the voltage source AC set to 1 and the current source AC set to zero. The ripple rejection is the negative value of the signal at Vout: so -20 dB means 20 dB ripple rejection (1 V ripple at Vin generates 0.1 V ripple at Vout at a given frequency.).
To measure the output impedance, again the AC analysis function is used but...
John Atkinson's 'WOW!' (more than once) to his measurement results from the Vivaldi DAC prompted me to have a closer look at how it does for noise modulation.
The AP's FFTs shown don't come with details about the number of points in the FFTs, so noise estimates are a tiny bit tricky. However there is enough detail to make some reasonable estimates.
I've attached the plot from which I'm making my estimates - if anyone notices I've made a slip-up, please do comment and correct me.
The red line shows white noise at peak level of -4dBfs. I generated white noise in Audacity at this level to compare - with the maximum 16k point FFT and BH windowing, I got the same level of noise as on this plot - -42dBfs in 22kHz bandwidth. That suggests to me that the FFT shown has 64k points.
Another way to estimate the noise is by looking at the difference between the blue plot (19.1kHz, 0dBfs) grass and the red. To my eye, the difference in level is 76dBr...
I've had the beta build Ozone (no input stage yet, fed from my QA550 via the I2S transcoder to down-convert to 32fs) playing out 24/7 for a few days now.
Overall I'm very happy with how it sounds, just a minor gripe about sibilance on some operatic vocals which I'd like to understand better. On the upside the jump-factor (read dynamics) and soundstage stability (holographic on the right disks) are about the best yet. I'm using a couple of Decca double CDs for this - 'La Traviata' and "La Boheme'. They're about the most transparent sounding and demanding disks I have. Demanding in the sense that they have lots of emotional drama which should positively engage my attention if the DAC's really up to snuff. I never much enjoyed opera until I got into building my own NOS DACs, but now I really enjoy my dramatic fixes and these two recordings are really top of the pile. They're about 50 years old but so far I've not found anything newer which touches them (not that I've looked...
I like to read and muse on what I'm reading. Human beings are interesting and trying to delve into the psyche of humans are fascinating. Subjectivists vs. objectivists, do cables/wires have an audible character, can fuses sound different from each other, linear vs. smps... The list goes on. We can't even agree on how listening are supposed to be done ABX, long term listening, whole songs or short passages, double blind or open, known material or unfamiliar.... The only thing we seam to be able to agree on is that we're mostly disagreeing about most things.
And it's incredible how much prestige people put in their opinions defending opinions as if their life depended on it.
I love a good discussion but sometimes it's just silly.
Posted 30th December 2013 at 03:35 AM bywlowes Updated 22nd September 2014 at 01:10 AM bywlowes(add pic)
Two months ago I had great ambitions to complete a long in the planning short on execution music server by xmas. Seems I am too busy and just having too much fun listening to my system to make progress on big projects.
However, in a very low key way my system has made some stunning gains over the 2013 holiday.
My happy 6 year journey with a Lightspeed linestage finally ran afoul when an LDR packed it in. I retooled with some on hand bits while waiting for new LDRs to arrive. A snaffu with my order delayed that whole process. Meanwhile, my wonderful wife asked me what I wanted for xmas. I had been reading Arthur Salvatore's site and became interested in using an autoformer from Dave Slagle for my linestage. If your read the reviews and the technical specs, its intriguing. My system is perfect for passive. I only listen to a music server with triode ouput stage, short interconnects to OTL amps and overall have lots of gain.
Posted 27th December 2013 at 06:08 AM byabraxalito Updated 30th December 2013 at 12:52 AM byabraxalito
Inspired by this thread http://www.diyaudio.com/forums/digit...-new-post.html
I've been giving a little thought for how to move beyond the 'Ozone' to a DAC able to handle more than 16bit inputs and up to 96kHz input rate. The target being 20bits and 120dB SNR (non-A weighted). The OP in that thread preferred a more marketable 'ultimate' DAC (with 192k and potentially 384k capability, along with DSD) - aims which to my way of seeing clearly conflict with a DAC having any pretentions to ultimacy.
The simplest solution - building on the digital part I already have - would be to add more TDA1545As in parallel and with a Cortex M4 direct the data to the respective chips. What's unclear though is how low the noise will go when the extra chips are added. From the DS, an A-weighted SNR is quoted of 101dB (typ) with 2mA, however this is a static noise (code = 0) and hence may well not translate to the noise achieved with bits toggling....
Well now I am on holidays over XMAS I got the time to really put in a solid chunk of time on the DDS based synthesiser.
Previous blog entries describe the PIC32MX based core to this. The thing works like a charm...
To get decent precision on the sinewave I have implemented an interpolation on a high precision Sinewave lookup table.
- The DDS references into a 12 bit "long", 24 bit "amplitude" precision sinewave LUT.
- Of itself this gives mediocre spurs, which in a DDS are heavily dependent on the frequency, but seem to result in 85-90dBc spurs. There is a fair hash of these without treatment.
- By adding a linear interpolation between samples in the Sinewave Lookup table, the spurs come out as shown below...
- The interpolation is actually quite simple in concept:
- The top 12 bits of the DDS Phase Accumulator looks up the "Sine Sample" ...