MPP

The 9V cells i use ( Alkali-Manganese ) have 280mA per hour so they can supply 20mA for 14 hours.
There are new cells based on the old Edison patent ( Zink-Carbon ) that could fulfill your requirement. I did not try them.
I friend of mine has build a preamp where the bias of the Fets is done by batteries. It works since at least 10 years.
Panasonic makes so called Eneloops that have superb low self discharge. There is a huge
variati of alternatives.
 
Here is a lineup of what is available .....
 

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I have a Billie Holiday recording for example and i listen to her voice.
The voice appears in the middle of the speakers as a phantom image.
When focus is good i can hear how far that voice is away from me and how high.
Also when it is well recorded the voice should not occupy too much space and there should be an impression that the image has sharp boundaries like in a good photo where the objects are well separated. On a few exceptional recordings there can even be the ilusion to hear " around " objects.

I have a question too. In your cap multiplier thread you showed a high voltage version that can cary 30mA. Can it be modified so that it can stand 50mA ?
 
Well, I definitely know what you mean by focus. I find harmonica is the hardest to achieve focus with. I did that very rarely with my BJT amplifiers.

I don't see why this shouldn't be able to output up to 50mA, and I'm not sure why I limited it to 30mA. That may be left from when I wanted to use a BC3x7 as the output.

Keantoken's CFP cap multiplier - Page 24 - diyAudio

PS. I see that the forum changed it's post link convention, so maybe direct post links will work for people with different posts-per-page settings.
 
The input stage is floating and works on batteries.
It is not a transconductance design and not a transimpedance design eather.
I call it "Power Transfer Input" because it has an input impedance of ca. 5 Ohm that is similar to the DC impedance of low output moving coils.
It is a modified common base design with base current recycling and what i call a " Hum Bugger". It is not really balanced but the earth pins of the cartridge are not directly connected to audio ground. It suppresses hum nearly as well as a balanced input but without the complication of doubling the circuitry.

I Hope it gets success, as it is a very nice concept. :)
If it is, what I think it is. ;)
 
Here is what I've come up with. This preserves the natural focus of the buffer. I'm not sure why. As you can see I used a toroidal coil to rule out interference (and also just for fun). The coil I used is actually 500nH, but 2uH is what it takes for unconditional stability, and still sounds great. You don't necessarily need the 47R output resistor, but it keeps the cable from resonating too bad, and it is something I hesitate to leave out of any buffer.

I tried switching the collector of Q1 to the emitter of Q2. This sounds very good, but has .01% THD with H2 at -80db. I thought it may have better focus but I'm really not sure. In any case the effect is very enjoyable. However ultimately I settled on this version for sounding more accurate in general, to me at least, and having less distortion.
 

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Yes, you inspired me to design it and i learned from you.
MiiB even build a balanced version. In his version the current for the input stage is supplied by solar panels i think. The panels are in a closed box and bulbs light the panels.
He had quite a few problems with heat.

Joachim, I have also learned a lot from you, especially in controlling noise.
It is for me a honor , to have inspired you in creating such a nice circuit. :)
 
I use an FFT resampler that was a little hard to get. Most resamplers use varying kinds of interpolation which can only be approximate. "Oversampling" on DACs is still some kind of interpolation. FFT resampling in a DAC would require a microprocessor to apply the FFT algorithm. Using an FFT however allows a much more exact conversion. It seems more and more producers are considering this.

The worst errors occur when converting between 44.1k and 48k. I helped with the development of Synthfont (MIDI sampler/sequencer) and suggested to the developer to include an FFT resampling algorithm. Since most of the samples available are 48k or 44.1k, this improved resampling quite a bit and really helped to reduce that "MIDI sound". This is very important for sequenced music.

2L has put up some music samples that were resampled using a "saracon" resampler, which I guess is popular. I wrote an email pointing out the heterodynes in the spectrogram compared to the non-resampled version but did not receive a response.

I get a kick out of playing their 350KHz samplerate files, FFT resampled to 192KHz. I don't think this has much to do with the samplerate, but I think the 2L recordings are still really great.

As far as sample conversion however, the most interesting thing to me is that the largest audible effect seems to be in the bass and lower midrange, not necessarily the treble where you would think.
 
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