Thanks Max for the answers despite I'm OT.
Yes venetian shellac for violins, I know it, said to be hard but in the same time have elastic properties (which seems the opposit of each others), very shiny though. Drole, it seemed your linseed oil + this particular vegetal resin gomma showed more a oil painter background....
strange strange this shellac on little chips as it's hard thin layer?! give a little less cooling on the chip so creating a nice johnson noise ???
Well doesn't matter, I will try. piezzo trap as graphic pen powder is on the surfaces ????
Let's go back to the thread, TDA1541 and R2R discrete dacs from EC Designs.
Yes venetian shellac for violins, I know it, said to be hard but in the same time have elastic properties (which seems the opposit of each others), very shiny though. Drole, it seemed your linseed oil + this particular vegetal resin gomma showed more a oil painter background....
strange strange this shellac on little chips as it's hard thin layer?! give a little less cooling on the chip so creating a nice johnson noise ???
Well doesn't matter, I will try. piezzo trap as graphic pen powder is on the surfaces ????Let's go back to the thread, TDA1541 and R2R discrete dacs from EC Designs.
Does the ETL192 module read only Wave files or also Flac ? Could you share prices specifications please as I am a young student and would like to know whether I will be able to afford it or not... ?
All the best sir!
Ben
Hi benou7580,
DA96ETF
Power supply: external, 5V/100mA (4.8 ... 5V5), connection USB-B (large). Matching low (ground-loop) noise, power supply is included.
Supply current: 25mA@44.1KHz (125 milliwatts), 35mA@96 KHz (175 milliwatts).
Digital audio interface: ElectroTos (Standard Toslink optical socket for use with / ElectroTos plug).
Supported protocol: ElectroTos low jitter protocol.
Note: the D/A96ETF does not support standard Toslink protocol, it will only produce noise when attempting to drive it with standard Toslink sources.
The DA96ETF pairs with U192ETL or UPL96ETL. These support the ElectroTos low jitter protocol.
-> ElectroTos low jitter protocol eliminates -unwanted- jitter in digital audio source, digital audio interlink, and all DAC circuits even when significant crosstalk is present. It has to be generated directly by software.
Sample rate support: 44.1, 48, 88.2, and 96 KHz.
Bit depth: automatic (ElectroTos protocol). U192ETL and UPL95ETL support 24 bit.
Internal digital audio interface system for the SIPO: (serial to parallel converter), 4 wire ElectroTos, ELE, ESC, ECK, and EDA.
Internal digital audio interface D/A converter: parallel interface.
Standard digital audio protocols are no longer used / generated anywhere in the DA96ETF.
D/A converter type: Fractal multi-pattern converter, 32 bit core / channel (this type of converter is -completely- different from all existing audio D/A converters).
Fractal multi-pattern converters offer extreme long term accuracy (with given component tolerances), low output impedance (375 OHms), and extremely low digital interface switching noise because of the parallel data interface (highest switching frequency equals the sample rate / ELE latch signal).
System: unfiltered NOS (fully passive output circuit). By using 88.2 or 96 KHz, 2 x oversampling can be obtained.
Output voltage: 3V6pp, DC-coupled.
Output connector: RCA, gold-plated, shielded connector soldered directly into the PCB for lowest possible noise / losses.
Output impedance: 375 OHms, can directly drive 250 OHms Beyerdynamic DT-990 studio or comparable headphones.
Recommended lossless volume control: Shunt volume control (shunt only), 375 OHm output impedance is used as fixed series resistor for the shunt volume control. This shunt regulator can be placed in parallel with Beyerdynamic DT-990 headphones for lossless volume control. Typical couput impedance 0.38 OHms (-60dB) ... 375 OHms (0dB). This way headphones can be driven without using any active linear circuit in the audio signal path, just a digital latch and a resistor network.
The DA96ETF consists of a logic board that contains all connectors, supply voltage filters, voltage regulators, ElectroTos receiver circuit, SIPO and indication LED (lock LED). The two Fractal multi-pattern D/A converters are located on a piggyback board that is plugged on top of the logic board, there is no wiring (very compact module, shortest possible connections).
I attached a sketch of the DA96ETF housing.
I assume you refer to the UPL96ETF WAV player. It supports WAV files that include meta-data (text, artwork). dbPoweramp or XLD can be used to easily generate such files. This is a top performance player so we only support WAV. FLAC support would lead to significant interference increase as it requires a more powerful processor. We kept this player as basic as possible and optimised the software to the limits in order to produce lowest possible noise. This WAV player also guarantees that the output is bit-perfect when the original audio file is bit-perfect. We advise to rip original CDs in order to be sure that the file hasn't been tampered with.
Prices will occur on our website as soon as the products become available.
DA96ETF
Power supply: external, 5V/100mA (4.8 ... 5V5), connection USB-B (large). Matching low (ground-loop) noise, power supply is included.
Supply current: 25mA@44.1KHz (125 milliwatts), 35mA@96 KHz (175 milliwatts).
Digital audio interface: ElectroTos (Standard Toslink optical socket for use with / ElectroTos plug).
Supported protocol: ElectroTos low jitter protocol.
Note: the D/A96ETF does not support standard Toslink protocol, it will only produce noise when attempting to drive it with standard Toslink sources.
The DA96ETF pairs with U192ETL or UPL96ETL. These support the ElectroTos low jitter protocol.
-> ElectroTos low jitter protocol eliminates -unwanted- jitter in digital audio source, digital audio interlink, and all DAC circuits even when significant crosstalk is present. It has to be generated directly by software.
Sample rate support: 44.1, 48, 88.2, and 96 KHz.
Bit depth: automatic (ElectroTos protocol). U192ETL and UPL95ETL support 24 bit.
Internal digital audio interface system for the SIPO: (serial to parallel converter), 4 wire ElectroTos, ELE, ESC, ECK, and EDA.
Internal digital audio interface D/A converter: parallel interface.
Standard digital audio protocols are no longer used / generated anywhere in the DA96ETF.
D/A converter type: Fractal multi-pattern converter, 32 bit core / channel (this type of converter is -completely- different from all existing audio D/A converters).
Fractal multi-pattern converters offer extreme long term accuracy (with given component tolerances), low output impedance (375 OHms), and extremely low digital interface switching noise because of the parallel data interface (highest switching frequency equals the sample rate / ELE latch signal).
System: unfiltered NOS (fully passive output circuit). By using 88.2 or 96 KHz, 2 x oversampling can be obtained.
Output voltage: 3V6pp, DC-coupled.
Output connector: RCA, gold-plated, shielded connector soldered directly into the PCB for lowest possible noise / losses.
Output impedance: 375 OHms, can directly drive 250 OHms Beyerdynamic DT-990 studio or comparable headphones.
Recommended lossless volume control: Shunt volume control (shunt only), 375 OHm output impedance is used as fixed series resistor for the shunt volume control. This shunt regulator can be placed in parallel with Beyerdynamic DT-990 headphones for lossless volume control. Typical couput impedance 0.38 OHms (-60dB) ... 375 OHms (0dB). This way headphones can be driven without using any active linear circuit in the audio signal path, just a digital latch and a resistor network.
The DA96ETF consists of a logic board that contains all connectors, supply voltage filters, voltage regulators, ElectroTos receiver circuit, SIPO and indication LED (lock LED). The two Fractal multi-pattern D/A converters are located on a piggyback board that is plugged on top of the logic board, there is no wiring (very compact module, shortest possible connections).
I attached a sketch of the DA96ETF housing.
Does the ETL192 module read only Wave files or also Flac ? Could you share prices specifications please as I am a young student and would like to know whether I will be able to afford it or not... ?
I assume you refer to the UPL96ETF WAV player. It supports WAV files that include meta-data (text, artwork). dbPoweramp or XLD can be used to easily generate such files. This is a top performance player so we only support WAV. FLAC support would lead to significant interference increase as it requires a more powerful processor. We kept this player as basic as possible and optimised the software to the limits in order to produce lowest possible noise. This WAV player also guarantees that the output is bit-perfect when the original audio file is bit-perfect. We advise to rip original CDs in order to be sure that the file hasn't been tampered with.
Prices will occur on our website as soon as the products become available.
Attachments
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Hi John,
Thank you for those informations!
Could you just describe a little bit more about what you said here - as it may be very difficult to understand for many people, like me :
Thank you for those informations!
Could you just describe a little bit more about what you said here - as it may be very difficult to understand for many people, like me :
Hi daanve,
Distortion in the audio signal path is cumulative, every component that is added -will- introduce more distortion and noise that is added to the total amount of noise and distortion.
The constant voltage steering of complex loads (speakers) produces highest distortion at the end of the audio signal path. In other words, constant voltage steering and complex impedances don't match very well, yes it sounds quite acceptable but it distorts. Even when using a perfect amp (zero distortion) we still end up with significant distortion at the speaker membrane.
I use a quasi constant power steering device for driving complex loads. This reduces distortion at the speaker membrane to almost inaudible levels. With this setup it is possible to detect distortion of active circuits for example that would be completely masked by distortion using conventional power amps. This quasi constant power driver (based on 8 medium power transistors) is the only active linear circuit in the entire audio signal path.
Active linear circuits add distortion and noise, and many oscillate at very high frequencies (many GHz) in order to maintain equilibrium. When a Op-amp gets quite hot it is also an indication that its oscillating. Almost all Op-amps do not remain stable when driving capacitive loads.
I tested active circuits for decades, every OP-amp, every discrete pre-amp and evert tube pre-amp adds coloration, none of these is able to maintain absolute transparency (almost zero noise, no audible coloration at all, and no audible loss in resolution and micro detail).
The easiest way by far to maintain best transparency is using -passive- DC-coupled circuits and lowest practical resistor / impedance values.
Therefore I only use parallel latches and a resistor network in the Fractal DAC. As indicated it offers 375 OHms output impedance using only latches and low Ohmic resistors.
I selected a specific logic family (after exhaustive testing) and this logic family has a typical supply voltage of 3V6, exceeding it has negative impact on logic circuit performance. So this sets the 3V6pp output amplitude.
With typical power amp gain of 40x and 3V6pp input we can generate 144Vpp or 50.9Vrms. Even with 8 Ohm speakers we then get 324W rms output per channel, with 4 OHm speakers we get 648W rms per channel. How much output power do you need?
The higher the DAC output voltage, the more attenuation is needed before amplification. This will always lead to poor S/N ratio.
I personally use a gain of 5x, it is more than sufficient driving my open baffle speakers with 90dB efficiency (1 x 8" full-range + 2 x 15"bass).
Distortion in the audio signal path is cumulative, every component that is added -will- introduce more distortion and noise that is added to the total amount of noise and distortion.
The constant voltage steering of complex loads (speakers) produces highest distortion at the end of the audio signal path. In other words, constant voltage steering and complex impedances don't match very well, yes it sounds quite acceptable but it distorts. Even when using a perfect amp (zero distortion) we still end up with significant distortion at the speaker membrane.
I use a quasi constant power steering device for driving complex loads. This reduces distortion at the speaker membrane to almost inaudible levels. With this setup it is possible to detect distortion of active circuits for example that would be completely masked by distortion using conventional power amps. This quasi constant power driver (based on 8 medium power transistors) is the only active linear circuit in the entire audio signal path.
Active linear circuits add distortion and noise, and many oscillate at very high frequencies (many GHz) in order to maintain equilibrium. When a Op-amp gets quite hot it is also an indication that its oscillating. Almost all Op-amps do not remain stable when driving capacitive loads.
I tested active circuits for decades, every OP-amp, every discrete pre-amp and evert tube pre-amp adds coloration, none of these is able to maintain absolute transparency (almost zero noise, no audible coloration at all, and no audible loss in resolution and micro detail).
The easiest way by far to maintain best transparency is using -passive- DC-coupled circuits and lowest practical resistor / impedance values.
Therefore I only use parallel latches and a resistor network in the Fractal DAC. As indicated it offers 375 OHms output impedance using only latches and low Ohmic resistors.
I selected a specific logic family (after exhaustive testing) and this logic family has a typical supply voltage of 3V6, exceeding it has negative impact on logic circuit performance. So this sets the 3V6pp output amplitude.
With typical power amp gain of 40x and 3V6pp input we can generate 144Vpp or 50.9Vrms. Even with 8 Ohm speakers we then get 324W rms output per channel, with 4 OHm speakers we get 648W rms per channel. How much output power do you need?
The higher the DAC output voltage, the more attenuation is needed before amplification. This will always lead to poor S/N ratio.
I personally use a gain of 5x, it is more than sufficient driving my open baffle speakers with 90dB efficiency (1 x 8" full-range + 2 x 15"bass).
Dear John,
Thank you for your answers and precisions.
I am looking for an excellent integrated amplifier that could match very well with your coming Fractal DAC and U192ETL. Would you give us any recommendation of a great sounding integrated amp (at a reasonnable price) that has little distortion and very good background noise management? What do you think about Class D amplifications from Pascal Audio ?
What could you say about the fact that computers are not "bit-perfect" player? Is it true?
Do you expect to launch UPL96ETL and U192ETL at the same time? What kind of sonic differences should we expect to hear comparing them? Will this be very important?
Many thanks for your great help and the time spent to answer us!
All the best,
Ben,
I don't know if anyone can answer your question about amplifiers. Nelson Pass would tell you the best amplifier for you is the one you most enjoy using. I like Class A and I use a First Watt F4 with an Aikido (valve) preamp. I like my Leak Stereo 20 and my 300B SET amp. I hate Naim - or at least I still did the last time I heard one. In the UK some of the forum members have 'bake offs' where a member will invite enthusiasts to bring their equipment for everybody to compare. Maybe this happens where you live too.
The pros and cons of 16bit vs 24bit in the current generation of UPL and Dac have been discussed previously - on here or elsewhere. Unless I am mistaken, the same comparisons will apply. I bought a 16 bit and Mos 16. I like them a lot.,
Hope this helps
Michael
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What about a Peter Daniel chip amp?
LM3875 Kit – Audiosector
Tons of info on DIYAUDIO. If you want a pair of assembled PCB's (premium version) and optionally the Plitron transformer that powers it, PM me as I have some no longer used.
Hi Benou7580:
The concept of (semiconductor) audio amplifiers is forcing a constant voltage on a complex load and hoping that the speaker membrane will accurately track this voltage. In practice this doesn't work as planned, the speaker membrane will not accurately track the power amplifier input signal at all, and there will be considerable distortion, even when using the perfect amplifier.
This resulted in many different amplifier concepts that all have advantages and disadvantages but none of these does everything right. The end result is always distortion that increases with increasing SPL.
When comparing very good amplifiers with a system that offers almost zero distortion there is a day and night improvement. Sound turns in to highly realistic music. So this amplifier distortion however small we think it is, prevents us from reproducing realistic music. This is a pity after going to extremes to get the source right.
I may have posted this before but please read this article:
The serious flaws of voltage drive | Current-Drive - The Natural Way of Loudspeaker Operation
So one would think, ok use constant current steering and all will be fine, unfortunately this is not the case. Think of speaker resonance where the speaker impedance peaks, this is only one of the problems with constant current steering. Most speakers are also more or less optimised for constant voltage steering, so constant current steering won't work well with these speakers.
What do we actually want? We want the speaker membrane to accurately track the amplifier input voltage, even when external forces (pressure) are working against or in favour of the membrane movement.
So I figured, why not ditch the classical open loop audio amplifier concept and replace it with a servo system. Philips made some attempts decades ago with the motional feedback system (MFB) for bass only.
What if we could turn the audio power amplifier into a large bandwidth servo system? We already have the feedback voltage (it is the ac voltage produced by a voice coil vibrating in a strong magnetic field).
I managed to build such servo driver (only works with one single loudspeaker). The speaker distortion basically vanished, and I obtained the most realistic detailed high resolution music reproduction.
This servo system produces no cross-over distortion at all and consumes constant power (power ratio method). The servo system parameters must be matched with the speaker being driven and multiple servo's and active crossover might be needed for a multi-way speaker.
The servo system won't work on multi-way speakers optimised for voltage steering. Modular or single (full-range) speakers could work with an universal servo driver that has adjustable parameters for matching servo and speaker.
My -personal- opinion about class D amps, not suitable for audiophile applications. Constant voltage steering is used and that results in unwanted speaker distortion. PWM introduces related distortion spectrum and a powerful switching noise spectrum. Class D amps internal clock must be synchronised with the clock of the digital audio source, if not inter-modulation occurs. Supply voltage must be -extremely- constant and totally free of any ripple voltage as this would change the PWM signal energy content and related output voltage. Clock needs to have -extremely- low jitter or we get similar jitter issues like with the DAC (DAC jitter issues multiplied by class D amplifier jitter issues).
The often applied comparator circuit (compares reference sawtooth or triangle clock with the analogue input signal) has limited resolution because of required hysteresis (to keep the comparator stable). The comparator also introduces more jitter. The required passive low-pass filter at the output will never be perfect and will introduce more (phase) distortion. The low-pass filter will interfere with the complex load (speaker) causing more distortion. Class D amps are very useful when saving power is most important and sound quality doesn't matter that much.
S/PDIF / I2S to PWM power conversion is likely to work better as some signal conversion errors can be eliminated. But problems remain to exist. I just think this will work better compared to A/D (comparator) followed by D/A (PWM -> low pass filter) conversion. The steering method is still the same (constant voltage steering) so distortion remains relatively high. Lossless volume control will also become problematic as SN ratio is now always at its worse and the theoretical amount of bits (24) cannot be realised with any existing audio set yet.
We developed a bit-perfect test years ago (Mosaic UV DAC). It uses a digital audio reference file. It measures if a digital audio source offers bit-perfect playback. Some sources do offer bit-perfect playback without further configuration, others need to be configured, tweaked hacked and there are some that cannot offer bit-perfect playback at all. Based on this one can never be sure until one measures it, measuring is knowing as we say over here in Holland.
We -could- design a simple, low cost USB stick size bit-perfect playback tester if people are interested.
The UPL96ETL will be introduced later as my brother is still working on firmware and PC app.
I assume that U192ETL is driven by a bit-perfect source.
You already tested the UPL16 with some good quality DACs using an experimental ElectroTos interlink?
The U192ETL with S/PDIF protocol exceeds UPL16 / UPL24 performance.
The UPL96ETL with S/PDIF tops this performance, not a day and night difference but clearly audible increased resolution and darker background, and of course the certainty of bit-perfect playback.
When switching to the ElectroTos low jitter protocol (only usable with the DA96ETF) there will be a day and night improvement relative to S/PDIF related performance levels.
Here again the UPL96ETL tops the U192ETL.
Both (using ElectroTos low jitter protocol and DA96ETL fractal DAC) are able to offer close to live music reproduction when using suitable servo drivers / speakers and good quality live recordings.
All recordings will sound different (recording quality is -very- clearly audible now). But every recording is way more enjoyable now and audiophile playback is no longer restricted to few audiophile recordings.
The concept of (semiconductor) audio amplifiers is forcing a constant voltage on a complex load and hoping that the speaker membrane will accurately track this voltage. In practice this doesn't work as planned, the speaker membrane will not accurately track the power amplifier input signal at all, and there will be considerable distortion, even when using the perfect amplifier.
This resulted in many different amplifier concepts that all have advantages and disadvantages but none of these does everything right. The end result is always distortion that increases with increasing SPL.
When comparing very good amplifiers with a system that offers almost zero distortion there is a day and night improvement. Sound turns in to highly realistic music. So this amplifier distortion however small we think it is, prevents us from reproducing realistic music. This is a pity after going to extremes to get the source right.
I may have posted this before but please read this article:
The serious flaws of voltage drive | Current-Drive - The Natural Way of Loudspeaker Operation
So one would think, ok use constant current steering and all will be fine, unfortunately this is not the case. Think of speaker resonance where the speaker impedance peaks, this is only one of the problems with constant current steering. Most speakers are also more or less optimised for constant voltage steering, so constant current steering won't work well with these speakers.
What do we actually want? We want the speaker membrane to accurately track the amplifier input voltage, even when external forces (pressure) are working against or in favour of the membrane movement.
So I figured, why not ditch the classical open loop audio amplifier concept and replace it with a servo system. Philips made some attempts decades ago with the motional feedback system (MFB) for bass only.
What if we could turn the audio power amplifier into a large bandwidth servo system? We already have the feedback voltage (it is the ac voltage produced by a voice coil vibrating in a strong magnetic field).
I managed to build such servo driver (only works with one single loudspeaker). The speaker distortion basically vanished, and I obtained the most realistic detailed high resolution music reproduction.
This servo system produces no cross-over distortion at all and consumes constant power (power ratio method). The servo system parameters must be matched with the speaker being driven and multiple servo's and active crossover might be needed for a multi-way speaker.
The servo system won't work on multi-way speakers optimised for voltage steering. Modular or single (full-range) speakers could work with an universal servo driver that has adjustable parameters for matching servo and speaker.
My -personal- opinion about class D amps, not suitable for audiophile applications. Constant voltage steering is used and that results in unwanted speaker distortion. PWM introduces related distortion spectrum and a powerful switching noise spectrum. Class D amps internal clock must be synchronised with the clock of the digital audio source, if not inter-modulation occurs. Supply voltage must be -extremely- constant and totally free of any ripple voltage as this would change the PWM signal energy content and related output voltage. Clock needs to have -extremely- low jitter or we get similar jitter issues like with the DAC (DAC jitter issues multiplied by class D amplifier jitter issues).
The often applied comparator circuit (compares reference sawtooth or triangle clock with the analogue input signal) has limited resolution because of required hysteresis (to keep the comparator stable). The comparator also introduces more jitter. The required passive low-pass filter at the output will never be perfect and will introduce more (phase) distortion. The low-pass filter will interfere with the complex load (speaker) causing more distortion. Class D amps are very useful when saving power is most important and sound quality doesn't matter that much.
S/PDIF / I2S to PWM power conversion is likely to work better as some signal conversion errors can be eliminated. But problems remain to exist. I just think this will work better compared to A/D (comparator) followed by D/A (PWM -> low pass filter) conversion. The steering method is still the same (constant voltage steering) so distortion remains relatively high. Lossless volume control will also become problematic as SN ratio is now always at its worse and the theoretical amount of bits (24) cannot be realised with any existing audio set yet.
We developed a bit-perfect test years ago (Mosaic UV DAC). It uses a digital audio reference file. It measures if a digital audio source offers bit-perfect playback. Some sources do offer bit-perfect playback without further configuration, others need to be configured, tweaked hacked and there are some that cannot offer bit-perfect playback at all. Based on this one can never be sure until one measures it, measuring is knowing as we say over here in Holland.
We -could- design a simple, low cost USB stick size bit-perfect playback tester if people are interested.
The UPL96ETL will be introduced later as my brother is still working on firmware and PC app.
I assume that U192ETL is driven by a bit-perfect source.
You already tested the UPL16 with some good quality DACs using an experimental ElectroTos interlink?
The U192ETL with S/PDIF protocol exceeds UPL16 / UPL24 performance.
The UPL96ETL with S/PDIF tops this performance, not a day and night difference but clearly audible increased resolution and darker background, and of course the certainty of bit-perfect playback.
When switching to the ElectroTos low jitter protocol (only usable with the DA96ETF) there will be a day and night improvement relative to S/PDIF related performance levels.
Here again the UPL96ETL tops the U192ETL.
Both (using ElectroTos low jitter protocol and DA96ETL fractal DAC) are able to offer close to live music reproduction when using suitable servo drivers / speakers and good quality live recordings.
All recordings will sound different (recording quality is -very- clearly audible now). But every recording is way more enjoyable now and audiophile playback is no longer restricted to few audiophile recordings.
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Dear John,
Such great news you shared here with us!
Thank you also for your opinion about Class D amps, servo loudspeakers and the difference in sound quality between the UPL96 and the U192.
We would be very thankful to you if you enable us to get our hand on bit-perfect test softwares through a USB key from you. Would it be possible to make it available to purchasing at the same time than the U192 launch?
Can't wait to hear your loudspeakers when it will be ready! ^^'
Good job about that at it seems very difficult to servo a subwoofer !
Many thanks again,
Ben
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Dear John,
What about your Analog interconnects' development projects? Could you say a little bit more about that? What have been your findings?
Same questions for open baffle loudspeakers and stepped shunt volume control for the Fractal DAC ?
What will be V/I converter? Is it linked or comparable to Servo technology or is it something completely new?
What would be the purpose of the headphones buffer (PVI/HB) you would like to create? Would it be like a traditionnal amplifier for headphones?
Many thanks in advance for your time and answer!
Ben
What about your Analog interconnects' development projects? Could you say a little bit more about that? What have been your findings?
Same questions for open baffle loudspeakers and stepped shunt volume control for the Fractal DAC ?
What will be V/I converter? Is it linked or comparable to Servo technology or is it something completely new?
What would be the purpose of the headphones buffer (PVI/HB) you would like to create? Would it be like a traditionnal amplifier for headphones?
Many thanks in advance for your time and answer!
Ben
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John of ECDesigns: "I managed to build such servo driver (only works with one single loudspeaker). The speaker distortion basically vanished, and I obtained the most realistic detailed high resolution music reproduction."
I would like to point out a potential flaw of this servo approach of the power amplifier.
First, how do we obtain the music file? It was recorded and digitized. When it was recorded, there was a conversion sound pressure vibration through air to oscillate membrane of the microphone, which was then converted to a voltage signal, digitized to be the music source for today's digital music files.
The diaphragm of the speaker and the membrane of the microphone have different dynamic characteristics. Making the speaker diaphragm to follow the voltage signal exactly does not guarantee the resulting air pressure will be the same as the air pressure of the original musical performance.
Now, there are great debates even just for the power amplifier (mainly tube amps) regarding the global feedback, which essentially is the servo concept. Many, many tube amp designers swear that the global feedback kills the sound. Personally, I really don't experience it or am sensitive enough to tell the difference; however, a closed-loop feedback system (servo) still has tons of issues with tracking error, frequency bandwidth, instability, etc., discussed in tons of research papers on control theories.
John, as a creative person, is always looking at different angles and he should be commanded for it. Music reproduction is very subjective. Eventually, I, as a layman listener, is looking for enjoyment. When I listen to a particular music, I want it to soothe me and to stimulate something inside me. Perfectly accurate or not is secondary. Sometimes, some distortions might be welcome by a human mind.
Just my thought.
Toh
I would like to point out a potential flaw of this servo approach of the power amplifier.
First, how do we obtain the music file? It was recorded and digitized. When it was recorded, there was a conversion sound pressure vibration through air to oscillate membrane of the microphone, which was then converted to a voltage signal, digitized to be the music source for today's digital music files.
The diaphragm of the speaker and the membrane of the microphone have different dynamic characteristics. Making the speaker diaphragm to follow the voltage signal exactly does not guarantee the resulting air pressure will be the same as the air pressure of the original musical performance.
Now, there are great debates even just for the power amplifier (mainly tube amps) regarding the global feedback, which essentially is the servo concept. Many, many tube amp designers swear that the global feedback kills the sound. Personally, I really don't experience it or am sensitive enough to tell the difference; however, a closed-loop feedback system (servo) still has tons of issues with tracking error, frequency bandwidth, instability, etc., discussed in tons of research papers on control theories.
John, as a creative person, is always looking at different angles and he should be commanded for it. Music reproduction is very subjective. Eventually, I, as a layman listener, is looking for enjoyment. When I listen to a particular music, I want it to soothe me and to stimulate something inside me. Perfectly accurate or not is secondary. Sometimes, some distortions might be welcome by a human mind.
Just my thought.
Toh
Just one more quick follow-up on the closed-loop speaker system, discussed by John of ECDESIGNS. The ultimate closed-loop speaker system will have a microphone (or several) placed at near the listener position. The microphone signal then is used as a feedback to control the signal to drive the speakers. Some live instrument performance can be used for calibration.
Dear John,
I hope all is going well for you.
Please, could you consider answering to those only 3 small questions, as it may help us when it will be the time to purchase the U192ETL, for instance.
- Will the U192ETL be compatible with music playback that allows some convolution correction? (through Foobar2000 or Jriver)
- Could you say that high-end USB cables will enable to get us a better sound or will it be perfect with the piece you will deliver with the U192ETL? Same question for power supply of both Fractal DAC and UPL96ETL.
- What have been your findings about your RCA interlinks project that may be used between U192ETL and your Fractal DAC? Have you planned to propose it to purchase in the future?
Many thanks in advance,
All the best.
Ben
As far as I can see, this cannot work because of the time delay. A feedback system should have no or very limited time delay to work properly.
There is always time delay in any closed-loop feedback control system. Some time delay is long, much longer than the sampling period and must be compensated for. Some time delay is short, which will cause phase shifts. I prefer human ears as the sensor for feedback. The closed loop is your adjustment of various components to get the sound to your liking.
Theoretically, the time delay is the time needed for the sound to travel from the speaker to your ears. If you sit 3 meters away, it would take about 3/343 seconds, approximately 10 ms, which is rather long. Put the microphone right next to the speaker will cut it down to very short to mitigate the time delay issue.
Who knows? John might build it and it could sound amazing!
Toh
Theoretically, the time delay is the time needed for the sound to travel from the speaker to your ears. If you sit 3 meters away, it would take about 3/343 seconds, approximately 10 ms, which is rather long. Put the microphone right next to the speaker will cut it down to very short to mitigate the time delay issue.
Who knows? John might build it and it could sound amazing!
Toh