X100 build
...it's been a long time coming, but I think I'm over due to build this. I've most of the parts here or on the way, so I'm getting organized to build. I thought I'd put this on the forum to keep me on task.
State of play to start...
State of play to start...
Dim bulb tester, build one in five minute.
- By chrisng
- Power Supplies
- 27 Replies
Using a Yellow Jacket 2177N 4 outlet power block to make a DBT:
1. Unscrew four screws on both sides of the yellow jacket to remove the metal bottom cover.
2. Unscrew six screws on the inner box (4 on top, 2 on bottom) to expose the wiring.
3. Cut the two wires (see 3rd picture) and connect them together with a wire nut.
4. Assemble everything back together.
5. Plug in the bulbs, Done!
1. Unscrew four screws on both sides of the yellow jacket to remove the metal bottom cover.
2. Unscrew six screws on the inner box (4 on top, 2 on bottom) to expose the wiring.
3. Cut the two wires (see 3rd picture) and connect them together with a wire nut.
4. Assemble everything back together.
5. Plug in the bulbs, Done!
Noisy preouts on an Arcam AVR350
- By sploo
- Solid State
- 6 Replies
I've picked up a used Arcam AVR350 to use in a simple 2.1 configuration. I've owned an AVR300 since new so I'm reasonably familiar with the kit.
Excusing the "quality" diagram below... the setup is a pair of speakers powered by the receiver's own FL and FR amplifier channels, and an external power amp connected to the SUB preout on the receiver; driving a passive subwoofer.
It was immediately obvious that the subwoofer output is very noisy. A loud buzzing sound at any volume level (except for when the receiver is muted). Sound from the FL and FR speakers is fine (no noticeable hiss unless I turn the receiver volume up to levels that I wouldn't want to use for listening).
Having (unsuccessfully) gone through all the usual checks of grounding issues I reduced the setup to just a single speaker, connected to the FR preout on the receiver, and powered by the same external amp:
That still resulted in a lot of noise (interestingly the FR channel via the receiver's own amp is fine, but noisy via the FR preout).
It could point to the external amp being the problem, so I simplified things even further, and just attached a digital recorder to the FR preout phono/RCA plug. Other than (obviously) power to the receiver there's nothing connected (no speakers, no input sources etc):
As far as I understand (with this receiver), when an analog source is selected you'll always get output to the FL and FL preouts. If the DSP is off (Direct button pressed) then it's a simple pass though, and with the DSP on the input gets processed, then sent to the preouts.
Generating a spectrum plot from the preouts with no input source but the amp volume turned up high shows a spike at 50Hz (UK mains frequency) but then significant spikes every 100Hz. This is present regardless of whether the DSP is on (top) or off (bottom). Understandably there's lower overall noise with the DSP turned off (Direct mode).
I've reproduced the above behaviour on different preout channels (though obviously for a stereo input all other channels are off when the DSP is off, as only FL and FR inputs get passed through in Direct mode).
So; it appears that anything that comes out of the preout phonos is noisy (with or without the DSP enabled), and anything that comes out of the receiver's own amp channels is clean (regardless of whether it's first gone via the DSP or not).
The affected board is the MULTI board (sometimes misspelled MUTI in the service manual). Having pulled the board from the chassis, you can see 4 of the 8 preout phonos (labelled in green). The row of capacitors labelled red are (what I assume to be) part of the line level amplifiers for each preout. On the right (labelled in blue) there's a capacitor (C680) and a black cable going to a ground point near the power supply:
The schematic for this board shows the 7 speaker preouts + sub preout on the right, and capacitor C680 is at the top of the page below:
Based on this section of the schematic I'm assuming this area around C680 is providing a ground for this board (the upside down triangle mark). Could it be as simple as a faulty C680 capacitor causing grounding issues and spikes every 100Hz (that only appears to affect the preouts)? I don't have an electronics background so I'm basically at the limit of my knowledge here:
Excusing the "quality" diagram below... the setup is a pair of speakers powered by the receiver's own FL and FR amplifier channels, and an external power amp connected to the SUB preout on the receiver; driving a passive subwoofer.
It was immediately obvious that the subwoofer output is very noisy. A loud buzzing sound at any volume level (except for when the receiver is muted). Sound from the FL and FR speakers is fine (no noticeable hiss unless I turn the receiver volume up to levels that I wouldn't want to use for listening).
Having (unsuccessfully) gone through all the usual checks of grounding issues I reduced the setup to just a single speaker, connected to the FR preout on the receiver, and powered by the same external amp:
That still resulted in a lot of noise (interestingly the FR channel via the receiver's own amp is fine, but noisy via the FR preout).
It could point to the external amp being the problem, so I simplified things even further, and just attached a digital recorder to the FR preout phono/RCA plug. Other than (obviously) power to the receiver there's nothing connected (no speakers, no input sources etc):
As far as I understand (with this receiver), when an analog source is selected you'll always get output to the FL and FL preouts. If the DSP is off (Direct button pressed) then it's a simple pass though, and with the DSP on the input gets processed, then sent to the preouts.
Generating a spectrum plot from the preouts with no input source but the amp volume turned up high shows a spike at 50Hz (UK mains frequency) but then significant spikes every 100Hz. This is present regardless of whether the DSP is on (top) or off (bottom). Understandably there's lower overall noise with the DSP turned off (Direct mode).
I've reproduced the above behaviour on different preout channels (though obviously for a stereo input all other channels are off when the DSP is off, as only FL and FR inputs get passed through in Direct mode).
So; it appears that anything that comes out of the preout phonos is noisy (with or without the DSP enabled), and anything that comes out of the receiver's own amp channels is clean (regardless of whether it's first gone via the DSP or not).
The affected board is the MULTI board (sometimes misspelled MUTI in the service manual). Having pulled the board from the chassis, you can see 4 of the 8 preout phonos (labelled in green). The row of capacitors labelled red are (what I assume to be) part of the line level amplifiers for each preout. On the right (labelled in blue) there's a capacitor (C680) and a black cable going to a ground point near the power supply:
The schematic for this board shows the 7 speaker preouts + sub preout on the right, and capacitor C680 is at the top of the page below:
Based on this section of the schematic I'm assuming this area around C680 is providing a ground for this board (the upside down triangle mark). Could it be as simple as a faulty C680 capacitor causing grounding issues and spikes every 100Hz (that only appears to affect the preouts)? I don't have an electronics background so I'm basically at the limit of my knowledge here:
Khozmo Display Help
I have a dual Khozmo attenuator set-up and somehow I seem to have "un-synced" the display brightness. The right display is one step brighter than the left and I do not know if there is a way to re-sync them.
It is most apparent at each end of the brightness range. Khozmo has been less than responsive lately at least via e-mail.
Any ideas?
Thanks
Dave
It is most apparent at each end of the brightness range. Khozmo has been less than responsive lately at least via e-mail.
Any ideas?
Thanks
Dave
Sony micro plug
- By safe1
- Headphone Systems
- 7 Replies
Hi all,
Sony back in the 90's was using a so called Micro Plug to connect their earphones to the remote control and then to their walkmans of that era.
The plug is looking like the picture attached.
I have a few Sony walkmans and I need to build an adapter "Sony Micro-plug" to "female 2.5mm regular phono plug".
Does someone know how this type of plug is called so I can look it up in mouser, or digikey?
Sony back in the 90's was using a so called Micro Plug to connect their earphones to the remote control and then to their walkmans of that era.
The plug is looking like the picture attached.
I have a few Sony walkmans and I need to build an adapter "Sony Micro-plug" to "female 2.5mm regular phono plug".
Does someone know how this type of plug is called so I can look it up in mouser, or digikey?
Transistor active low pass filter
- By Tom Vyborny
- Electronic Design
- 5 Replies
Hi Guys,
I have been aske by one of my customers about designing a transistor 12 band graphic eq. My first idea was to use gyrators but the customer stated that he wanted to use active LC filter with discrete transistors. I do not have any experience with that but this is the result that LT Spice is showing. What do you think about that or do you have any experience with that? Thank you very much.
I have been aske by one of my customers about designing a transistor 12 band graphic eq. My first idea was to use gyrators but the customer stated that he wanted to use active LC filter with discrete transistors. I do not have any experience with that but this is the result that LT Spice is showing. What do you think about that or do you have any experience with that? Thank you very much.
PeeCeeBee Preamplifier GB
- By shaan
- Group Buys
- 581 Replies
Hello everyone.
The PeeCeeBee Preamplifier is here! PCBs will be available via group buys arranged in this thread.
_________________________________
UPDATE October 8 2021: All boards from last batch gone. Entry open for 3rd group buy. Shipping has resumed to most countries.
_________________________________
BOM with parts placement indication uploaded >Here<.
Setup Guide uploaded >Here<
Drill Template uploaded >Here<
_________________________________
Introduction:
It is a fully bipolar transistor based symmetrical "Class-A" push-pull preamplifier with real-time adjustable pure Single Ended harmonic distortion level - from less than 0.001% to over 1% - with second harmonic being predominant in all levels! It also includes its own low noise regulated power supply section on-board, that uses the same transistors used in the active section. It can be used as a very good headphone amplifier for voice coil impedance of 32ohm and above. By using a higher gain than the default +6dB, it can be made to put out 10VPP into 600ohm headphones or upto 25VPP to power-follower type amplifiers.
Description:
The preamplifier consists of two gain segments per channel, one push-pull (PP) and one single ended (SE), preamp output is taken from the PP segment. The PP segment uses low gain with very high global negative feedback and generates a clear replica of the input signal with minuscule THD. The SE segment uses high gain with zero global negative feedback and produces the same signal with massive amounts of low-order THD. The PP segment is directly fed the input signal from the volume control. The SE segment's output is coupled to the feedback node of the PP segment, and the SIG ("Signature") potentiometer controls the input signal to the SE segment. With SIG control set to lowest, the SE segment will contribute almost nothing in the main output due to its high output impedance compared to the feedback node of the PP segment. As we gradually increase the input to the SE segment, its output will present gradually more harmonic content to the feedback node of the PP segment where it will be compared with the feedback signal from the output of PP segment, and the difference between the two signals is presented to preamp's output. Since high amount of harmonic content at high frequencies is often unwelcome, the input to the SE segment can be lowpass filtered with user chosen maximum frequency of interest, so that the high frequency signals can be chosen to contain the least amount of harmonics, and treble definition remains intact.
How it sounds:
It sounds neutral, and it sounds warm, and everything in between! How much neutral or warm depends on how much you want to turn the SIG pot! In my system I have been listening to it from morning to midnight, day after day, with the SIG pot turned towards a low setting during daytime and to a slightly higher setting in the evening.
As expected, in its minimum SIG setting, the sound is extremely clean, with a large soundstage, and in its maximum setting, it is extremely sweet (masked), with a narrow soundstage. In my main system I found the sweet spot to be around 5-10% of the turn when playing music tracks where a lot is going on, and around 45-50% when I am listening to chants or other music types where the music is minimal, calm and relaxed. At near 20% I feel the sound is very balanced, with the right amount of depth and detail I like, along with a touch of that "SE" niceness.
I feel that at the highest and lowest settings the system will have a tendency to become "special case", i.e. in both these extremes it may choose to sound very good with some types of music and not so good with other types, subjectively speaking! You feel the sound is not coming good for the music? Turn it slowly and find a sweet spot. But it is Extremely important that you wait a few minutes after reaching the new setting! Read on to know why.
The design was finished in March! Although the pandemic and subsequent suspension of shipping services delayed everything, I kept listening to the veroboard prototype everyday. And something very interesting and important regarding the SIG control has become clear to me in the last couple months.
The position of the SIG control that suits your taste must be found by spending some time listening in a set position. The potentiometer can be turned without any problem during music playback. But instantly finding out a sweet-spot this way is almost impossible - at that new setting it will sound pretty much the same as the last setting and make you feel like the control is not working, although distortion may have changed by an order of magnitude or more!
This is because our aural perception needs time to notice changes in the level of harmonic content, and significantly more time to make a memory out of it. So it is suggested that the user sets the SIG level to a position and then gives it some time of listening, then changes the setting to another level and repeat the listening. Expect at least a day of listening to find all the sweet spots for your system. I say sweet spot's' because if you listen to multiple types of music, you will almost certainly find one sweet spot for each type. Each listener will have his/her own sweet spots and the speed of finding them will vary, as some listeners have a more acute perception than others. That said, as I mentioned previously, around 20% setting the preamp can sound good for almost all types of music. Links to a collection of the harmonic levels at different output voltage into different load impedances are shared below in the "THD Spectrums" section. Note that there is about 1dB of output signal attenuation when the SIG potentiometer is at its maximum, this doesn't cause any problem during listening.
Side Note: I noticed that the phase of the second harmonic relative to the input signal was easy to manipulate by choosing the polarity of Q7/Q8 transistors (output of SE segment). Since March I have been using an NPN here which was adding negative phase second harmonic to the output. Recently I changed it to PNP and got the positive phase second harmonic. At first it sounded quite different, and felt like the overall definition has increased. It got me excited and I posted a page worth of it >here<, thinking this is the way to move forward with the preamp. But after a couple days of listening it became apparent that the sonic appeal of the system has been lost with the positive phase second harmonic! I could not keep playing it for more than 15 minutes in a go! There is no high-order harmonics present beyond the 4th, but the sound started to feel thin and kind of bitter. Off to the workbench and reverted back to NPN, and the lushness of the output came back! Hence, back to old way (the better sounding way). The NPN actually outputs normal phase second harmonic, but as the SE segment's output is coupled to the feedback node (which is the inverting input of the PP segment), the harmonics get inverted at the output.
_____________________________________
All active devices are commonly available discrete bipolars. The PCB is a bit large, I decided to make it a little more spacious (I received many complaints about how congested my amplifier PCBs are. They're still congested! 🙂 ).
The onboard power supply is comprised of simple dual rail CRC filters followed by regulated capacitance multipliers in Sziklai follower config, powered from a 15V-0-15V/30VA center tapped transformer. There are two pots in the power supply section (VR5,VR6). Using these two pots the rails can be closely matched between +/-12V to +/-14.5V, and each rail can be trimmed individually. Instead of two large reservoir capacitors it uses 32 smaller capacitors, having much better filtering performance and keeping the profile low. Even with 40mA bias per channel I can hear absolutely no hum from the speakers or headphones. The preamp has no audible residual wideband noise, there is virtually zero hiss with headphones or speakers over ear.
Drill template and setup guide are being prepared and will be shared in a few days.
______________________________________
Input Impedance: 67Kohm
Output Impedance: 32ohm
Layout snapshot:
Schematic:
The prototype:
Input-Output delay:
Slew rate:
Harmonic waveform (at 1% THD):
Video Snapshots:
Power Supply Rail Behavior -> Power Supply Rail Behavior
Power On/Off Transient Output -> Power On/Off Transient Output
Clipping Waveform -> Clipping Waveform
THD Spectrums (Peak-to-Peak output, Gain +6dB, 1KHz):
10Kohm load:
500mV - Low SIG, High SIG
1V - Low SIG, High SIG
2V - Low SIG, High SIG
4V - Low SIG, High SIG
8V - Low SIG, High SIG
680ohm load:
1V - Low SIG, High SIG
2V - Low SIG, High SIG
4V - Low SIG, High SIG
8V - Low SIG, High SIG
100ohm load:
500mV - Low SIG, High SIG
1V - Low SIG, High SIG
2V - Low SIG, High SIG
4V - Low SIG, High SIG
-------------------------------------
All the above shared documents and pictures can be availed in a single zip file >Here<.
-------------------------------------
-------------------------------------
The PCBs are 1.6mm FR4 material with 60micron copper, black solder mask, white silkscreen and HASL finish. Dimensions: 170.18mm x 119.38mm.
-------------------------------------
Important information regarding group buy:
>> The PCBs cost USD20 for one unit.
>> International payments are accepted through PayPal only and payments from within India are accepted through bank account transfer/deposit, NEFT, IMPS and UPI.
>> Delay between placing the batch order to PCB plant and delivery of the boards to me will be 20-25 days.
>> Shipping is done for 3-4 packets a day to avoid stagnation at the clearing hubs due to the slow movement of international shipments.
Thanks.
shaan
The PeeCeeBee Preamplifier is here! PCBs will be available via group buys arranged in this thread.
_________________________________
UPDATE October 8 2021: All boards from last batch gone. Entry open for 3rd group buy. Shipping has resumed to most countries.
_________________________________
BOM with parts placement indication uploaded >Here<.
Setup Guide uploaded >Here<
Drill Template uploaded >Here<
_________________________________
Introduction:
It is a fully bipolar transistor based symmetrical "Class-A" push-pull preamplifier with real-time adjustable pure Single Ended harmonic distortion level - from less than 0.001% to over 1% - with second harmonic being predominant in all levels! It also includes its own low noise regulated power supply section on-board, that uses the same transistors used in the active section. It can be used as a very good headphone amplifier for voice coil impedance of 32ohm and above. By using a higher gain than the default +6dB, it can be made to put out 10VPP into 600ohm headphones or upto 25VPP to power-follower type amplifiers.
Description:
The preamplifier consists of two gain segments per channel, one push-pull (PP) and one single ended (SE), preamp output is taken from the PP segment. The PP segment uses low gain with very high global negative feedback and generates a clear replica of the input signal with minuscule THD. The SE segment uses high gain with zero global negative feedback and produces the same signal with massive amounts of low-order THD. The PP segment is directly fed the input signal from the volume control. The SE segment's output is coupled to the feedback node of the PP segment, and the SIG ("Signature") potentiometer controls the input signal to the SE segment. With SIG control set to lowest, the SE segment will contribute almost nothing in the main output due to its high output impedance compared to the feedback node of the PP segment. As we gradually increase the input to the SE segment, its output will present gradually more harmonic content to the feedback node of the PP segment where it will be compared with the feedback signal from the output of PP segment, and the difference between the two signals is presented to preamp's output. Since high amount of harmonic content at high frequencies is often unwelcome, the input to the SE segment can be lowpass filtered with user chosen maximum frequency of interest, so that the high frequency signals can be chosen to contain the least amount of harmonics, and treble definition remains intact.
How it sounds:
It sounds neutral, and it sounds warm, and everything in between! How much neutral or warm depends on how much you want to turn the SIG pot! In my system I have been listening to it from morning to midnight, day after day, with the SIG pot turned towards a low setting during daytime and to a slightly higher setting in the evening.
As expected, in its minimum SIG setting, the sound is extremely clean, with a large soundstage, and in its maximum setting, it is extremely sweet (masked), with a narrow soundstage. In my main system I found the sweet spot to be around 5-10% of the turn when playing music tracks where a lot is going on, and around 45-50% when I am listening to chants or other music types where the music is minimal, calm and relaxed. At near 20% I feel the sound is very balanced, with the right amount of depth and detail I like, along with a touch of that "SE" niceness.
I feel that at the highest and lowest settings the system will have a tendency to become "special case", i.e. in both these extremes it may choose to sound very good with some types of music and not so good with other types, subjectively speaking! You feel the sound is not coming good for the music? Turn it slowly and find a sweet spot. But it is Extremely important that you wait a few minutes after reaching the new setting! Read on to know why.
The design was finished in March! Although the pandemic and subsequent suspension of shipping services delayed everything, I kept listening to the veroboard prototype everyday. And something very interesting and important regarding the SIG control has become clear to me in the last couple months.
The position of the SIG control that suits your taste must be found by spending some time listening in a set position. The potentiometer can be turned without any problem during music playback. But instantly finding out a sweet-spot this way is almost impossible - at that new setting it will sound pretty much the same as the last setting and make you feel like the control is not working, although distortion may have changed by an order of magnitude or more!
This is because our aural perception needs time to notice changes in the level of harmonic content, and significantly more time to make a memory out of it. So it is suggested that the user sets the SIG level to a position and then gives it some time of listening, then changes the setting to another level and repeat the listening. Expect at least a day of listening to find all the sweet spots for your system. I say sweet spot's' because if you listen to multiple types of music, you will almost certainly find one sweet spot for each type. Each listener will have his/her own sweet spots and the speed of finding them will vary, as some listeners have a more acute perception than others. That said, as I mentioned previously, around 20% setting the preamp can sound good for almost all types of music. Links to a collection of the harmonic levels at different output voltage into different load impedances are shared below in the "THD Spectrums" section. Note that there is about 1dB of output signal attenuation when the SIG potentiometer is at its maximum, this doesn't cause any problem during listening.
Side Note: I noticed that the phase of the second harmonic relative to the input signal was easy to manipulate by choosing the polarity of Q7/Q8 transistors (output of SE segment). Since March I have been using an NPN here which was adding negative phase second harmonic to the output. Recently I changed it to PNP and got the positive phase second harmonic. At first it sounded quite different, and felt like the overall definition has increased. It got me excited and I posted a page worth of it >here<, thinking this is the way to move forward with the preamp. But after a couple days of listening it became apparent that the sonic appeal of the system has been lost with the positive phase second harmonic! I could not keep playing it for more than 15 minutes in a go! There is no high-order harmonics present beyond the 4th, but the sound started to feel thin and kind of bitter. Off to the workbench and reverted back to NPN, and the lushness of the output came back! Hence, back to old way (the better sounding way). The NPN actually outputs normal phase second harmonic, but as the SE segment's output is coupled to the feedback node (which is the inverting input of the PP segment), the harmonics get inverted at the output.
_____________________________________
All active devices are commonly available discrete bipolars. The PCB is a bit large, I decided to make it a little more spacious (I received many complaints about how congested my amplifier PCBs are. They're still congested! 🙂 ).
The onboard power supply is comprised of simple dual rail CRC filters followed by regulated capacitance multipliers in Sziklai follower config, powered from a 15V-0-15V/30VA center tapped transformer. There are two pots in the power supply section (VR5,VR6). Using these two pots the rails can be closely matched between +/-12V to +/-14.5V, and each rail can be trimmed individually. Instead of two large reservoir capacitors it uses 32 smaller capacitors, having much better filtering performance and keeping the profile low. Even with 40mA bias per channel I can hear absolutely no hum from the speakers or headphones. The preamp has no audible residual wideband noise, there is virtually zero hiss with headphones or speakers over ear.
Drill template and setup guide are being prepared and will be shared in a few days.
______________________________________
Input Impedance: 67Kohm
Output Impedance: 32ohm
Layout snapshot:
Schematic:
The prototype:
Input-Output delay:
Slew rate:
Harmonic waveform (at 1% THD):
Video Snapshots:
Power Supply Rail Behavior -> Power Supply Rail Behavior
Power On/Off Transient Output -> Power On/Off Transient Output
Clipping Waveform -> Clipping Waveform
THD Spectrums (Peak-to-Peak output, Gain +6dB, 1KHz):
10Kohm load:
500mV - Low SIG, High SIG
1V - Low SIG, High SIG
2V - Low SIG, High SIG
4V - Low SIG, High SIG
8V - Low SIG, High SIG
680ohm load:
1V - Low SIG, High SIG
2V - Low SIG, High SIG
4V - Low SIG, High SIG
8V - Low SIG, High SIG
100ohm load:
500mV - Low SIG, High SIG
1V - Low SIG, High SIG
2V - Low SIG, High SIG
4V - Low SIG, High SIG
-------------------------------------
All the above shared documents and pictures can be availed in a single zip file >Here<.
-------------------------------------
-------------------------------------
The PCBs are 1.6mm FR4 material with 60micron copper, black solder mask, white silkscreen and HASL finish. Dimensions: 170.18mm x 119.38mm.
-------------------------------------
Important information regarding group buy:
>> The PCBs cost USD20 for one unit.
>> International payments are accepted through PayPal only and payments from within India are accepted through bank account transfer/deposit, NEFT, IMPS and UPI.
>> Delay between placing the batch order to PCB plant and delivery of the boards to me will be 20-25 days.
>> Shipping is done for 3-4 packets a day to avoid stagnation at the clearing hubs due to the slow movement of international shipments.
Thanks.
shaan
LM334 cascode CCS
- By beetel
- Tubes / Valves
- 17 Replies
I am not able to properly implement this circuit with 12AX7 tube preamp in LTSPICE. LTSPICE file is attached. Simulation is not giving proper result. Similar circuit with LM317 show correct simulation in LTSPICE. I am trying to learn about use of CCS in tube circuits.
Tube amplifiers with PMC EB1i speakers
- By VladimirK
- Tubes / Valves
- 111 Replies
About 10 years ago I purchased PMC EB1i loudspeakers for use with solid state amplifiers. These were industrial AB class amplifiers, and selfmade single-ended class A amplifiers, some of which I discussed on this forum. After some time, I began to try to connect tube amplifiers to these loudspeakers, in particular, I purchased a NEM A300SE single-ended tube amplifier. While I liked the sound with tube amplifiers more, it was obvious that with these large loudspeakers, tube amplifiers did not provide the ability to sound full at low frequencies. So I started trying single-ended designs with larger output tubes. First, with GM70 output tubes, then I assembled output monoblocks with GU48 tubes (a complete analogue of 833, but with a metal anode). The impressions from the evolution of output tubes are such that the more power dissipated at the anode, the more confident and full the system with these loudspeakers sounds. The output impedances of all three tube amplifier options were 0.65 ohms with the 300B tubes (special low Rout version), 0.8 ohms with the GM70, and 0.8 ohms with the GU48. The sound impressions increased with the increase in the power dissipated at the anode. Tube PP tube amplifiers did not fit into my preferences, the sound was not as interesting as with single-ended amplifiers. Since the NEM A300SE amplifier was little used, I converted it into a driver unit. To do this, all layers of the secondary winding were connected in series (instead of parallel connection, the heater voltage for 300B was rectified, the ripple of the anode voltage was also additionally suppressed. As a result, now there are two independent driver units, and two independent sets of output units. From listening experience, I can say, that for full sounding it is important to reduce the output impedance of the driver units (but not by using cathode followers). Both units use interstage transformers at the output. One driver unit has UO186 (YO186) tubes at the output and an interstage transformers with 2: 1 step down. Another driver unit is with 300B tubes at the output and a step-down interstage transformer 3 : 1 (this ratio has occured when the sections of the secondary winding were connected in series.) Now a completely harmonious sounding system has been created, and most importantly, all the attractiveness of the sound of single-ended tube amplifiers has been preserved.
No sound just static/hum on brand new input selector. Please help
- By Kelvin75
- Solid State
- 21 Replies
Attachments
Configuration of the ES9038Q2M for synchronous operation
- By Yuri PL
- Electronic Design
- 2 Replies
Hi,
I am working on design, that uses ES9038Q2M as auxiliary DAC chip. I am attempting to run its in synchronous mode, however it is constantly misbehaving.
The system is as follows:
-MCLK is 22.5792/24.576MHz, delivered from low jitter clock distribution directly to the DAC chip.
-Signal source is on board XU208, LRCK and BCLK are delivered directly to DAC chip via short, impedance matched traces (signal integrity is perfect, as far as my 500MHz scope can measure).
-Interface is set to 32 bit I2S.
I am attempting following configuration:
-In register 10, I am setting 128fs mode bit to enabled
-Register 12 is set to 0 (DPLL disabled)
-Register 1 is set to 0xC0 (32 bit input, standard I2S mode, force fixed input type, input type SERIAL)
-Register 6 is set to 0x00 (linear phase fast roll-off filter), but leaving it on default values have no effect
-Register 15 and 16 are set to 0x00 (full volume).
-Register 14 is set to enable soft start and begin outputting analog signal.
During sample rate changes, I am attempting to set Reg 0 CLK_GEAR to value that always matches 128*fs relationship, that is:
-0b10 for 44.1/48ksps
-0b01 for 88.2/96ksps
-0b00 for 192/176.4ksps
However, resultant DAC behavior is clearly abnormal, that is - produced spectral response is completely skewed.
For 44.1/48, both distortion and bandwidth are excellent:
However, for any higher sample rate, two problems appear. First, the bandwidth fall to -3db at 20kHz, second - massive aliasing appears out of band:
So, I have realized i have not set ASRC disable bit in the register 27. I have done so, and it caused even worse problems.
At 44.1ksps, there is no gain loss at 20kHz, however - noise floor is massively elevated, and 0dbfs output became limited to -20db.
At 192ksps, we still have aliases and too fast roll-off, along with reduced full scale amplitude.
So, as a next step, I have decided to disable clock gearing and leave it at 0b00 (full input clock frequency) regardless of sample rate.
Now, at 44.1ksps, noise floor is elevated, 0dbfs is limited by 20db, and there is roll-off at 20khz.
At 192ksps, 0dbfs point of DAC is still reduced by 20db, noise floor is still elevated, but there is no drop at 20kHz and no aliasing:
Lacking any idea what this chip tries to do, I have decided to leave clock gear fixed at 0b00, but enable ASRC (reg 27 bit 7 set to 1).
Now, at 192ksps everything works properly. No elevated noise floor, no 20kHz drop, no aliases and 0dbfs input produces full scale on DAC output.
But on 44.1ksps, while all other problems disappear, 20kHz drops have appeared.
Honestly, I am stumped. Does anyone have experience with implementing synchronous mode in this chip, and can suggest what I am missing?
I am happy to provide any additional information. 🙂
Best regards
I am working on design, that uses ES9038Q2M as auxiliary DAC chip. I am attempting to run its in synchronous mode, however it is constantly misbehaving.
The system is as follows:
-MCLK is 22.5792/24.576MHz, delivered from low jitter clock distribution directly to the DAC chip.
-Signal source is on board XU208, LRCK and BCLK are delivered directly to DAC chip via short, impedance matched traces (signal integrity is perfect, as far as my 500MHz scope can measure).
-Interface is set to 32 bit I2S.
I am attempting following configuration:
-In register 10, I am setting 128fs mode bit to enabled
-Register 12 is set to 0 (DPLL disabled)
-Register 1 is set to 0xC0 (32 bit input, standard I2S mode, force fixed input type, input type SERIAL)
-Register 6 is set to 0x00 (linear phase fast roll-off filter), but leaving it on default values have no effect
-Register 15 and 16 are set to 0x00 (full volume).
-Register 14 is set to enable soft start and begin outputting analog signal.
During sample rate changes, I am attempting to set Reg 0 CLK_GEAR to value that always matches 128*fs relationship, that is:
-0b10 for 44.1/48ksps
-0b01 for 88.2/96ksps
-0b00 for 192/176.4ksps
However, resultant DAC behavior is clearly abnormal, that is - produced spectral response is completely skewed.
For 44.1/48, both distortion and bandwidth are excellent:
However, for any higher sample rate, two problems appear. First, the bandwidth fall to -3db at 20kHz, second - massive aliasing appears out of band:
So, I have realized i have not set ASRC disable bit in the register 27. I have done so, and it caused even worse problems.
At 44.1ksps, there is no gain loss at 20kHz, however - noise floor is massively elevated, and 0dbfs output became limited to -20db.
At 192ksps, we still have aliases and too fast roll-off, along with reduced full scale amplitude.
So, as a next step, I have decided to disable clock gearing and leave it at 0b00 (full input clock frequency) regardless of sample rate.
Now, at 44.1ksps, noise floor is elevated, 0dbfs is limited by 20db, and there is roll-off at 20khz.
At 192ksps, 0dbfs point of DAC is still reduced by 20db, noise floor is still elevated, but there is no drop at 20kHz and no aliasing:
Lacking any idea what this chip tries to do, I have decided to leave clock gear fixed at 0b00, but enable ASRC (reg 27 bit 7 set to 1).
Now, at 192ksps everything works properly. No elevated noise floor, no 20kHz drop, no aliases and 0dbfs input produces full scale on DAC output.
But on 44.1ksps, while all other problems disappear, 20kHz drops have appeared.
Honestly, I am stumped. Does anyone have experience with implementing synchronous mode in this chip, and can suggest what I am missing?
I am happy to provide any additional information. 🙂
Best regards
Easy to carry outdoor audio using mobile battery (Hi-Res and DSD support)
- By nandappe
- Full Range
- 1 Replies
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Question for Home Audio System
- By Brad0920
- Everything Else
- 2 Replies
I bought a house that already has a Bose system and speakers installed throughout the house. I'd like to use it. What do I need to plug in so that it can function? There are volume knob on each floor and speakers on the celling. Thanks in advance!
