Marantz PM 500 working again
- By MadMeier
- Solid State
- 0 Replies
It was in pretty bad shape with various burnt parts ...
... not always consistent with the service manual I found online (e.g. R743..746) are 390 Ohm and not 10 Ohm
... and some changes in the area of the speaker protection which I had to replace
Does somebody have experience with the model?
... not always consistent with the service manual I found online (e.g. R743..746) are 390 Ohm and not 10 Ohm
... and some changes in the area of the speaker protection which I had to replace
Does somebody have experience with the model?
InConcert Duke
Attachments
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InConcert Duke 2_page-0001.jpg -
InConcert Duke 2_page-0002.jpg -
IMG_20230420_194506.jpg -
IMG_20230420_194522.jpg -
IMG_20230420_194543.jpg -
Opstelling Duke - Basis.jpg -
Opstelling Duke - Basis2.jpg -
Opstelling Duke - Basis3.jpg -
Opstelling Duke.jpg -
Opstelling Duke-2.jpg -
Opstelling Duke-3.jpg -
Opstelling Duke-4.jpg
Replacing Broken 1/4” Horizontal SM Jack
- By widmark
- Construction Tips
- 13 Replies
Hi fellow tinkerers. Need to source and replace a broken 1/4” horizontal surface mount instrument TS input jack for a Focusrite brand audio interface. The model is the Clarett 8PreX. It’s a HighZ input jack used for direct connection to an electric guitar, primarily.
Any help finding a solid replacement would be greatly appreciated. Because this is used for audio recording and mixing, a high quality replacement in terms of specs and build are the goal. Thanks in advance!
P.S. - I asked ChatGPT to analyze the photo and tell me who made the existing jack, and below was the response. I didn’t find a jack with this name in my search:
Any help finding a solid replacement would be greatly appreciated. Because this is used for audio recording and mixing, a high quality replacement in terms of specs and build are the goal. Thanks in advance!
P.S. - I asked ChatGPT to analyze the photo and tell me who made the existing jack, and below was the response. I didn’t find a jack with this name in my search:
“From the embossed “J.T.” marking and the style of the jack, it appears these are made by Changzhou Jutai Electronics Co., Ltd. (“Jutai” for short), a Chinese/Taiwanese manufacturer whose phone‐jack housings are often branded with “J.T.” in that same rectangular logo. Although Focusrite does not publicly publish the exact Jutai model number, these typically fall into Jutai’s “PJ-63x” or “PJ-62x” family of 6.35 mm (1/4″) horizontal PCB‐mount jacks.”
Pearl 2 phono stage, Pass DIY version plus partial completion kit
Pass DIY Pearl 2 phono stage kit including PCB set and jfets, plus most of the components to stuff the boards, all for $180 shipped to the USA. A list of the included components follows below. Dale 1/4 watt resistors, Wima caps, etc, from Mouser. Ordered in a COViD isolation haze in 2021, but I got impatient and purchased a phone stage instead.
The caps and regulators for the power supply section are NOT included because I intended to use a Salas shunt regulated power supply instead.
The caps and regulators for the power supply section are NOT included because I intended to use a Salas shunt regulated power supply instead.
The 'Circles of Doom’... Open baffleless full range speakers
- By bushmeister
- Multi-Way
- 607 Replies
Attachments
Please Help replacing HA1457 with OPA604
- By moshi
- Solid State
- 7 Replies
i have these amps (Sony TA-F6b, Kenwood KA-405, JVC JA-s44)with faulty HA1457's that need to be replaced with OPA604, which components from the existing boards should i add, change values or completely remove for it to work properly? here's the schematics
For Sale Lots of connections brand names brand new and soldering station and soldering fan
- By Jeffrey 01
- Swap Meet
- 3 Replies
Attachments
First attempt at refoaming speakers
Good evening, I wanted to share my journey in refoaming two speakers my dear father gifted me many, many years ago.
I don't know much about the quality of them, and honestly I don't care, they have an emotional value to me so I just didn't want to toss them away or replace them when I could have just tried to change the foam myself.
All the information relative to this are well written and illustrated on my blog https://www.mimifactory.com/, otherwise this post would've become too long.
The speakers were in really bad shape. In 15+ years the foam completely degrated and turned into dust..
I got some replacement rings of the right size, cleaned the metal frame and the cone really well with isopropyl alcohol and removed the cover of the voice coil with a heat gun (making the glue soft)
As you can see I put some paper in the gap between the voice coil and the magnet, so the cone was centered and not scraping against the metal after gluing. It would've made a terrible sound and damaged the speakers.
As for the glue, I used some water soluble glue. In case I need to refoam them in the future... If I used epoxy or superglue that would have been a nightmare
After gluing the central part of the ring, I waited until it was dry, then glued the external part of the ring, the black gasket and applied pressure on it until everything was solid
..and don' forget to remove the paper and glue the voice coil cover back.
Overall it was quite tedious and time consuming, but I successfully refoamed both speakers and they sound good with no weird noises or scratching.
I don't know much about the quality of them, and honestly I don't care, they have an emotional value to me so I just didn't want to toss them away or replace them when I could have just tried to change the foam myself.
All the information relative to this are well written and illustrated on my blog https://www.mimifactory.com/, otherwise this post would've become too long.
The speakers were in really bad shape. In 15+ years the foam completely degrated and turned into dust..
I got some replacement rings of the right size, cleaned the metal frame and the cone really well with isopropyl alcohol and removed the cover of the voice coil with a heat gun (making the glue soft)
As you can see I put some paper in the gap between the voice coil and the magnet, so the cone was centered and not scraping against the metal after gluing. It would've made a terrible sound and damaged the speakers.
As for the glue, I used some water soluble glue. In case I need to refoam them in the future... If I used epoxy or superglue that would have been a nightmare
After gluing the central part of the ring, I waited until it was dry, then glued the external part of the ring, the black gasket and applied pressure on it until everything was solid
..and don' forget to remove the paper and glue the voice coil cover back.
Overall it was quite tedious and time consuming, but I successfully refoamed both speakers and they sound good with no weird noises or scratching.
OpAmp driven diamond buffer amplifier
- By No ideas
- Solid State
- 119 Replies
i think i biased it a little too much seems to consume 3W at idle.
DIY Sony VFET pt 1
- By Nelson Pass
- Pass Labs
- 1543 Replies
Now is the winter of our discontent made glorious summer by sunny skies
and celebration of my 20 years on this forum, not to mention recent
vaccinations...
I present the DIY Sony VFET pt 1, detailing the design for the P channel
amplifier version using up the last of the Sony 2SJ28 VFETs, to be followed
by part 2, using up the 2SK82's.
diyAudio will be offering a limited quantity of complete kits of both versions,
so watch for the announcement in the store.
Later there will also be offerings for those who have these and the other
Sony VFETs already, and also other FET devices.
Further reading and discussion:
and celebration of my 20 years on this forum, not to mention recent
vaccinations...
I present the DIY Sony VFET pt 1, detailing the design for the P channel
amplifier version using up the last of the Sony 2SJ28 VFETs, to be followed
by part 2, using up the 2SK82's.
diyAudio will be offering a limited quantity of complete kits of both versions,
so watch for the announcement in the store.
Later there will also be offerings for those who have these and the other
Sony VFETs already, and also other FET devices.
Further reading and discussion:
Loudspeaker Enclosures are Waveguides
- Multi-Way
- 81 Replies
I believe that you can use the shape of your loudspeaker as a waveguide.
I've mentioned this offhand in a few threads, but I thought I'd provide some evidence. Basically I'd been obsessing about the shape of my newest loudspeaker and some people were wondering why I was being so obsessive about the width and the depth of the enclosure.
The answer? Because the width and the depth of the enclosure act as a waveguide.
To demonstrate, here's a 35.6cm wide waveguide mounted in a baffle that 122cm wide:
In this sim, you can see the expected behavior of a waveguide. Basically it controls directivity down to 964Hz, and then as the wavelengths become larger than the waveguide, the directivity broadens. Basically any wavelength larger than 964Hz are too large for the waveguide to control, and due to this, the beamwidth gets broader.
Here's the exact same waveguide, same driver. But this time, I've narrowed the baffle by 66.6%, down to 40.67cm. More importantly, the depth of the baffle is now 40.67cm.
By making the baffle narrower, the beamwidth of the waveguide is narrower across it's entire bandwidth, and directivity control is extended downwards by about one octave.
Here's a real world example of the phenomenon. This is the Revel M106 speaker. It features a 15cm wide waveguide mounted on a baffle that's 21cm wide and 28cm deep.
Conventional waveguide theory suggests that the M106 waveguide will control directivity down to 2267Hz. This is because the M106 waveguide is 15cm wide, and 2267Hz is 15cm long.
But in the measured beamwidth of the M106, we see beamwidth control down to approximately 400Hz.
I believe that the enclosure is acting as a waveguide.
I believe that the width and the depth of the enclosure, and the angle of the walls, is not just cosmetic. It's been specifically designed to provide directivity control to a frequency that's much lower than what the waveguide on the baffle provides.
It's a neat trick I think, because most of us thought we needed waveguides that are nearly a meter in diameter to control directivity down to 300-500Hz. But we don't. You just have to manipulate the width and depth of the enclosure, and likely make some adjustments to the angle of the sidewalls.
I've mentioned this offhand in a few threads, but I thought I'd provide some evidence. Basically I'd been obsessing about the shape of my newest loudspeaker and some people were wondering why I was being so obsessive about the width and the depth of the enclosure.
The answer? Because the width and the depth of the enclosure act as a waveguide.
To demonstrate, here's a 35.6cm wide waveguide mounted in a baffle that 122cm wide:
In this sim, you can see the expected behavior of a waveguide. Basically it controls directivity down to 964Hz, and then as the wavelengths become larger than the waveguide, the directivity broadens. Basically any wavelength larger than 964Hz are too large for the waveguide to control, and due to this, the beamwidth gets broader.
Here's the exact same waveguide, same driver. But this time, I've narrowed the baffle by 66.6%, down to 40.67cm. More importantly, the depth of the baffle is now 40.67cm.
By making the baffle narrower, the beamwidth of the waveguide is narrower across it's entire bandwidth, and directivity control is extended downwards by about one octave.
Here's a real world example of the phenomenon. This is the Revel M106 speaker. It features a 15cm wide waveguide mounted on a baffle that's 21cm wide and 28cm deep.
Conventional waveguide theory suggests that the M106 waveguide will control directivity down to 2267Hz. This is because the M106 waveguide is 15cm wide, and 2267Hz is 15cm long.
But in the measured beamwidth of the M106, we see beamwidth control down to approximately 400Hz.
I believe that the enclosure is acting as a waveguide.
I believe that the width and the depth of the enclosure, and the angle of the walls, is not just cosmetic. It's been specifically designed to provide directivity control to a frequency that's much lower than what the waveguide on the baffle provides.
It's a neat trick I think, because most of us thought we needed waveguides that are nearly a meter in diameter to control directivity down to 300-500Hz. But we don't. You just have to manipulate the width and depth of the enclosure, and likely make some adjustments to the angle of the sidewalls.
SSE let out some smoke
Hello all it's been awhile since i posted anything, my Tubelab SSE has been working for years now with no issues. Last year i purchased a set of KT88's and have been running them since. After a move i hooked up the amp and puff, smoke arose.
Has anyone had an issue with R14 blowing up? Anything else i should replace along with R14?
Any help would be greatly appreciated.
Has anyone had an issue with R14 blowing up? Anything else i should replace along with R14?
Any help would be greatly appreciated.
Hi folks, looking for help/advice on my 5532 buffer line driver
- By BARuk
- Analog Line Level
- 32 Replies
So I am designing a line driver with 5532 opamps. The PCB I'm trying to make is for 12x buffer stages to drive 12x 600ohm vintage line transformers, simply to add colour to a signal straight from a modern rack digital audio interface. All of the boards and transformers will be mounted in a 19" rack unit.
I opted to try and make the signal fully differential going in and out of the device, so its balanced going back into the analog inputs on the audio interface.
This is my first build of this type and I guess I just wanted some reassurance before I make this PCB design.
I need it to amplify 6dB and be adjustable via trimmer.
Ive put together a linear regulated power supply that I intend to have on a seperate board and connect to the buffer boards with connectors. ERC is just pointing at the Mains AC earth that will go to unit chassis, and the regulator inputs , i guess because i havent specified any actual voltage sources:
Here is two channels of my balanced line driver circuit. I plan to have 6 per board ( 2 boards ) Let me know if theres anything I could change or improve. I created it on breadboard today and it seemed to work well. I had to play around with resistor values to get the gain trimmer values in good ranges:
I didnt add any caps to GND or inductors on the opamp outputs, which I still might.
The output 3 pin connectors will go to 2 core shielded wire to the audio transformers. The secondaries on the audio transformers will go to the back panel TRS sockets of the unit. I understand the Shields on the output TRS need to be grounded to chassis.
I might run a simulation to make sure everything is ok
Any input really helps 🙂
I opted to try and make the signal fully differential going in and out of the device, so its balanced going back into the analog inputs on the audio interface.
This is my first build of this type and I guess I just wanted some reassurance before I make this PCB design.
I need it to amplify 6dB and be adjustable via trimmer.
Ive put together a linear regulated power supply that I intend to have on a seperate board and connect to the buffer boards with connectors. ERC is just pointing at the Mains AC earth that will go to unit chassis, and the regulator inputs , i guess because i havent specified any actual voltage sources:
Here is two channels of my balanced line driver circuit. I plan to have 6 per board ( 2 boards ) Let me know if theres anything I could change or improve. I created it on breadboard today and it seemed to work well. I had to play around with resistor values to get the gain trimmer values in good ranges:
I didnt add any caps to GND or inductors on the opamp outputs, which I still might.
The output 3 pin connectors will go to 2 core shielded wire to the audio transformers. The secondaries on the audio transformers will go to the back panel TRS sockets of the unit. I understand the Shields on the output TRS need to be grounded to chassis.
I might run a simulation to make sure everything is ok
Any input really helps 🙂
No iron deficiency here
- By Fuling
- Tubes / Valves
- 11 Replies
My latest breadboard, a 3 stage 300B SET with interstage transformers between each stage.
The tube/transformer lineup is: CV1135 - LL1660 mA - 6S4S - LL1671 50mA - 300B - LL1679 70mA.
Each stage has its own tube rectified HT supply, 6AX5GT for the input and drivers and dual 6AU4GTA for the output tubes. Coleman regs for the direct heated drivers and output tubes.
Plenty of hum to get rid of before I can say much about the sound...
The tube/transformer lineup is: CV1135 - LL1660 mA - 6S4S - LL1671 50mA - 300B - LL1679 70mA.
Each stage has its own tube rectified HT supply, 6AX5GT for the input and drivers and dual 6AU4GTA for the output tubes. Coleman regs for the direct heated drivers and output tubes.
Plenty of hum to get rid of before I can say much about the sound...
Peavey CS-1200 amplifer, DC present on both channels
- By douglasfir97
- Solid State
- 10 Replies
Attachments
HELP: 3-Way DSP Class D Mono Build
- By thepix
- Digital Line Level
- 3 Replies
I'm in the process of building some 3-Way Studio Monitors which would require approximately 2x100W + 1x200W @ 4Ohms, with DSP that would work with both Balanced and Digital inputs.
With the prices of the Fusion FAX53 Amps, I've been plugging away for weeks on KiCad and I believe I'm 90-95% there with a schematic, but wanted to get some feedback as to whether or not its a decent design, if there are any errors in my thinking and any other general considerations that I should probably be making?
I've attached the block diagram for reference questions and comments are all welcome, bearing in mind I have as close to zero knowledge in this field other than what has been learned along the way.
With the prices of the Fusion FAX53 Amps, I've been plugging away for weeks on KiCad and I believe I'm 90-95% there with a schematic, but wanted to get some feedback as to whether or not its a decent design, if there are any errors in my thinking and any other general considerations that I should probably be making?
I've attached the block diagram for reference questions and comments are all welcome, bearing in mind I have as close to zero knowledge in this field other than what has been learned along the way.
Attachments
Versatile 4-Channel Guitar Preamp Design
- Instruments and Amps
- 20 Replies
4-Channel Preamp Design for Guitar
This is an experimental design for a solid-state preamp made using op-amps and is intended to result in a guitar preamp that is more versatile than typical. I start with the input stage, which is simply a normal voltage amp using an OPA2134 op-amp, chosen because it is reported to be one of the cleanest op-amps around, having noise figures better than most other op-amps on the market, and which was purpose designed for professional audio applications in which low noise is of primary concern.
The circuit is also a distribution amp, though gain is rather low. Yet, the gain should be sufficient to send the input signal to four separate channels. I have shown it with input and output jacks on purpose. Each sub-circuit of the project is shown this way so as to emphasize the versatility of the project. Making each sub-circuit’s input and output user accessible, say via a patch-bay in the back of the completed project’s enclosure, allows the user to patch the sub-circuits together as preferred, and to treat each in/out interface as an effects loop. An Input Level control has been included to tame large input signals that result from hot pickups and stomp-box effects, if needed. The gain here can be set from unity to 25, and each output has its own volume control.
As can be seen below, each channel is enabled independently of the others. I suggest using Channel 1 as a direct channel, though with a gain stage and active tone-control, so the input signal can keep up with the signals from the other three channels.
There is no digital circuitry in this project, although the switching IC (a CD4066B quad bilateral switch) is often used in digital equipment. Rather, the switching IC is only there to perform as a quiet switch the user employs to enable a given channel.
The genius of this IC in audio applications is that it makes doing quiet switching easy, though simple passive pop-eliminators (shown later) must be used on the footswitch lines run in a non-shielded 5- or 6- conductor cable. One line carries a +5V control voltage, with no need for a ground wire, while the other lines are the returns in which push-on / push-off footswitches are installed. Builders of this project must design and construct their own footswitch array, but there are many sources for enclosures that can be used for that purpose. Or a suitable replacement 4-switch array made for an existing guitar-amp could be used, though wired according to the needs of this project.
Pinout diagrams for all the ICs in this project are given at the end of this post.
What follows are the sub-circuits designed to be used in the various channels, but which are presented in no mandatory order, since they can be connected in any order within a given channel, and any sub-circuit can even be used independently for some purpose other than including it with the others in this project. In other words, any given output can be used as an effect send, or high-Z output to an amplifier or to recording gear, but which signal need not necessarily be returned.
Shown next is a 3-band tone-control in which each control adjusts the amplitude of an actual band, not just the center of each band. The Bass band has frequencies < 1 kHz, the Mid band is from 500 Hz to 5 kHz, and the Treble band has frequencies > 3 kHz.
It is recommended that four of these circuits are built, using four quad op-amps, in case the user wants one in each channel, but they need not all be used. Below is a gain stage of which four are also recommended, though using only one quad op-amp (with one gain stage for the direct channel).
Here therefore are three different distortion circuits, labeled Fuzz 1, 2, 3, but which can be used in any order, though four of each circuit are not needed. Rather, the purpose is to use one each in a channel of its own while keeping the direct undistorted signal in Channel 1 (though it too should be given a gain stage and 3-band tone-control). Two of the Fuzz circuits use diodes to get distortion while the third uses overdriven LF353 op-amps. The OPA2134 op-amps are here to provide clean amplification, but the LF353, also a dual JFET-input op-amp, is used here because of its inferior noise figures. This op-amp was one of the earliest versions of JFET-input op-amps and should therefore be fairly easy to cause to clip with sufficient input signal voltages.
At this point let me explain the operation of Fuzz Unit 1, shown below. An input level control is provided in case a Gain Stage is used before it in its channel, or if it is to be used independently. Gain is fixed fairly high; around 220. It drives a diode circuit that uses germanium diodes, said to produce a distortion sound similar to that of overdriven preamp tubes. In fact, they would convert a sine-wave into a square-wave, but whose edges are more rounded than what would happen with silicon diodes. LEDs are also said to produce good distortion, and a passive diode distortion circuit using both types has been included with this project.
The only other control for Fuzz Unit 1 is an output level control, which adjusts the fuzz signal’s amplitude going to the input of a buffer, which is there simply to serve as an active circuit placed between the diodes and any given succeeding circuit.
Up next is Fuzz Unit 2, which uses diodes in a feedback loop rather than as a shunt on the signal path. It cannot be predicted as to which method of obtaining distortion will sound better than the other. Only building them and testing them will reveal the answer to that question. Yet, both circuits were based on preamps in proven guitar-amps.
On the other hand, Fuzz Unit 2 has more adjustability than Fuzz Unit 1, sporting not only input and output volume controls but separate “Intensity” and “Fuzz” controls. And below is given Fuzz Unit 3, which has no diodes but relies on overdriven op-amps to get distortion, as said. It has four volume controls, labeled in order Drive, OD1, OD2, and Level. [OD = overdrive.] Drive sets the circuit’s input level, OD1 and OD2 control the interstage signal voltages, and the Level control sets output voltage. However, there is also a Feedback control, employed in case the user wants to tame the distortion down while still getting some distortion from the circuit. It is connected backwards so the circuit will sound as if it is getting louder as the control is turned clockwise.
Here are a couple of passive distortion circuits. If built, for best sound they should be driven by one of the Gain Stages. How they work is that the diodes cause clipping of the input signal, though there is some loss due to the diodes actually constituting shunts to ground -- hence the need for an active device driving each. And the best distortion tone, emphasizing even-order harmonics, is obtained when a given diode array has one diode on one side but two on the other. It does not matter which side has the lone diode since what matters is that there is an imbalance. Of course, any setting except when all diodes are disabled will produce distortion. The user can set switches according to their personal taste. In some cases, for instance, harsher distortion, emphasizing odd-order harmonics, may be desired, instead of the sweetness of even-order harmonics.
I recommend using the 3-Band Tone controls at the end of each channel, between the final sub-circuits and the Mixer, so that the tonality of each channel can be adjusted at the inputs of the Mixer, shown below.
The op-amp circuit in the Mixer is configured as a basic summing amp, where each of its inputs has a separate volume control, but there is only one Mix Volume control on the output. This is meant to be the final output of the project and is suitable for the input of any guitar-amp or any other equipment that can accepts a high-Z guitar signal, including another preamp, a stomp-box effect, PA mixing console, recording gear, etc.
Shown below are the pinout diagrams for the ICs used in this project, along with the pop-eliminator circuit, with four required, and which can be placed within the project enclosure or in the remote footswitch array’s enclosure, as desired, as long as a given pop-eliminator is between its footswitch and the Switching IC. There is also a sketch of how to properly connect each IC to its respective supply-voltage leads.
Here is a block diagram indicating the suggested initial connections scheme, though the user can connect individual subcircuits in any sequence or in any channel.
Last but not least is the power supply. This circuit is of my own design but is based on long-standing power-supply topologies. Major components are specified below.
EOF
This is an experimental design for a solid-state preamp made using op-amps and is intended to result in a guitar preamp that is more versatile than typical. I start with the input stage, which is simply a normal voltage amp using an OPA2134 op-amp, chosen because it is reported to be one of the cleanest op-amps around, having noise figures better than most other op-amps on the market, and which was purpose designed for professional audio applications in which low noise is of primary concern.
The circuit is also a distribution amp, though gain is rather low. Yet, the gain should be sufficient to send the input signal to four separate channels. I have shown it with input and output jacks on purpose. Each sub-circuit of the project is shown this way so as to emphasize the versatility of the project. Making each sub-circuit’s input and output user accessible, say via a patch-bay in the back of the completed project’s enclosure, allows the user to patch the sub-circuits together as preferred, and to treat each in/out interface as an effects loop. An Input Level control has been included to tame large input signals that result from hot pickups and stomp-box effects, if needed. The gain here can be set from unity to 25, and each output has its own volume control.
As can be seen below, each channel is enabled independently of the others. I suggest using Channel 1 as a direct channel, though with a gain stage and active tone-control, so the input signal can keep up with the signals from the other three channels.
There is no digital circuitry in this project, although the switching IC (a CD4066B quad bilateral switch) is often used in digital equipment. Rather, the switching IC is only there to perform as a quiet switch the user employs to enable a given channel.
The genius of this IC in audio applications is that it makes doing quiet switching easy, though simple passive pop-eliminators (shown later) must be used on the footswitch lines run in a non-shielded 5- or 6- conductor cable. One line carries a +5V control voltage, with no need for a ground wire, while the other lines are the returns in which push-on / push-off footswitches are installed. Builders of this project must design and construct their own footswitch array, but there are many sources for enclosures that can be used for that purpose. Or a suitable replacement 4-switch array made for an existing guitar-amp could be used, though wired according to the needs of this project.
Pinout diagrams for all the ICs in this project are given at the end of this post.
What follows are the sub-circuits designed to be used in the various channels, but which are presented in no mandatory order, since they can be connected in any order within a given channel, and any sub-circuit can even be used independently for some purpose other than including it with the others in this project. In other words, any given output can be used as an effect send, or high-Z output to an amplifier or to recording gear, but which signal need not necessarily be returned.
Shown next is a 3-band tone-control in which each control adjusts the amplitude of an actual band, not just the center of each band. The Bass band has frequencies < 1 kHz, the Mid band is from 500 Hz to 5 kHz, and the Treble band has frequencies > 3 kHz.
It is recommended that four of these circuits are built, using four quad op-amps, in case the user wants one in each channel, but they need not all be used. Below is a gain stage of which four are also recommended, though using only one quad op-amp (with one gain stage for the direct channel).
Here therefore are three different distortion circuits, labeled Fuzz 1, 2, 3, but which can be used in any order, though four of each circuit are not needed. Rather, the purpose is to use one each in a channel of its own while keeping the direct undistorted signal in Channel 1 (though it too should be given a gain stage and 3-band tone-control). Two of the Fuzz circuits use diodes to get distortion while the third uses overdriven LF353 op-amps. The OPA2134 op-amps are here to provide clean amplification, but the LF353, also a dual JFET-input op-amp, is used here because of its inferior noise figures. This op-amp was one of the earliest versions of JFET-input op-amps and should therefore be fairly easy to cause to clip with sufficient input signal voltages.
At this point let me explain the operation of Fuzz Unit 1, shown below. An input level control is provided in case a Gain Stage is used before it in its channel, or if it is to be used independently. Gain is fixed fairly high; around 220. It drives a diode circuit that uses germanium diodes, said to produce a distortion sound similar to that of overdriven preamp tubes. In fact, they would convert a sine-wave into a square-wave, but whose edges are more rounded than what would happen with silicon diodes. LEDs are also said to produce good distortion, and a passive diode distortion circuit using both types has been included with this project.
The only other control for Fuzz Unit 1 is an output level control, which adjusts the fuzz signal’s amplitude going to the input of a buffer, which is there simply to serve as an active circuit placed between the diodes and any given succeeding circuit.
Up next is Fuzz Unit 2, which uses diodes in a feedback loop rather than as a shunt on the signal path. It cannot be predicted as to which method of obtaining distortion will sound better than the other. Only building them and testing them will reveal the answer to that question. Yet, both circuits were based on preamps in proven guitar-amps.
On the other hand, Fuzz Unit 2 has more adjustability than Fuzz Unit 1, sporting not only input and output volume controls but separate “Intensity” and “Fuzz” controls. And below is given Fuzz Unit 3, which has no diodes but relies on overdriven op-amps to get distortion, as said. It has four volume controls, labeled in order Drive, OD1, OD2, and Level. [OD = overdrive.] Drive sets the circuit’s input level, OD1 and OD2 control the interstage signal voltages, and the Level control sets output voltage. However, there is also a Feedback control, employed in case the user wants to tame the distortion down while still getting some distortion from the circuit. It is connected backwards so the circuit will sound as if it is getting louder as the control is turned clockwise.
Here are a couple of passive distortion circuits. If built, for best sound they should be driven by one of the Gain Stages. How they work is that the diodes cause clipping of the input signal, though there is some loss due to the diodes actually constituting shunts to ground -- hence the need for an active device driving each. And the best distortion tone, emphasizing even-order harmonics, is obtained when a given diode array has one diode on one side but two on the other. It does not matter which side has the lone diode since what matters is that there is an imbalance. Of course, any setting except when all diodes are disabled will produce distortion. The user can set switches according to their personal taste. In some cases, for instance, harsher distortion, emphasizing odd-order harmonics, may be desired, instead of the sweetness of even-order harmonics.
I recommend using the 3-Band Tone controls at the end of each channel, between the final sub-circuits and the Mixer, so that the tonality of each channel can be adjusted at the inputs of the Mixer, shown below.
The op-amp circuit in the Mixer is configured as a basic summing amp, where each of its inputs has a separate volume control, but there is only one Mix Volume control on the output. This is meant to be the final output of the project and is suitable for the input of any guitar-amp or any other equipment that can accepts a high-Z guitar signal, including another preamp, a stomp-box effect, PA mixing console, recording gear, etc.
Shown below are the pinout diagrams for the ICs used in this project, along with the pop-eliminator circuit, with four required, and which can be placed within the project enclosure or in the remote footswitch array’s enclosure, as desired, as long as a given pop-eliminator is between its footswitch and the Switching IC. There is also a sketch of how to properly connect each IC to its respective supply-voltage leads.
Here is a block diagram indicating the suggested initial connections scheme, though the user can connect individual subcircuits in any sequence or in any channel.
Last but not least is the power supply. This circuit is of my own design but is based on long-standing power-supply topologies. Major components are specified below.
EOF
Curiosity: Tubes for nothing?
- By snapper
- Tubes / Valves
- 7 Replies
https://www.ebay.de/itm/275721584797
The Chinese are selling silicon amplifiers with 2x 25W,
Tube types: 2x 6N1 & 2x 6P1 with "primary optical function"[/B][/SIZE]
As if the tubes are just for show !
Who would fall for that? (47 Sold)
Do she install her scrap Tubes there?
The Chinese are selling silicon amplifiers with 2x 25W,
Tube types: 2x 6N1 & 2x 6P1 with "primary optical function"[/B][/SIZE]
As if the tubes are just for show !
Who would fall for that? (47 Sold)
Do she install her scrap Tubes there?
Apex A40 Amplifier Build Thread
- By HatlessChimp
- Solid State
- 120 Replies
This is a dedicated thread to the Apex A40 Amplifier. Feel free to share your build and discuss and share information about the A40.
I will do my best to keep this 1st post updated with as much information as possible. I am still building this out, please be patient. If there are any errors please let me know! 🙂
The Project
The Apex A40 Amplifier was designed by Apex Audio and kindly shared to DiyAudio community. He has continually shared various designs, revisions and countless feedback over many years. All of his designs are popular and well endorsed by many experienced members in the DIYA Community. The original thread can be found here - 100W Ultimate Fidelity Amplifier - diyAudio
Schematic
Apex A40 Files
Gerbers, Mirrored PCB images, Photos, Schematic and Partial .txt BOM can be gound here - https://www.diyaudio.com/forums/solid-state/338942-apex-a40-amplifier-build-thread.html#post5817489
Bill of materials
Anyone with an up to date BOM for Digikey please contact me
Transistor Choices
2SA1943 – Toshiba, On Semiconductor - Transistors - Bipolar (BJT) - Single | Discrete Semiconductor Products | DigiKey
2SC5200 – Toshiba, On Semiconductor - Transistors - Bipolar (BJT) - Single | Discrete Semiconductor Products | DigiKey
MJL4281 - On Semiconductor - Transistors - Bipolar (BJT) - Single | Discrete Semiconductor Products | DigiKey
MJL4302 - On Semiconductor - Transistors - Bipolar (BJT) - Single | Discrete Semiconductor Products | DigiKey
NJW0281GOS-ND - On Semiconductor - https://www.digikey.in/products/en?keywords=NJW0281
NJW0302GOS-ND - On Semiconductor - https://www.digikey.in/products/en?keywords=NJW0302
Are my transistors a fake/copy thread - My Transistors, original or copy?
Power supplies
Construction tips
Please contact me with any tips you have to share.
Project modifications
Know of any proven mods? Please share them!
Recommended DIY Pre Amp
Apex P30
A40 Members and build photos
Please Share your builds!
JUST WAITING ON MODERATOR APPROVAL FOR UPLOADING MORE FILES TO THIS THREAD/POST.
Cheers!
I will do my best to keep this 1st post updated with as much information as possible. I am still building this out, please be patient. If there are any errors please let me know! 🙂
The Project
The Apex A40 Amplifier was designed by Apex Audio and kindly shared to DiyAudio community. He has continually shared various designs, revisions and countless feedback over many years. All of his designs are popular and well endorsed by many experienced members in the DIYA Community. The original thread can be found here - 100W Ultimate Fidelity Amplifier - diyAudio
Schematic
Apex A40 Files
Gerbers, Mirrored PCB images, Photos, Schematic and Partial .txt BOM can be gound here - https://www.diyaudio.com/forums/solid-state/338942-apex-a40-amplifier-build-thread.html#post5817489
Bill of materials
Anyone with an up to date BOM for Digikey please contact me
Transistor Choices
2SA1943 – Toshiba, On Semiconductor - Transistors - Bipolar (BJT) - Single | Discrete Semiconductor Products | DigiKey
2SC5200 – Toshiba, On Semiconductor - Transistors - Bipolar (BJT) - Single | Discrete Semiconductor Products | DigiKey
MJL4281 - On Semiconductor - Transistors - Bipolar (BJT) - Single | Discrete Semiconductor Products | DigiKey
MJL4302 - On Semiconductor - Transistors - Bipolar (BJT) - Single | Discrete Semiconductor Products | DigiKey
NJW0281GOS-ND - On Semiconductor - https://www.digikey.in/products/en?keywords=NJW0281
NJW0302GOS-ND - On Semiconductor - https://www.digikey.in/products/en?keywords=NJW0302
Are my transistors a fake/copy thread - My Transistors, original or copy?
Power supplies
Construction tips
Please contact me with any tips you have to share.
Project modifications
Know of any proven mods? Please share them!
Recommended DIY Pre Amp
Apex P30
A40 Members and build photos
Please Share your builds!
JUST WAITING ON MODERATOR APPROVAL FOR UPLOADING MORE FILES TO THIS THREAD/POST.
Cheers!
NAD 7220PE still blowing outputs
- By bamavette
- Solid State
- 14 Replies
I am working on this for a friend, the left channel was out. I went through the amp and replaced the outputs/drivers and a couple of resistors, on power up the right channel was pulling too much current. I went through the amp again, the soft clipping and power supply. In the power supply one diode was bad D510, which I replaced.
I powered up with a DBT and everything looked good, I made sure the idle current was low enough so it wouldn't blow without the DBT.
I powered up without the DBT, voltages looked good, Per the service manual I adjusted the DC offset to 4 or 5mv, adjusted the idle current to 27-28mv (spec is 26-30mv). I replaced the solder bridge and powered it up. A sine wave looked OK, square wave showed some ringing (8ohm speaker load).
I hooked up a CD player and let it play at low volume for about 10 minutes and all was well. I slowly turned the volume up and at about 9:30-10 oclock on the potentiometer the left channel blew the outputs, some drivers and R454.
Obviously I am missing something, any ideas?
This is the amp/power supply schematic
This is a closer view with the blown components marked
I powered up with a DBT and everything looked good, I made sure the idle current was low enough so it wouldn't blow without the DBT.
I powered up without the DBT, voltages looked good, Per the service manual I adjusted the DC offset to 4 or 5mv, adjusted the idle current to 27-28mv (spec is 26-30mv). I replaced the solder bridge and powered it up. A sine wave looked OK, square wave showed some ringing (8ohm speaker load).
I hooked up a CD player and let it play at low volume for about 10 minutes and all was well. I slowly turned the volume up and at about 9:30-10 oclock on the potentiometer the left channel blew the outputs, some drivers and R454.
Obviously I am missing something, any ideas?
This is the amp/power supply schematic
This is a closer view with the blown components marked
Japanese 2A3 schematic opinion wanted
- By Neiljd990
- Tubes / Valves
- 13 Replies
I'm a noob so I'm looking for an opinion of this schematic. There is an amp for sale locally to me that uses it. I've reversed imaged searched it and it appears to be only floating around Japanese and Chinese websites. I'll be looking to upgrade the transformers and other parts but just wanted to know if it would be worth getting first. I'll maybe convert it to a 45 too down the line.
Cheers!
Cheers!
Biasing B&K amp with 2SK1058/2SJ162
- By dbxdx5
- Solid State
- 1 Replies
I have a B&K AVR 507 Series 2. At idle, the heat sink on one side is running about 10 degrees hotter than the other. What should I be aiming for in terms of bias per channel?
Opinions on Vandersteen's posts and canopy
This is a random pic of a pair of Vandersteen, model 2, from the web. Of course, Vandersteen is iconic and earned it. We are not challenging the design, just curious: What's the purpose of the posts, canopy, and sleeves? Aesthetic or acoustic? What's your take? Also, what's you opinions on these speakers?
3e Audio TPA3250D2 (SY-DAP1002) Mods & DSP Tuning
I came onto the Class D bandwagon quite late with BRZ HiFi TPA3255 amp last year. Since then, I found diyaudio and I am grateful that I have learned a lot from many posters here.
To me, the attractiveness of 3e Audio SY-DAP1002 is in its choice of TPA3250 over TPA3255, and the inclusion of ADAU1701 programmable DSP chip. The both points are not at all widely accepted as a virtue, and are subject to debate. I will go into each, but in short, this 3250 sounds better than most 3255 for me.
Here's the initial review to start with:
The 3eAudio SY-DAP1002 arrived on March 9th here in Tokyo where I live. It came with Rev. 1.1 pcb, which means it was in a production batch after the initial 100 Rev. 1.0 boards with OPA1602 OpAmp were sold. Rev. 1.1 comes with DIP8 OpAmp sockets in place of the soldered SOIC OPA1602, and through-hole Elna Silmic II's replacing various through-hole and SMD capacitors.
As you can see, 3 big through-hole PS caps at C134, C150 and C151 came unpopulated, and what you might not see is the four "Background noise reduction filter" positions came fully populated with SMD chips. I was not sure if I'd get the Rev. 1.0 when I placed the order, and was ready to do this mod myself, so it was a nice surprise to find the amp totally silent at full volume with no signal from the listening position. The noise, the level of which does not change with volume position, is ever slightly audible with my ear at less than 1" away from a tweeter.
(Please ignore the CoilCraft info, that has nothing to do with DAP1002 layout.)
When compared to BRZ HiFi TPA3255 (unmodified, but with MUSES8920), the sound without any modification or aging was mediocre at best. There was a slight advantage or emphasis on mid range 'energy', but otherwise the BRZ HiFi with the sweet OpAmp outperformed it in almost every way.
3e Audio has not published the Sigma Studio .dspproj file yet for making the SY-DAP1002 volume control logarithmic instead of straight line increase, but one noticeable characteristic was the high level of gain at the lowest volume position, giving the impression like going from 3250 to the higher power 3255, when in reality I was used to 3255 of the BRZ to try the lower power 3250 of this 3e Audio. This later led me to believe the OpAmp is not used with unity gain, used probably at a gain of 2, just like in Aiyima TPA3251. This likely makes the amp more accommodating to exotic OpAmps that are not designed for audio which are more susceptible to oscillation, and is a huge designed-in advantage these two amps have over many others in 1. Wider OpAmp rolling choices, and 2. A lighter gain demand on the TI TPA chips at any given sound pressure level (SPL), in my opinion.
Anyway, the culprit was found as soon as I tried OPA1656 on this amp:
SY-DAP1002 Rev. 1.1 came with this SOIC 'N5532' on a SOIC to DIP conversion board.
Just by swapping the OpAmp to OPA1656, this 3e Audio amp outperformed the BRZ HiFi 3255 with MUSES8920 in almost every aspect of the sound quality. Probably the 'N5532' is a fake, but who would make a fake 'SOIC' NE5532 (produced mostly in DIP8 package in the Millions) for what purpose?? It really is a strange world.
Later I compared the two amps both with OPA1656, and the result became closer but did not change in the end. The largest difference was the midrange energy (or slightly more level of midrange 'punch' 3e Audio has) that made the music more involving and enjoyable. With the OPA1656, the both amps have excellent highs and resolution without being harsh, sterile or too analytical.
The unique parts of SY-DAP1002 that are shared with its TPA3255 version, SY-DAP2002. Top left is the STM32 32bit micro CPU by ST Micro that interprets the push button volume control knob switch (Power on = wake up from standby, power off = not a complete shut down but standby, bluetooth link-up, etc.), controls the LED ring around the knob, and more importantly, it detects if an established Bluetooth connection to a host is currently sending an audio signal. When the bluetooth link is idle for a while, it redirects the input from Bluetooth to RCA. Unlike many other amps with a bluetooth connectivity like the BRZ, once-connected smart phones do not need to switch off the bluetooth for the amp to restart accepting input from the RCA connector because of this. Lower left is Qualcomm QCC3008 chip for Bluetooth 5.0 with AptX, SBC and AAC codecs. With a long-ish antenna that comes with the amp, the bluetooth connectivity is quite good.
On the middle right is the ADAU1701 programmable DSP. The whitish square below it is the crystal, and the 8pin SOIC to the left of the crystal is FT24C64A EEPROM for AU1701 (STM32 has its program/data storage EEPROM built in). ADAU1701 and the EEPROM are accessible via 10pin USBi interface connector at the front of pcb, that requires a USBi programmer and an USB cable to a PC that runs Analog Devices' Sigma Studio DSP development environment (IDE) that can be downloaded for free.
I will be going into my background, the mods and eventually into the dsp programming, so stay tuned.
To me, the attractiveness of 3e Audio SY-DAP1002 is in its choice of TPA3250 over TPA3255, and the inclusion of ADAU1701 programmable DSP chip. The both points are not at all widely accepted as a virtue, and are subject to debate. I will go into each, but in short, this 3250 sounds better than most 3255 for me.
Here's the initial review to start with:
The 3eAudio SY-DAP1002 arrived on March 9th here in Tokyo where I live. It came with Rev. 1.1 pcb, which means it was in a production batch after the initial 100 Rev. 1.0 boards with OPA1602 OpAmp were sold. Rev. 1.1 comes with DIP8 OpAmp sockets in place of the soldered SOIC OPA1602, and through-hole Elna Silmic II's replacing various through-hole and SMD capacitors.
As you can see, 3 big through-hole PS caps at C134, C150 and C151 came unpopulated, and what you might not see is the four "Background noise reduction filter" positions came fully populated with SMD chips. I was not sure if I'd get the Rev. 1.0 when I placed the order, and was ready to do this mod myself, so it was a nice surprise to find the amp totally silent at full volume with no signal from the listening position. The noise, the level of which does not change with volume position, is ever slightly audible with my ear at less than 1" away from a tweeter.
(Please ignore the CoilCraft info, that has nothing to do with DAP1002 layout.)
When compared to BRZ HiFi TPA3255 (unmodified, but with MUSES8920), the sound without any modification or aging was mediocre at best. There was a slight advantage or emphasis on mid range 'energy', but otherwise the BRZ HiFi with the sweet OpAmp outperformed it in almost every way.
3e Audio has not published the Sigma Studio .dspproj file yet for making the SY-DAP1002 volume control logarithmic instead of straight line increase, but one noticeable characteristic was the high level of gain at the lowest volume position, giving the impression like going from 3250 to the higher power 3255, when in reality I was used to 3255 of the BRZ to try the lower power 3250 of this 3e Audio. This later led me to believe the OpAmp is not used with unity gain, used probably at a gain of 2, just like in Aiyima TPA3251. This likely makes the amp more accommodating to exotic OpAmps that are not designed for audio which are more susceptible to oscillation, and is a huge designed-in advantage these two amps have over many others in 1. Wider OpAmp rolling choices, and 2. A lighter gain demand on the TI TPA chips at any given sound pressure level (SPL), in my opinion.
Anyway, the culprit was found as soon as I tried OPA1656 on this amp:
SY-DAP1002 Rev. 1.1 came with this SOIC 'N5532' on a SOIC to DIP conversion board.
Just by swapping the OpAmp to OPA1656, this 3e Audio amp outperformed the BRZ HiFi 3255 with MUSES8920 in almost every aspect of the sound quality. Probably the 'N5532' is a fake, but who would make a fake 'SOIC' NE5532 (produced mostly in DIP8 package in the Millions) for what purpose?? It really is a strange world.
Later I compared the two amps both with OPA1656, and the result became closer but did not change in the end. The largest difference was the midrange energy (or slightly more level of midrange 'punch' 3e Audio has) that made the music more involving and enjoyable. With the OPA1656, the both amps have excellent highs and resolution without being harsh, sterile or too analytical.
The unique parts of SY-DAP1002 that are shared with its TPA3255 version, SY-DAP2002. Top left is the STM32 32bit micro CPU by ST Micro that interprets the push button volume control knob switch (Power on = wake up from standby, power off = not a complete shut down but standby, bluetooth link-up, etc.), controls the LED ring around the knob, and more importantly, it detects if an established Bluetooth connection to a host is currently sending an audio signal. When the bluetooth link is idle for a while, it redirects the input from Bluetooth to RCA. Unlike many other amps with a bluetooth connectivity like the BRZ, once-connected smart phones do not need to switch off the bluetooth for the amp to restart accepting input from the RCA connector because of this. Lower left is Qualcomm QCC3008 chip for Bluetooth 5.0 with AptX, SBC and AAC codecs. With a long-ish antenna that comes with the amp, the bluetooth connectivity is quite good.
On the middle right is the ADAU1701 programmable DSP. The whitish square below it is the crystal, and the 8pin SOIC to the left of the crystal is FT24C64A EEPROM for AU1701 (STM32 has its program/data storage EEPROM built in). ADAU1701 and the EEPROM are accessible via 10pin USBi interface connector at the front of pcb, that requires a USBi programmer and an USB cable to a PC that runs Analog Devices' Sigma Studio DSP development environment (IDE) that can be downloaded for free.
I will be going into my background, the mods and eventually into the dsp programming, so stay tuned.
Q17 Sigma22 Regulated Ultrastable Ultralow Noise Linear Power Supply +-50/60Vcc
- By stef1777
- Power Supplies
- 99 Replies
