Need help with a low pass active filter

So as the title says i need a little bit of help to create this filter, i was able to calculate everything for my desired cutoff (80hz) and was able to build everything it works but there is a+
strange behaviour:
Screenshot (109).png

this is the circuit i followed (did twice having 2 op-amps in a single one) i am using a KIA 4558P but following the circuit for the gain feedback circuit (r1-r2) it doesnt filter anymore it doesnt even work if i dont want to use the gain and just wire directly the output to the inverting input it still doesnt work, but the moment i connect the output to the non-inverting input (+) it does work but now i dont have gain and loose usable power of the sub.


sorry for bad english not my main language, sorry for wrong words/terms eletrical engeenering is not my main subject is just a hobby. Thanks in advance

Hello my name is Brian Hernandez I live in Brea Orange County California need help and I will pay it forward I’m a certified hvac Tech but work on all

I have a T1500-1bdcp I got from my cousin for free I contacted it and it went to protected mode opened it and picture and video will show what I found can anyone help me with exact parts to purchase and where to purchase them please

Attachments

  • IMG_2872.jpeg
    IMG_2872.jpeg
    795.2 KB · Views: 84
  • IMG_2871.jpeg
    IMG_2871.jpeg
    740.9 KB · Views: 85

rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool

as rePhase has evolved a lot since day one of this long thread, it is certainly not a good starting point to understand what rePhase does and what its current state of development is. It is nonetheless a good place to ask questions, report bugs, or request features.

This first post is kept up-to-date with links and a change log for the current version of rePhase.

Here is the rePhase project page from where the last version can be downloaded.
It also list a few interesting tutorials.

enjoy 🙂

original post:

Here is a little piece of software I have been working on for some months now.
It is called rePhase, and is a tool for loudspeaker phase linearization, EQ and FIR filtering.
It does so by producing correction impulse responses that you can then use in your favorite convolution engine.

It is a free software, without restriction (but without warranty either).

Download: rePhase project page

At start the first goal of rePhase was to let you generate impulses specifically tailored to reverse the phase shifts introduced by the crossovers of your loudspeakers (passive or active) and boxes, resulting in a linear-phase system.
It is similar to phase arbitrator in this regard (but it is not a processing plugin, just an impulse generator).

rePhase has evolved and now includes other features: it can generate linear-phase EQ and crossover filters of arbitrary slopes, including Linkwitz-Riley (albeit linear-phase) and Horbach-Keele shapes.

With the paragraphic EQ sections you can alter phase and amplitude independently.

You will need a convolution engine to run the generated impulse, such as the minidsp openDRC, foobar2000 convolver or VST convolver, among many other software and hardware tools.

You can use rePhase to linearize passive loudspeakers, active ones, and turn IIR active crossovers into linear-phase crossovers (turn a DCX2496 into a DEQX or a Dolby Lake 😉 ).
You can also design the whole filter with rePhase.
In any case you will need one convolution per way (Jriver can be used to run the convolutions for example)

Features include:
  • generate impulse responses (FIR) for convolution engines
  • measurement import and real-time correction
  • loudspeakers phase linearization (passive or active crossovers)
  • linear-phase and minimum-phase gain paragraphic EQ and shelving
  • constant-gain phase praragraphic EQ
  • multiple gain EQ algorithms (constant Q, proportional Q, constant shape, raised cosine)
  • arbitrary slopes linear-phase filters (Linkwitz-Riley, Brickwall, Horbach-Keele, etc.)
  • arbitrary slopes minimum-phase filters (Linkwitz-Riley, Butterworth, etc.)
  • real-time graphical monitoring of target and results curves
  • automatic optimization for the best result with a given number of taps
  • multiple windowing choices

Sorry for the lack of documentation.
Questions, remarks and suggestions are welcome!

Changelog:

Code:
1.4.3 2019-01-16
  Bug corrections:
    - after loading a preset, correctly show "rotate" option when set in
      Filters Linearization tab
    - resolved the zoom out (right click) bug where too much zoom states
      were added
  Adjustments:
    - measurement interpolation is now logarithmic in both magnitude and
      frequency axes, so that an interpolation between two points will
      always show as a straight line

1.4.2 2018-12-27
  Bug corrections:
    - corrected graphical EQ manipulation behavior when gain offset is used
      on a measurement
    - corrected active EQ focus bug after loading a measurement

1.4.1 2018-12-23
  Adjustments:
    - improved compatibility of frd format:
        * enforce decimal form instead of scientific notation
        * use semicolon instead of tabulation as column separator
        * add commented info (software, url, columns description)
    - improved information in EQ bank drop-down menu, including EQ type and
      number of bypassed EQs
    - dynamically adjusted choices in FFT drop-down menu, removing unusable
      options
    - removed unused EQ type drop-down menus in paragraphic phase EQ tab
    - reworked links and contact info

1.4.0 2018-12-19
  New features:
    - graphical zoom functionality, similar to HOLMImpulse:
        * clicking and dragging the mouse over the response graph draws a
          zoom box that defines the new graph area when button is released
        * right clicking cancels last zoom operation
        * zoom only affects frequency and magnitude scales, not phase one
    - real-time graphical edition of gain EQ points:
      clicking an EQ fader or entries will turn them yellow to reflect the
      fact that this particular EQ point now has a special focus:
        * mouse wheel changes its Q value while on the response graph
        * middle click or ctrl-click on the response graph updates its
          freq and dB values in real-time while the button is held pressed
        * dB position is relative to the existing target magnitude curve,
          including optional measurement, and is limited to ±96dB per EQ
          (note: it is recommanded to use "constant shape" EQ type for
          high amplitude corrections)
        * modifications can be cancelled as long as focus is not lost
    - "frequency response (.frd)" format to export the generated correction
      (ie the red curves) as a three columns frequency/magnitude/phase file
    - Added "linearize"/"rotate" option in Filters Linearization tab.
      "linearize" is the default and compensates for phase rotation of a
      given filter (inverse all-pass), whereas "rotate" emulates the phase
      rotation of the chosen filter without affecting magnitude (all-pass).
    - New "throughout banks" EQ tools to bypass or order EQ points
      throughout all banks at once. Confirmation is requested as this can
      be an irreversible operation. Ordering EQ points between different
      banks requires EQ types to be identical in all banks.
    - try to let the user save current settings before exiting in case of a
      crash
  Adjustments:
    - frequency marker on the magnitude target curve is replaced with a
      vertical yellow line that reflects both magnitude and phase
      corrections
    - up/down key binding on drop-down menus to iterate values (same idea
      as existing incrementation/decrementation of entries with numerical
      values)
    - link to rephase.org in Help menu
    - stay in same tab after settings load/reset
    - more compact setting file
    - view preferences are saved on the fly instead of when quitting
    - force entry focus loss when switching tab
    - added Nyquist frequencies of a few common sampling rates in frequency
      upper limit choices in the Range tab
    - removed bypassed EQs from EQ points count in "Bank" drop-down menu
    - avoid saving measurement summary in the setting file, and generate it
      on the fly

1.3.0 2018-02-15
  New features:
    - "Load Recent Settings" entry in File menu, keeping track of the last
      15 opened or saved setting files
    - ctrl-click on a fader will reset its value to zero, similar to what a
      middle click does (useful for persons using a mouse pad)
    - improved measurement parsing heuristics to handle more formats;
      recommended measurement format is still the one described in the info
      box when failing to load a measurement
    - added CSV output format: 64 bit floating-point values in text format,
      separated with commas
    - allowed measurement importing from clipboard, both from File menu
      and Measurement tab
    - added "hide magnitude" button in Measurement tab
  Adjustments:
    - added a "Donate" entry in Help menu linking to a paypal donation page
    - made "Load Settings From Clipboard" functionality tolerant to leading
      and trailing spaces and newlines in clipboard
    - reworked output format names
    - renamed "invert" button to "invert response" for clarity in
      Measurement tab
    - rename "impulse offset" to "impulse delay" for clarity in impulse
      status report
    - add max impulse level in impulse status report, complementing max
      response level
    - reworked impulse and measurement status report areas to make them
      more visible
    - enabled DPI adaptation if forced to by the operating system (not
      recommended, looks nice but crashes might occur), and added scale
      ratio and DPI indication in View menu
    - added a no warranty disclaimer in "About" info box

1.2.0 2016-12-08
  New features:
    - REW automated EQ settings generated using the 'rePhase' equaliser
      type (as implemented in REW V5.17 beta 14 and up) can now be imported
      directly into a paragraphic EQ bank
    - EQ points in paragraphic EQ tabs can now be individually bypassed
    - added a "tools" menu in paragraphic EQ tabs, effective on current
      bank:
      * load/save current bank into a '.eq' file as a JSON object
        (gain paragraphic EQ tab only)
      * load/save current bank into the clipboard as a JSON object to
        easily copy it to other banks or rePhase instances, or share it
        through forum posts
        (gain paragraphic EQ tab only)
      * import REW EQ settings generated with 'rePhase' equaliser type
        (gain paragraphic EQ tab only)
      * convert back and forth between constant and proportional Q types
        (gain paragraphic EQ tab only)
      * invert corrections
      * bypass or activate all EQ points
      * order by frequency, active or reversed order
    - "Help" menu entry (albeit probably not very helpful :( )
  Bug corrections:
    - The long lasting encoding issues with paths when loading, saving, and
      generating files should now at last be solved. It was already
      supposed to be the case in version 0.9.7, then 1.1.0, and should now
      *at last* be effective. Please report any problem with files or paths
      containing special characters (accents, etc.).
    - Corrected a bug introduced in version 1.1.1: fader position could
      sometimes change based on the position of the mouse cursor after
      loading or saving a file
    - Stop confining mouse cursor within faders, as it could stay stuck
      under some rare circumstances
  Adjustments:
    - set default optimization setting to none, as optimization process can
      increase preringing and should only really be used when the number of
      available taps is too limited to obtain the desired magnitude curve
    - increased default number of taps to 16384 to reflect an increase in
      CPU and DSP power in the last few years (wishful thinking? :) )
    - boost FFT length calculation ratio to improve precision
    - changed default windowing algorithm from rectangular to hann for a
      more generic default behavior
    - EPS vector files screenshots including result curves are now
      significantly lighter and result in smoother curves compared to
      versions 1.1.0 and 1.1.1
    - suppressed flickering when switching between Views buttons
    - default to "Large" view mode

1.1.1 2016-10-29
  New features:
    - added a 64 bit IEEE-754 output format to accommodate BruteFIR:
      [url=http://tinyurl.com/htqvln8]rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 124 - diyAudio[/url]
  Bug corrections:
    - fader focus bug solved: [url=http://tinyurl.com/ztbp7cl]rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 123 - diyAudio[/url]
    - beefed-up clipboard handling to avoid bugs when loading measurements
      by dragging them over the interface: [url=http://tinyurl.com/jygt7jx]rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 125 - diyAudio[/url]
    - removed the final optimization step that was recently added in
      version 1.1.0 as it had some ill side effects in specific scenarios
  Adjustments:
    - renamed measurement "compensate" function to "invert" to avoid
      confusions: [url]http://tinyurl.com/j76wm5w[/url]
    - reworked focus and entries editing in paragraphic EQ tabs:
        * using the tab key now goes from one entry to the other as
          expected, making it faster to edit multiple EQ points in a row
        * clicking on an entry does select the entire content for ease of
          editing
        * middle click or double left click on the dB/° entry used to reset
          the value to 0. This is now gone, but the same result can still
          be achieved with a middle click on the fader
    - internal DPI scaling adjustments

1.1.0 2016-10-27
  New features:
    - screenshot functionalities added to the file menu:
        * "Save Graph Screenshot As..." saves current graph view with a
          logo added to the bottom right corner
        * "Save Window Screenshot As..." saves current window view with the
          directory entry hidden for anonymity concerns
        * supported bitmap file formats are PNG, GIF and JPEG
        * graph screenshot also supports EPS vector file format
    - 64 bits output formats handling (mono/stereo IEEE wav, txt)
    - All-Pass filters added to the Minimum-Phase Filters tab
    - measurement compensate mode to manually replicate a given measurement
      (eg replicating a mic calibration file and getting its missing
      minimum-phase response, replicating a target curve, etc.)
    - added a "Clear Result" entry in the File menu, clearing result curves
      and status
    - reworked "what's new" changelog window to make it easier to read
  Removed features:
    - removed "complex" windowing algorithm which was deprecated since
      version 0.9.0
      An alert box will popup when loading settings using it
  Bug corrections:
    - bug correction for 2nd order minimum-phase filters with Q<0.5
      An alert box will popup when loading settings containing that bug
        * bug report: [url]http://tinyurl.com/z59qw4t[/url]
    - filename encoding bugfix: accents and other special characters
      handling in setting, impulse, and measurement filenames and paths
      (was supposed to be resolved since 0.9.7, but was not...)
    - last updated graph setting sometimes remained unchanged after
      resetting or loading new settings
    - fixed a few instabilities:
        * Horbach-Keele filters with R=1
        * clipboard corner case errors
        * NaN detection bugs
    - fixed various graphical bugs:
        * generation status remaining after reset
        * misbehaved result phase curve when hide=-Inf
        * view slots flickering in large view mode
  Adjustments:
    - faster constant Q, proportional Q and constant shape EQs calculation:
      should be around three times as fast now, and should be notable when
      manipulating EQ faders as well as during the first generation step
      when a lot of EQ points are used
    - added "Pano Phase Shuffler" presets in the Paragraphic Phase EQ tab
        * source: [url]http://tinyurl.com/pano-shuffler[/url]
        * settings: [url]http://tinyurl.com/pano-shuffler-preset[/url]
    - status text under the generation button 
    - added a final optimization step in moderate/extensive/maximal modes
      with a correction factor set to 1
    - added ± 15dB and ± 18dB ranges for convenience in paragraphic EQ tab
    - increased EQ dB precision from 0.1dB to 0.01dB for manual editing
    - set default optimization floor to -100dB
    - faster startup time
    - made clear the fact the subsonic filters linarization options were
      based on optimized approximations (cf [url]http://tinyurl.com/hslsb8m[/url] ) by
      by naming them as such and adding "textbook" versions for the most
      reckless users :) 

1.0.0 2015-06-25
  New features:
    - Albrecht cosine windows implementation
      Ref: A Family of Cosine-Sum Windows for High-Resolution Measurements
    - multiple memory slots in range settings to be able to quickly go from
      one view to the other and focus on different aspects of the response
      curves
      These slots are preset with (hopefully) useful values but can be
      manually modified and copied.
    - "Load Settings From Clipboard" and "Save Settings To Clipboard" menu
      entries in order to be able to easily share corrections on web forums
    - frequency marker for the last correction point (5 sec persistence)
    - fader values can now be manually edited to arbitrary values
  Bug corrections:
    - bug correction in Minimum-Phase Filters tab: the polarity of low-pass
      Linkwitz-Riley filters of order 2(2n+1) was reversed
      (eg 12dB/oct, 36dB/oct, 60dB/oct, 84dB/oct, etc.)
      A warning will be emitted when loading correction files from prior
      versions using an odd number of such filters, as the polarity will
      now be correct and reversed compared to the prior bogus correction.
    - bug correction with higher than normal noise floor with even order
      taps (introduced in version 0.9.9 while solving a similar problem
      for odd taps numbers!)
    - bug correction with txt output file with 0.000(...)0 values
      (especially pregnant when using Hann window)
    - correction of the bogus flat top window implementation
    - corner case instabilities corrections (undue octal conversions on
      some value entries)
  Adjustments:
    - set "32 float txt" as the default output format instead of "32bit
      LPCM wav" in order to avoid  rising the result noise floor because of
      the fixed point format
    - added de-empahasis and pre-emphasis presets in the Paragraphic EQ tab
    - added Linkwitz-Riley linearization orders 11th to 16th (why not?) 
    - reduce default phase EQ range to ± 45° (was ± 90°) and removed
      unpractical ranges
    - increase default EQ range to ± 12dB (was ± 6dB)
    - added 384 and 352.8kHz sampling rates as drop menu options for ease
      of use (any other value can still be manually entered)
    - got rid of the "Curves" tab for the time being, waiting for the
      capture functionality to be implemented in some future version...

0.9.9 2014-12-10
    - shelving EQs with variable Q in Paragraphic EQ tab, with associated
      monotonic high and low shelv presets
    - centering can now be manually set to values in samples, percentage,
      time (us/ms/s) and distance (mm/cm/m).
      It is also possible to add or subtract several values, for example
      "middle+270us"
    - new centering adjustment layout:
        * 'float' is now 'use closest perfect impulse' and is explicitly
          recommended
        * 'int' is now 'round to closest sample'
        * 'use exact centering value' has been added for exact delays
    - 32 bits IEEE-754 float WAV output format added
    - output format noise floor is now shown in result curve
    - improved impulse and windowing symmetry, especially when an odd taps
      value is used
    - import/clear measurement file menu entries
    - fix partial installation catch

0.9.8c 2014-09-29
    - correction of a bug introduced in version 0.9.8 for closed-box phase
      linearization

0.9.8 2014-09-28
    - minimum-phase filters tab with common IIR filter types:
        * 1st order
        * 2nd order with arbitrary Q
        * Butterworth with slopes ranging from 6dB/oct to 996dB/oct in 6dB
          increments
        * Linkwitz-Riley with slopes ranging from 12dB/oct to 996dB/oct in
          12dB increments
    - 'compensate' mode for generalized arbitrary order Linkwitz
      Transform-like manipulations in minimum-phase filters tab
    - new centering options expressed as a percentage to easily obtain
      matched delays
    - default to "middle" centering instead of "energy" to avoid delay
      mismatch problems for the unaware user (principle of least surprise)
    - praxis measurement format handling, scientific notation in frequency
      column
    - smaller executable, new installation method
    - bug correction: crash on impulse generation with some specific filter
      settings
    - directory handling bug correction
    - measurements can now be loaded from the command line or drop on exe,
      similarly to settings
    - revamped file extension handling (settings)
    - revamped icon
    - smoothed out taps/fft size calculation

0.9.7 2013-09-03
    - Brickwall filters implementation.
      /!\ result slope relies solely on windowing /!\
      Iterative optimization and energy centering algorithms are
      automatically defeated when a brickwall filter is set, to make it
      possible to build complementary crossovers. It is up to the user to
      make sure he uses the exact same number of taps and same windowing
      algorithm on both sides of the crossover to ensure complementarity
    - sampling rate drop down menu can now also be directly edited to input
      arbitrary values, so menu options have been reduced to the most
      common values for clarity and ease of use
    - frequency, amplitude and phase ranges can now also be set to
      arbitrary values
    - optimization floor can now be set (was -40dB fixed)
    - B-weighting in optimization calculation was removed (for now)
    - new amplitude paragraphic presets with fixed frequencies (1kHz) for
      various Q values (0.5, 1, 2, 4, 8, 16)
    - got rid of scientific notation in txt output format to broaden
      compatibility
    - dark graph theme
    - various graphical bugs resolution
    - C float array output formats
    - filename encoding bugfix: accents and other special characters
      handling in setting and measurement filenames
    - "reject" filter slopes bugfix
    - "Save Settings" menu option 
    - "Save modifications" dialog box before loading/resetting/exiting
    - window title now shows the settings name instead of the impulse name
    - updated 'tips' in Linearization and filtering tabs

0.9.6 2013-04-16
    - show frequency, amplitude and phase from current cursor position in
      graph
    - improved measurement handling:
        * drag and drop loading
        * loading speed up (twofold increase)
        * gain and time offset settings
        * polarity inversion and phase hiding functionality
        * bypass option
        * description: name, number of points, frequency and dB ranges
        * ARTA format handling (trailing spaces in frequency column)
    - Save measurement inside *.rephase settings files together with
      corrections and other parameters
    - new 'constant shape' EQ, both for linear-phase and minimum-phase EQ.
      Equivalent to a constant Q EQ at 6dB, it keeps exactly the same shape
      at any dB setting. It should be preferred to constant Q and
      proportional Q at high dB settings as those two are bound to their
      2nd order definition and have to stay within a ±90° phase range,
      thus leading to odd gain shapes at high dB settings...
    - "What's new" menu entry, exposing this changelog, instead of having a
      separate REDAME file
    - bugfix when loading settings from version prior to 0.9.2: filter
      frequencies were lost
    - going back to forced 'middle' in energy centering when only
      linear-phase corrections are used
    - curve capture functionality teasing...

0.9.5 2013-04-06
    - measurement import implementation, following HOLMImpulse import rules
      and interpolation strategy
      (first draft with limited functionality)
    - Nyquist frequency is now explicitly represented in gain result curve
      as a brickwall low-pass
    - result curves are now cleared upon settings reset or loading
    - improved energy centering algorithm
    - stop forcing energy centering to middle when only linear-phase
      corrections are in use
    - middle click on a fader reset its value to 0
    - bugfix for result phase curve unwrapping
    - bugfix on curves when polarity is inverted and a phase range larger
      than ±180° is chosen
    - bugfix for 1st order high-pass filters

0.9.4 2013-03-16
    - crash at start problems (previsously requiring temp/ dir content to
      be deleted) should now be solved
    - up to 16 banks can now be used in paragraphic gain and phase EQ
    - removed bank EQ tabs (settings saved with banks EQ will be
      automatically reported to paragraphic EQ banks)
    - improved graph range options: frequency and phase range can now be
      set and saved in settings
    - new phase wrapping implementation, automatically adapted to current
      phase range
    - view mode (compact/normal/large) is now automatically saved and
      restored from one run to the next
    - double-click on a fader value entry reset the fader to 0
    - improved raised cosine EQ. Interactions between EQs should now behave
      exacly like the "Ideal Graphic Equalization" exposed here in this
      application note: [url]http://www.nordicsales.dk/imgdb/docs/lakewh_981.pdf[/url]
    - better precision for frequency entries (fractional up to 5 chars
      total to fit the entry) and appropriate up/down key binding (0.1hz
      steps under 10hz)
    - real 2/3 and 1/3 octave frequencies in paragraphic EQ sections
      (mandatory to make the raised cosine graphical EQ "magic" work...)
    - improved biquad precision (constant Q and proportional Q EQs) by
      adapting the sampling rate of each biquad to its fc
    - increased Q range (0.1 to 100)
    - improved phase deg precision in paragraphic EQ
    - added ESS sabre frequencies

0.9.3c (misnamed 0.9.31) 2013-01-29
    - mini bugfix for the taps entry...

0.9.3 2013-01-29
    - new Paragraphic EQ implementation, with multiple EQ types:
        * constant Q minimum-phase (new default)
        * constant Q linear-phase 
        * proportional Q minimum-phase
        * proportional Q linear-phase
        * constant slope linear-phase (former implementation)
        * raised cosine linear-phase (beta version...)
      ( bank EQ section remains constant slope linear-phase )
    - FFT size can now be set by user (minimum size is two times the
      smallest power of two equal or bigger than the requested number
      of taps). Setting a larger FFT size makes generation and optimization
      slower, but can increase the precision of the optimization and also
      makes result curves more precise (just a visual effect for that one
      though: no effect on the actual impulse)
    - bug correction: negative gains can now be entered directly from
      the keybord in the Gain EQ Bank tab.
    - exit on repeated errors to avoid "panic mode" effect

0.9.2 2012-11-04
    - added back '24bit LPCM mono' output format, missing since 0.9.0
    - improved up/down key bindings on frequency entries
    - added up/down key bindings for taps entry, with color warnings for
      extreme values
    - new '1st order' and '2nd order' linear-phase filters, meant to be
      combined with an existing (and already corrected in phase) rolloff
      to obtain a linear-phase acoustical Linkwitz-Riley filter
    - made sure 'middle'+'float' centering ends up within -0.5/+0.5 sample
      from middle (was -1/+0.5)

0.9.1 2012-10-29
    - centering 'int' option was not working, this is now fixed
    - more explicit error message when loading a wrong setting file

0.9.0 2012-10-28
    - new file format '.rephase', saving/loading all settings, including
      correction settings, impulse settings, and graph settings
      (old '.jason' files can still be loaded, but impulse and graph
      settings get reset)
    - impulse file is now a three-part thing: directory, filename, and
      format extension. The directory is the only thing that is not saved
      in the '.rephase' file
    - '.rephase' files can be loaded upon start (as a parameter or by drag
      and drop on rephase.exe) or by drag and drop on the user interface
    - new offset option "float" for fractional sample centering, avoiding
      HF ripples in the impulse when the phase target is not a multiple
      of 180° at the Nyquist frequency
    - make "rectangular" the default window function: this should be the
      best choice for phase-only corrections, and "complex" windowing is
      not needed anymore with the "float" offset
      Note: "rectangular" window is still likely to be the worst choice
      for filter generation, when gain target goes far below 0dB...
    - new "ovelapping" filters, to be used for example in the midbass
      region, under Schroeder's frequency...
    - stereo wav formats are now available
    - bug correction in offset calculation in time=inv mode

0.8.4 2012-10-14
    - new time inversion option in general tab, to reverse the generated
      impulse, thus opposing phase corrections. This can be used to better
      visually track a phase target (inverse during correction, and return
      to normal before generating the impulse), or to evaluate the
      audibility of a given correction with headphones for example (in this
      case the convolution of the impulse will simulate the speaker before
      correction)
    - confirmation box when exiting whithout saving modified correction
      settings, and avoid asking for confirmation on reset when correction
      where saved or loaded without modification
    - Improves advice section in the linear filter tab, and add one in the
      linearization tab

0.8.3 2012-10-12
    - range choices in paragraphic EQs (up to ± 48dB and 720°)
    - improve arrow keys binding after click in faders
    - stop constraining frequencies to 16Hz-25khz in paragraphic EQs
      (now 1hz-99khz like in EQ banks)
    - ask for confirmation before resetting correction settings
    - bug correction when resetting settings ("wrong format")
    - stop using '.rephase' as default extension when saving an impulse

0.8.2 2012-10-07
    - try to play nice with multiple screens
    - new "large" layout, and "View" menu for layout choice
    - change Q interpretation for phase EQ to be more in line with gain EQ
      (to maintain an ascending compatibility, phase corrections saved from
      versions 0.8.0 and 0.8.1 get their Q divided by 1.8 upon loading)

0.8.1 2012-10-03
    - fix small bug with slope/ratio display when loading a FIR filter

0.8.0 2012-10-02
    - real-time amplitude/phase curves for both target and result
    - save/load correction settings (/!\ beware /!\, still experimental)
    - Horbach-Keele 'last' ratio (special tweeter) is now a different
      filter type for ease of use and clarity reasons
    - added some more window functions

0.7.6 2012-09-20
    - resolved (hopefuly) some issues with windows XP with the program
      refusing to actually start

0.7.5 2012-09-18
    - bug fix (crash during otpimisation step)

0.7.4 2012-09-16
    - bug corrections
    - optimization iterations are now faster
    - new optimization options ("moderate" and "extensive")
    - Horbach-Keele filters
      (ratio above 4.5 is the special "tweeter" ratio)
    - "Reject low" and "Reject high" filters for higher low or high rolloff

0.6.0 2012-08-26
    - first version on SourceForge

Attachments

  • FIR.PNG
    FIR.PNG
    61.7 KB · Views: 18,230
  • PEQ.PNG
    PEQ.PNG
    55.6 KB · Views: 17,807
  • IIR linearization.PNG
    IIR linearization.PNG
    55.9 KB · Views: 17,542

GB for Salas Reflektor-D Power Supply for Digital

This is a GB for Salas Reflektor-D and Reflektor-Mini Power supply.
This is a Low Voltage supply designed specifically for digital.
3.3V to 7V at 600mA max output using board level sinks.
Schematic and build info in PDF attachment at the bottom of post.

parts_pads.png


In the words of Salas -

The Reflektor concept is a shunt voltage regulator made around a current mirror circuit. To keep equal current between its input and output legs, the mirror reacts to flow disturbances and drives a MOSFET in opposite direction. A current loop simple regulator is another way to describe it. Specifically Reflektor-D is a version focusing on the voltage region common for powering digital source projects. For convenient mounting and setting it utilizes board level sinks and simple ways to choose its current limit and voltage output levels


The board is offered in a Matte Black 2MM thick PCB with 2 OZ copper. The cost is $16.00 plus shipping.

image.jpeg



It was a very easy build, it took me 45 minutes to complete and then start testing, and got near target voltages rather easily via the LED swaps.
Three different kits will be offered. A transistor kit, a resistor kit, and a full kit, which includes all parts for board except standoff parts and MOSFET bolts and nuts for the heatsink.

image-2.jpeg



The Reflektor-Mini was created to be installed into some tight spots. It is a much smaller board that will require its MOSFETS to be chassis mounted and electrically isolated for sinking. It keeps a lower height profile too. It is DC input only, needs an incoming DC feed.
It uses a J2 2SK117GR to stabilize and tune output voltage. For keeping small size It has a conventional 2Pin output. It's a slightly more complicated build than the original in some aspects and simpler in some other, but will serve it purposes in being a potentially lower cost unit to construct, as well as a smaller footprint.





In conjunction with the Mini release, the DC Flexy board is introduced. It allows for use of industry standard Snap-in Elcos or 4 pin/pole Mundrorf M-Lytic AG+. These boards can be used as Raw DC units to front end the power supply for the Reflektor-D Mini. They come in a breakable pair so to be used in double mono or multi isolated configurations of more than one regulator to feed. if used in a system of one regulator only, the spare may be broken off and kept aside since it can be utilized in a variety of projects needing Raw DC. Even for the standard Reflektor-D when done in its DC input mode version.

DCFlexy.jpg


The signup for the boards is in my signature.

Attachments

GB for Salas SSHV2 regulator

This is a group buy for the purchase of the second generation Salas Shunt High Voltage Regulator.

It's a newer design that Salas did, with CRT brilliant layout work and a bunch of us prototyped over the holiday season. We had a zero percent failure rate of setups amongst us. (Which I consider great, as most of my work , well needs work)


Let's get to the point.
The board is $15 USD.

Mini-Kits will be available. The cost is higher than I wanted them to be - so I split them in two.

img_05841.jpg



Again a disclaimer - this is the most dangerous product I have shipped yet. Failure to respect the high voltage of the product can have deadly consequences. This is not a beginners project!

The full specifications are in the PDF file. (attached as zip)

SSHv2Works.jpg


The google spreadsheet needs to be completed with your username and includes paypal payment details. Please don't post that address here.


Transistor Minikit 17 USD
1-Toshiba 2SK117's unmatched
3-Vishay 12V .5W Zener
1-20ma Red LED's
2-Supertex DN2540
1-IRF840
2-Fairchild MJE350
2-TO-220 Mica Insulator
2-Keystone plastic washers

Resistor Minikit 9 USD
2- Vishay PR2 - 2W 68K
1-KOA M/F 1/4W 100ppm 1.82R
1-3296Y Bourns 1/2W 100ppm 200R
1-3296Y Bourns trimmer - 1/2W 100ppm 1K
3-Vishay Bayershlag 50ppm 100R
1-271 Series Xicon 1/4W 50ppm 10R
1-271 Series Xicon 1/4W 50ppm 1.8k
1-271 Series Xicon 1/4W 50ppm 470R
1-271 Series Xicon 1/4W 50ppm 220R


SSHV2 FULL KIT - Does not include board (And no BIG heatsink for IRF840)
$50

1-Toshiba 2SK117's unmatched
1-20ma Red LED's
2-Supertex DN2540
1-IRF840PBF
2-MJE350
2-TO-220 Mica Insulator
3-12v 1/2W Zener
2-Keystone plastic washers (above is transistor kit)
2- Dale CPF2 100ppm 68.1k
1-KOA M/F 1/4W 100ppm 1.82R
1-3296Y Bourns 1/2W 100ppm 200R
1-3296YBourns trimmer - 1/2W 100ppm 1K
3-Vishay Bayershlag 50ppm 100R
1-271 Series Xicon 1/4W 50ppm 10R
1-271 Series Xicon 1/4W 50ppm 1.8k
1-271 Series Xicon 1/4W 50ppm 470R
1-271 Series Xicon 1/4W 50ppm 220R (above to 68K resistor kit)
1-Molex 4P Fixed Terminal Block Black
1-TE Connectivity 2P Terminal Block
2-274-1AB Wakefield TO-220 Heatsink
1-647-10ABP Wakefield TO-220 Heatsink
1-Wima 10uf MKP4 1100v
1-Wima .33uf MKP10 630v

Attachments

GB For Salas SSLV1.3 Ultra-BIB

SSLV1.3 - The Ultra-BIB
The group buy is for an updated Low Voltage shunt reg designed by Salas. This updated board uses no NOS JFETs making parts to supply and costs easier to predict. The circuit has been updated an initial clinical listening test have been positive, and noted as an improvement to the original BIB design. The board uses a universal set of components. So there is no need for calculations for voltage setting etc. As long as the appropriate transformers are used, the voltage can easily be set with the components listed in the schematics. The R1 current setting resistors will need to be calculated. This should make buying components for the project less worrisome.

Build Thread SSLV1.3 "Ultra-BIB"

Salas SSLV1.3 UltraBiB shunt regulator - diyAudio

Salas thoughts..
As the beloved SSLV1.1 BiB shunt reg was getting long in the tooth especially for NOS JFETS I had in mind for some time now to design its successor. The goals were: 1. In production parts 2. Much simpler to set up. 3. Better technical and subjective performance.

After many breadboard experiments and two prototype PCB iterations I feel that my goals were finally met. So here comes the UltraBiB

-Uses no NOS parts. (This was a short lived story, now the PF5102 and BC560c are NOS, but I have thousands)
-Can do 5V to 40V output without changing a thing in its configuration.
-Nothing to choose and match. No tolerances in predicting its CCS limit setting.
-Has 45dB more open loop gain and many times less output impedance than 1.1
-Sounds easily better.
-Its an electrically and mechanically drop in replacement for an upgrade.

Some initial quotes on sound

From Dimdim
"When we swapped it in place of the BiB 1.1 in my Soekris, the improvement was immediately obvious and not subtle. There was a general improvement in clarity and silence, but the biggest improvement (imho) is that the music appeared to have more energy in the lower mid area, where before it was kind of "dry". This was with Salas' very first prototype, built with standard (non-boutique) components. The board that I built with audio grade capacitors in the filter bank and MUSE BP caps in the output sounded even better."

From VGeorge
"The change in sound to better was apparent at first listen. As other have described, better clarity and definition throughout the audio range, but for me it was also apparent up high the frequency where I could hear more power but without any harshness."



The board will consist of three different reg's on one board. The board is V-Scored and can be broken into three boards, two boards etc.

On each full board there is a two positive regs and one negative rail reg.

Extra mini-kits (partial kits) are available to support building of the board - but leave out C1 Filtering cap and most resistors .

The boards will sell for $25 each, with $8 minimum shipping USA and Canada 10 min International, paypal only. Shipping has increased substaintially for international shipping, so I can only send a limited amount of boards and kits before it becomes a "package" as opposed to a "flat" and the cost typically doubles.


BOM List for board
UltraBIB Minikits - Google Sheets

UBIB Positive Minikit - Please note if needed 5V operation, J3 must be <7.5ma. You must also use RED leds here the 1.85 vF type. Please add this to private message so I can put a measured one to put in your kit. WP113IDT is a sub for WP1773ID LED stated below.


UBIB Negative Minikit
  • Like
Reactions: Alexlau 84 gogo

Marantz PM94D mosFETs

Hi! I'm new to electronics diy, basically retired and since no one near me wants to fix my receivers, yep, they will receive the service they need...somehow. my immediate issue is with the marantz pm-94 and finding replacement mosfets. The factory ones are 2sk405/2sj115, which are long gone, and according to an article that I read, somewhere, the replacement for those are 2sk1530/2sj201, which are no longer available. I really love this amp and would love to find a great pair of replacement mosfets. Thank you all!

Tony

Audio Research D115 repair

Hello does anyone have any detailed photos of the top and bottom circuit boards of an Audio Research D115mk11. Specifically I'm lookin for where the op amps, pass transistor and V17 tube socket are mounted. These were all removed from an Amp I just bought and I want to restore it back to original. There may be a small circuit board somewhere that had the op amps mounted on it? There are all kinds of parts and circuit modifications on this amp.. I need some clear photos to see how the original was. I do have the schematic but a board layout would save hours of time. Thanks anyone that can help. Guenth

Human frequency PERCEPTION range? (its not 20-20kHz)

Hello found these two quite interesting videos i wanted to share

Login to view embedded media Login to view embedded media
Cool guy! he makes a lot of these edge-case videos, quite interesting stuff

personally i definitely feel 0-20Hz
and i also can hear a lowpass set at 20-22khz even tho my hearing range rolls off after 15khz

while science claims 20-20khz is our hearing range, our actual "perception" of different frequencys might be complete different

Whats your expierence or thoughts on this?

I2S-Hat: A Raspberry Pi Hat for SPDIF <-> I2S Communication and DSP

For the past few months, I have been working on a project to utilize a Raspberry Pi with CamillaDSP in a standalone, fully-digital DSP system. To this end, I have created an add-on board, also known as a "hat", for the Raspberry Pi that allows fully bidirectional SPDIF <-> I2S communication with the Pi. As an advantage to other designs on the market, it also performs sample rate detection, providing the information to the Pi GPIO, and allows full software control of the SPDIF transceiver IC. No resampling of the incoming/outgoing digital audio is required, and it supports all stereo formats from 44.1kHz to 192kHz as tested.

The DSP capability of a modern Raspberry Pi 4 greatly exceeds that of the ADSP/SHARC implementations available. This should open up significant possibilities for much more complex and accurate DSP.

Please read more about the project on the GitHub page:

GitHub - raptorlightning/I2S-Hat: An SPDIF Hat for the Raspberry Pi 2-X for SPDIF Communication

KiCad files, Gerbers, and code are all available for anyone to use for their own build. This post is targeted for an open discussion about the board and implementation. Please let me know if you have any questions or comments about the information on the Git page, or if there is anything I can clarify further.

Enclosed.jpg

Electrocompaniet Ampliwire 100 With Issues

I've got one with issues, I'm WAY out of my comfort zone (just finished fixing a Marantz 2330B but I had the crutch of a SM and lots of internet archives).

Powering up on a 60W DBT bulb I got a malevolent glow but not being experienced with the amp I thought that moght not be enough current. Switching to a 100W DBT I got a puff of smoke (see pic for 220ΩX2), after disassembly I thought I found a shorted transistor but upon further review the trace is actually shorting them.

I'm pretty new but I don't think this is a thing?

I have the HFE schematic but I think I have the other one because they don't seem to marry up.

Anybody with AW100 experience is welcome to chime in.

PXL_20230802_022754450.jpg


2.jpg

PXL_20230803_022128953~2.jpg

For Sale Apex PA88 High Voltage (+/-250V) Op-amp modules

I have a pile of boards that each have two Apex PA88 high voltage op-amps on them with heatsinks. these are rated for something like 450V and 100ma each. the boards were used to drive Piezo sensors so the are set up with a single ended input, One opamp inverting and one non inverting with a bridged output. power requirements would be +/- 200v up to 225v max. so they could output 400-450V P-P at 200ma! gain is set around 18-20db something like that but can easily be changed.
The Apex PA88 are crazy expensive new. so they have been hanging on to these old boards and have finally decided to let go of them. so I am curious if they would have any use for audio stuff? maybe for driving electrostatic panels directly? or as Tube drivers or. driving output transformers directly for testing or something.

$20each plus shipping

Attachments

  • 20240827_185344.jpg
    20240827_185344.jpg
    348.7 KB · Views: 168
  • 20240827_185346.jpg
    20240827_185346.jpg
    370.5 KB · Views: 159
  • 20240827_185356.jpg
    20240827_185356.jpg
    368.1 KB · Views: 161
  • 20240827_185422.jpg
    20240827_185422.jpg
    400 KB · Views: 162

My first attempt at winding a transformer + iron core analysis and electron scanning

Good evening, I wanted to share the DIY story (2 years old now) on how I winded my first mains transformer starting from a broken one.
DISCLAIMER: I am a hobbyist, I do not claim my work to be professional or that this is the correct way to do it, this is just A way to do it.
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.
So if you wish to understand more please visit the link.

1) I started by taking the old transformer apart, removing the burnt copper windings and the lamination (being careful to not remove the thin layer of insulation on them, you WANT your lamination insulated to avoid eddy currents in your iron core and prevent power losses)

2) I winded the copper, using nothing but my hands and two whole spools of double enameled copper wire. At the end of the page in the link you can find the calculations of the necessary amount of windings, you don't want to exceed the max magnetic flux density of your metal and turn all that energy into heat.
Transformer1.jpg
Transformer2.jpg
Transformer4.jpg


3) Put the lamination back in, then used a vacuum chamber to evacuate the transformer and replace the air with insulating varnish

Transformer5.jpg
Transformer6.jpg
Transformer7.jpg

4) Drying, assembling and testing are required before using the transformer in the device.

photo_2023-12-07_00-40-27.jpg
Transformer10.jpg
Transformer11.jpg

I didn't have much hope this would work, but the device is still operational after two years and the transformer doesn't hum or vibrates, indicating that the vacuum and varnishing really did its job. The amplifier is a classic 35+35w 2SC5200/2SA1943 from the '80s recovered at the scrapyard. The smoke detector is always on, just in case...

In the webpage I included more pictures and an analysis of the composition of the metal core that helped to understand the parameters for the right amount of turns.

Bench power supplies...

I'm starting to dabble in solid state a bit (mostly a tube guy) and want an inexpensive bench power supply to help. I have a Lambda high voltage supply that was given to me for working with tubes but that won't help with ss. My first ss project is a subwoofer amp so I need something that can handle 150W output or more, ideally under (or close to) $100. I'd like it adjustable to at least 30v, but 60v would be ideal. I've been looking around and new adjustable switching supplies are everywhere and can be had for under $100. I've also found a few linear supplies that might fit the bill, but they tend to be much lower current for a given voltage and price. I'm not looking for a high-end, highly accurate supply here (though maybe I should...).

So I have a few questions. First, I know linear supplies are in general quieter (in terms of output) than switching, but how important is that for building and simple testing? And if it is important, could I add a small cap across the outputs of a switching supply (small for higher frequency noise) to help reduce that noise? I've read that many switching supplies are also not very good at fast transients so I assume I could in theory also add a large reservoir cap. Basically, will I be happier with a linear supply even given the lower current/power? Any recommendations for something that fits the bill?

Or, should I DIY a linear supply to meet these general goals? I did some research on this many years ago but gave it up as too complex. I think I could do it now. If so, can anyone suggest some designs? I could probably design a very simple supply, but am looking for something a bit more carefully designed than I could do.

OK, I'm not a highly skilled builder (though I've built quite a few amps (tube and solid state), preamps, phono-stages, speakers, etc.) and I understand that this may be a lot more complex question than I realize, but some general replies would be helpful. At least for now. Thanks so much!

DIY bass sound absorbers

Since sound absorbers like I need are not sold, I came up with the idea of building them myself.
I need damping in a relatively narrow band on bass frequencies, at about 40, 60 and 80 Hz, it makes sense to build dampers with a membrane.
I found one page what seems to give easy absorber calculations: http://mh-audio.nl/Acoustics/PResonator.html
I also found an example where the damper made based on the previous calculation seems to work https://www.musiker-board.de/threads...echnet.741282/
You can see see results on example link on the post before last on pictures with header "Zur besseren Übersicht vorher - nachher", I used Google translate to read this page.
However, there are small doubts for me.
How the calculation sheet doesn't take into account membrane stiffness at all, only mass and panel depth?
How can such a small absorber have such a large effect on 30 Hz as seen the sample page images?
Did I miss something here?
Have someone other bass sound absorbers built ideas what can be DIY?

Kicker ZX700.5 low volume from Amp 1

Hello I have a Kicker zx700.5 that I got at a yard sale. Amp2 seems to work fine but amp1 outputs have very low volume in comparison like it's muted. Sub works fine. I can hear something buzzing on the amp but I haven't located it yet. Any suggestions? Thank you

Here are some measurements I took from the preamp board to speaker ground:

Fr in -18.3 mv
Fl in 9.6 mv
Rr in -19.3 mv
Rl in 10.0 mv
-15v -14.14 v
+15v 13.95 v
Rem 11.51 v
Gnd a -4.5 mv
+ Batt 11.63 v
Gnd p -4.5 mv
Rem- bas 12.3v
Sub in 15.3 mv
Sub pre -4.4 mv

Attachments

  • PXL_20250211_170556286.jpg
    PXL_20250211_170556286.jpg
    644.9 KB · Views: 92
  • PXL_20250211_170618219.jpg
    PXL_20250211_170618219.jpg
    664.1 KB · Views: 81
  • PXL_20250211_170626019.jpg
    PXL_20250211_170626019.jpg
    743 KB · Views: 74
  • PXL_20250211_170636476.jpg
    PXL_20250211_170636476.jpg
    618.7 KB · Views: 68
  • PXL_20250211_170802270.jpg
    PXL_20250211_170802270.jpg
    690.3 KB · Views: 69

Steinway Lyngdorf Model B

I’m trying to confirm how the 6 woofers (3 forward facing and 3 rear facing in adjacent vertical arrays) are configured in this speaker, including the baffle arrangement. It appears from some photos and videos that I’ve seen that the front and rear baffles only extend half way across the width of the speaker so as to allow the sound wave emerging from the rear of each set of 3 speakers to travel to the front or rear of the speaker, as the case may be. To put it another way, both sets of 3 woofers are operating as true dipoles and are projecting sound into the room in both directions.

Is this correct?

Also, presumably the rear facing set of 3 are electronically out for phase with the front facing 3 so that all 6 push in the same direction at the same time - ?

Troubleshooting an NAD T751 With Low Output on One Channel

I am more of a tube radio guy, but I am trying to figure out a low output channel on a friend's solid state amplifier. I have tracked down a number of symptoms and I am hoping that someone might be able to help me with a diagnosis. I would be very appreciative of any ideas you might provide. The schematic for the Front Amplifier board is shown, but I am not sure if it will be legible.

  • There is very low output on the front left channel, regardless of the input source. (Yes, speakers have been switched - problem remains on the left side.)
  • When the receiver is turned on, the left channel is too quiet to hear, but slowly increases in volume until it is about 25% of the right channel. The sound is intermittent until the volume is increased, when it becomes steady, though quiet.
  • A number of component in the left channel are warmer than those on the right. For example, R521 (right channel) is 44 C, while its corresponding resistor on the left (R522) is 51 C. R523 is 31 C and R524 is 44 C. More interesting is that C525 (470 uF / 63 V) is 29 C and C526 about 36 C.
  • Measuring the Voltage at the idle current measurement points, I found the right channel measurements at P501 to be 4.9 mV DC / 0.1 V AC, whereas the corresponding left channel measurements at P502 were 21.6 mV DC and 0.02 V AC. Cap leakage?
  • The problem is unlikely to be in 10,000 uF filter caps (C561 and C562) as they feed both right and left channels.
It seems to me that the problem is likely to be a leaking electrolytic cap on the left channel rather than a bad transistor and that I should remove the board and replace the electrolytic caps. Does that make sense to the more experienced techs?

Thanks as always for any help/suggestions/insights/next steps you can provide.

Andrew

1730310269121.png

Electrocompaniet Ampliwire 100

I have an opportunity to pick up an Ampliwire 100 with a shorting problem. Doing a quick take I was able to determine that the problem is from the right channel after the power supply. By any chance does anyone know if this particular model amp is plagued with problems? A few folks seem to think so - I just do not have much history on its reliability.

Thank you for the help.

what do you really think of Wilson Audio?

yes we all know that anyone in this forum can make much better speakers than wilson audio does, but still, what is it that wilson audio speakers do so well that so many audiophiles love? and audiophiles and critics have loved wilson audio speakers for a long time now!

it seems absurd that wilson audio would be a bad speaker manufacturer but still have so many fans

https://www.stereophile.com/content/icing-munich-cake-mcgrath-fon-nagra-wilson-impex

Restoration and Modification of Pioneer PL-71 turntable

This thread will be for the documentation of the restoration and (probable) modification to a Pioneer PL-71 turntable.

Why the PL-71? A couple of reasons:

1) It's a good example of pre-PLL direct drive. The motor is very quiet and it's bearing structure are good.

2) It has a wonderful tonearm made by Acos.

3) The turntable is generally considered to be a good example of "better than the sum of it's (quite nice) parts".

4) It's the turntable I grew up with, and 5) I didn't actually expect to win the auction.

Fantastic reference thread here - A new toy - PL-71.....

IMG_2428.jpg

Here it is in the condition I recieved it. The wood is a bit dry, there is a general light yellowing of all the metal parts indicative of it living in a smoker's household at some point, but other than that, it's in very nice shape.

IMG_2431.jpg

And here it is after a thorough cleaning. The yellowish tint is off the metal, the wood has a coat of Danish oil, and generally it look much, much better.

I didn't take any photos as I was cleaning it, just imagine a bunch of paper towels, cotton swabs, alcohol, wood oil and the like all strewn about. It took approximately 1.5-2hr of scrubbing, dabbing, cleaning, wiping, brushing and elbow grease.

I still need to treat the mat, it's clean, but the rubber needs something to restore a bit of moisture to it.

IMG_2433.jpg


IMG_2435.jpg

The dustcover is is great shape for it's age. I'm very pleased.

IMG_2436-1.jpg

The bottom cover is very 1973. But the metal chassis bottom is a neat piece, making a metal interface for the sprung feet into the wood chassis that makes up the rest of the table.

IMG_2437.jpg

Here's the up-skirt shot.

IMG_2448.jpg


IMG_2441.jpg

A few things worth mentioning, the tonearm is rigidly coupled into the chassis, and with the tagboard and jacks easily removed, would be a good candidate for a continuous rewire or conversion to DIN if that's you kind of thing.


IMG_2439.jpg

This board is mainly for AC distribution.

IMG_2438.jpg

DC rectifier and 'regulator' (Really just a zener-referenced cap multiplier.)

IMG_24401.jpg

The power/speed selector switches and speed trim pots.

IMG_2442.jpg

These little screws holding the motor cover were a royal pain to remove.

The power transformer is mounted on rubber feet. Remarkably quiet. Of course it would benefit from being in it's own external case, and I may try that. It also has a universal primary and voltage selection with one of those neat plug thingys.

The motor is rigidly coupled to the chassis and the control board is under the black cover.

IMG_2445.jpg

Cover removed showing the motor control PCB. The 38yr-old capacitors need to be replaced. (Yes, one is already replaced in this photo...)

IMG_2446.jpg

Not too bad of a job. There are a number of wires that got the the motor windings on the other side of the PCB that seemed to be in the way, but other than that, it's straightforward. No values changed

IMG_2443.jpg

The regulator board also got a set of fresh capacitors. The filter caps are a bit bigger than stock, but the can size is still the same. 😎

IMG_2447.jpg

Dead soldiers.

IMG_2450.jpg

I replaced some screws that hold the strobe to the chassis, over the years 3 of the 4 screws had fallen out. I had to scrounge for hardware that fit, but now it's solid.


IMG_2451.jpg

Lastly I added some secret lubricant to the bearing, I got it from a kindly old Dutchman who horse-swapped it from a Polar Bear named SY. I have little idea what's actually in it, other than it was made for low-heat, high-pressure interfaces, specifically TT bearings.


In my opinion, all I did was to get the 38-yr old table back in a condition similar to when it left the factory. So far it's just a tune-up, no hot-rodding, no mods.

Yet.

🙂




Edit -

There has been a number of reports of people trying different mats, and always returning to the stock one. I have a theory why -

(click link for video) https://www.youtube.com/watch?v=zxbj0a-AzuI
  • Like
Reactions: Bonsai

Electrocompaniet Ampliwire 100 (AW 100 AW100) different Versions

From this power amplifier I have service manual together with two different schematics (year of production approx, between 1979 and 1983). Additional I have newer version for repair service. The schematic, that I have create (Reverse Engineering), is different again compare to the already present orig. versions. Even the cabinet version is different in opposite to the older version.
How many different versions of the AW 100 are exist at whole ?
Thank you for your advices.
  • Like
Reactions: Nessman

DIY 4 Phase Sinewave Generator for Turntable Motor Drive

This is a shared DIY project for non-commercial use.

The SG4 generates 4 low distortion, high accuracy sinewaves suitable for driving conventional audio power amps to create a multi-phase drive for turntable motors. The generator outputs a reference sinewave at 60/81Hz or 50/67.5Hz on the 0° pin. The 90° pin outputs an exact replica of the reference sinewave, but shifted +90° (Cosine) for driving 2 phase AC synchronous motors. The 120° & 240° pins output an exact replica of the reference signal, but shifted +120° and +240° respectively for driving a 3 phase motor. The SG4 is a sinewave generator only. You will need to add the necessary audio Power Amps and step-up transformers (if needed) to create the final signal to drive the motor. Low cost linear and class D amps are readily available on e-Bay and other on-line sources. Working with High Voltage can be dangerous. Do not attempt this part of the project if you are not trained in handling power electronics: Seek competent technical help if needed.

The PCB uses all thru-hole components for easy assembly, but some soldering skills are still required. The µP is a PLCC package but there is a socket for it on the board.

The project consists of a bare PCB, a parts "kit" available as a shared cart from Mouser electronics, a µP with the operating system pre-programmed into it and the on-line documentation you see here.

The PCB is available from OshPark PCB fabricators at the following link: https://oshpark.com/shared_projects/E5zXjeJd
The PCB is created in multiples of 3 for a cost of ~$45 or $15/board.

24-Feb-2022 The parts kit at Mouser has been updated to include the MCP101 reset controller that replaces the DS1833 part. The pin out of the MCP101 is the mirror image of the DS1833 so the PCB has also been updated to Rev C to reflect this. If you are using a Rev A or Rev B PCB, you must insert the MCP101 part backwards; the Rev C PCB does not require any change.

The parts kit can be ordered from Mouser Electronics: http://www.mouser.com/tools/projectcartsharing.aspx
Enter the Access ID code: 7A6A645FFA. The parts kit to build 1 PCB costs $32.34.

The pre-programmed µP is available in the US from DIYAudio member Seth Hensel via email: sethhensel (at) icloud.com. Cost will be ~$12 plus $8 handling plus shipping.

The pre-programmed µP is available worldwide from DIYAudio member ralphfcooke via PM. Cost will be ~£10 plus shipping.

The following documentation is available below to aid in construction of the project:

SG4 Schematic.pdf
SG4 Parts Locator.pdf
SG4 Assembly Instructions.pdf
SG4 BOM.pdf (Generic bill of materials with part references)
SG4 CART.pdf (Mouser cart with mfr's part numbers and costs)
SG4.zip (Gerber X274 files if you want to use your own PCB fabricator)
SG4.png (X-Ray view of the PCB w/traces, pads and silk screen)

The SG4 uses a 20 bit DDS core implemented in software to generate the reference sinewave. Frequency resolution is 0.01Hz. Frequency range is 40.00-70.00 Hz for 33 RPM and 60.00-90.00 Hz for 45 RPM. There is an on-line video showing the frequency operation Here and a video for phase adjustment Here.

There are four 8 bit phase accumulators to generate the four phases. The reference signal is fixed, but the other 3 are adjustable in 1.5° steps ±15° maximum. D to A conversion for the 4 signals is done with 8 bit PWM at 18kHz. There are 4 LPFs on the board to convert the PWM signal to analog. The outputs are DC coupled, 5VPP and centered at 2.5VDC. Distortion is ~0.5% (-46dB) and frequency stability is 30 PPM.

Update: The firmware has been updated to version 1.02. A 7 bit linear taper attenuator has been added to ramp the voltage from 0 to 5VPP at start up when exiting standby mode. This should prevent the amplifiers from shutting down when first started as they will have time (~650mS) to overcome the core magnetization of the transformers. I also added a reduced output voltage mode, where the output voltage will automatically be reduced to a user programmed level after 5 seconds. The level is adjustable from 128 (maximum) to 64 (half voltage) in ~40mV steps which equates to ~1V steps at the transformer outputs.

19-Nov-16: I returned the phase adjustment to 1.5°/step. One of the peculiarities of using a 16 bit phase accumulator with an 8 bit DAC (PWM in this case), is the limited precision math can create different points where a carry occurs (and thus an additional phase step). For most frequencies, this isn't noticeable as the difference in phase shifts is usually in the mSec range. In certain cases, the math works out where it becomes quite noticeable and in the audio range where it could affect performance. Using a 16 bit phase accumulator in the SG4, a nasty phase spur will occur on either side of 81.92Hz, which is fairly close to the frequency needed for 45 RPM. The new firmware hasn't hit the field as of yet, so there will be no need to do another exchange. V1.02 will ship with the attenuator capability, but will retain the 1.5° phase adjustment of the original.

Users who upgrade to the new firmware should perform a Factory Default Reset after installing the new firmware.


28-Dec-2016: Just added the PCB files to OshPark for a Rotary Encoder to SG4 interface PCB. The circuit converts the 2 quadrature signals from the encoder to a single pulse train on the UP pin when turned CW and a single pulse train on the DN pin when turned CCW. The momentary push button switch of the encoder is connected to the STBY button of the SG4. This allows all of the normal operating functions to be performed by one rotary control.

The 1 inch square board uses all surface mount components in order to keep the size down. The IC is a CD4013 in SO14 package. C1 and C2 are both 0.47uFd Tantalum caps 10V rating in a 1206 size package. R1, R2 and R3 are not necessary if you use the Arduino Rotary Encoder which has the pull up resistors already on its PCB. If you use another encoder without pull ups, add the 3 resistors (all 10K 0805 size). Vcc is connected to the 5VDC output of the regulator on the SG4.

You can order the PCB here ($5 for three):

[url]https://www.oshpark.com/shared_projects/AbsVI39H[/URL]

The Encoder PCB can also be ordered with thru-hole component layout instead of SMT:

[url]https://www.oshpark.com/shared_projects/W6QWOCOF[/URL]

1-May-2017: Version 1.03 of the SG4 firmware reduces the lower frequency limit to 1.00 Hz for both 33 and 45 RPM. The reduced lower limit was needed for 3 phase BLDC motors, some of which require 20Hz for 600 RPM. If you are not using a 3 phase BLDC motor, there is no need to update the firmware.

16-May-2017: A DIY 3 phase amplifier project is now available to drive a BLWR172S-24-2000 or BLWS231S-24-2000 BLDC motor from Anaheim Automation.
It is not a universal controller and will only work with these two motors: 3 phase class D DIY BLDC motor drive amp


A suitable audio power amp and step up transformers are available here: 60-wpc-amplifier-diy-turntable-motor-drive


13-Dec-2021 The PCB has been updated to Rev B and the firmware has been updated to v1.04. These changes add the ability to use a 2 x 16 character LCD display that has an I²C interface PCB with a PCF-8574A interface chip. Version 1.04 firmware is backward compatible with the previous PCBs so an LCD display can be added to a previous build. The details of all the changes can be seen here: SG-4 Version 1.04 Update

It is important the LCD interface chip is a PCF-8574A and not a PCF-8574; the two chips work nearly identically, but have different address schemes. The new SG4 firmware will only work with the PCF-8574A addressing.

Attachments

Trying to Restore Mitsubishi DA-R35, Zener Diode Question

I have a couple of these nice 1980 receivers. Not top shelf material, but quite good. They use ICs for L/R voltage amplifiers (STK 3076/3106) and individual ICs for L/R power amps (STK-1080 II), three ICs in total 😕. Both have blown voltage amplifier ICs, but I am finding other surrounding problems such as 4558 op amp blown and zener diode MZ324.

Does anyone have an idea on this MZ324 zener diode and its specs? I need to find a current match.

I am attaching the part in the schematic with the MZ324 zeners highlighted in red. Maybe someone can have an educated guess as to one that will work in both positions.

Thanks in advance.

Attachments

  • ZENERS.png
    ZENERS.png
    211 KB · Views: 322

New project - PSU confusion

Hi,

Got a visit from my neighbor yesterday while enjoining my ACA + Korg, apparently it was to loud. Enter Honey Badger.

Still going through all the docs here and creating BOMs so I can pull the trigger on the parts, and of course have a ton of questions.
As the amp section is pretty much as is regarding the PCBs and parts (except some transistors and caps, still figuring that out...), I'm not sure about the PSU. I see there are two on the store: Nelson's bipolar PSU and Universal PSU.

The idea is to go with the 2x150W stereo config, and update to mono-blocks later if needed....

1. Which PSU to go with?
2. One or two PSU boards?
3. What spec toroidal to match with the PSU board(s)?

As always, much appreciated for all the help!

Cheers!

Cambridge 640A V2 with DAC module Musical Fidelity V-Dac V1

Hello

I modified my amplifier Cambridge 640a V2 & I install inside the amplifier a dac module from Musical Fidelity V-Dac V1 (Usb-Optical-Coaxial Inputs)
I need space for this Dac and I remove the A-Bus PCB Board, and I disconnect on Input PCB Board, connectors CN9 and CN10 (From A-Bus PCB Board)

The amplifier working fine, but like you, the remote control is not working !!!!!!

I am not a specialist of electronic systems
Do you have a solution to solve may problem, because I saw in the past you have same trouble
Let me know
Regards
Rabia

Attachments

  • 640A V2 - A-BUS PCB Board.jpg
    640A V2 - A-BUS PCB Board.jpg
    228.5 KB · Views: 40
  • 640A V2 - INPUT PCB Board.jpg
    640A V2 - INPUT PCB Board.jpg
    138.3 KB · Views: 40

For Sale Miro AD1862 DAC

I am selling my fully built AD1862 DAC based on Miro's design. Used all good components like DAC chips from Rochester, Opamps - OPA1655DR and OPA1611AID and I love both of them so will include both in the sale. JLSounds i2s dac board sitting on top of the dac which is USB powered and I have not modified it to power externally. I like this way as I use a galvanically isolated USB from my streamer. RCORE to handle the power supply using Miro mini psu works perfectly. IEC socket have used a EMI/RFI filter built in for filtration. So its a clean and distortion free DAC playback I can hear.

The dac is being used in my second system which replaced my Soekris R2R 🙂. I am moving homes where unfortunately I do not have enough space for 2nd setup, so most of my second system is being disposed off. Slight scratch marks on the bottom plate because of usage otherwise rest of the chassis is pristine without any marks.

Asking price $576 including CONUS shipping and I will include a IEC power and a USB cable.

More information of this DAC here :
https://electrodac.blogspot.com/p/dac-ad1862-almost-tht-i2s-input-nos-r.html
https://www.diyaudio.com/community/...s-input-nos-r-2r.354078/page-209#post-7049224

Attachments

  • IMG_0001.jpg
    IMG_0001.jpg
    556.3 KB · Views: 377
  • IMG_0002.jpg
    IMG_0002.jpg
    468.1 KB · Views: 394
  • IMG_0003.jpg
    IMG_0003.jpg
    420.7 KB · Views: 334
  • IMG_0004.jpg
    IMG_0004.jpg
    438.3 KB · Views: 335
  • IMG_0005.jpg
    IMG_0005.jpg
    649.1 KB · Views: 300
  • IMG_0006.jpg
    IMG_0006.jpg
    452.2 KB · Views: 356

Building a Crossover for 450w RMS Woofers

Hey Diy Audio Crew!
This first post here, don't be to hard on me:')

in the planning stage of building a pa loudspeaker, including six 6" woofers, rated at 75w rms each. (Dayton Audio PA165-8 6)
wondering what sort of cross-over components won't blow at 450 wats. going to cross it over probably at 2500 hz or earlier.
also thinking of using the Pyramid TW44 Heavy Duty Titanium Super Tweeter with the included crossover. (posting that here in case people see a fault in my parts choice.)
just having struggles with finding cross-over components that won't blow. 🙂

all parts on parts express if you want to look it up.

Revox A77 Re-Cap Job

Hi All

I assume someone here has done this already. I already sourced most of the trimmers for this job. I have already tackled the power supply too.

I am looking to re-cap these boards:

Switch Board
Input Amplifier
Record Amplifier (2x)
Oscillator
Record Relay
Playback and Drive Amplifier (2x)

Gee, this cap list is looking evil right now. Can anyone share their experience (and maybe their list) for this?


Some of the electrolytic values are hard to find too. Some of the 3,3uF values could probably be replaced by Film (MKP) once I look at the schematics. I am tempted to try Film caps in what looks like coupling positions. Any reason not to do this?

My machine is a high speed IEC one. Sounds really good. Fixing the meters was a challenge, but they are once again good.

FaitalPro XL3000 series

At ISE 2025, FaitalPro released its new beast 21XL3000.

21XL.jpg


The 6,5" CCAW coil 3000W/6000W behemoth.
Normalyzed motor force of ~313 puts it in the high end class with all these new B&Cs, 18Sounds, RCFs and such.
Looking forward to reviews and use cases.

Now, birds on the trees are chirping that we might:
A) See 4Ohm version, because this will be really difficult to feed with mortal-grade amps.
B) See 18" version. That would be way too awesome. I praised FaitalPro crew to them myself and asked for tad longer voice coil (to add my voice to that idea if it already exists, no way they are going to listen just to ME 🙂 ).

If that is the case, I am considering to not follow up on my oncoming 18" purchases in the name of this.
The price seems to be somewhat higher, but still realistically competitive with quite some competition being more expensive already.

Anyone also interested and waiting to put it into some sick design?

Preliminary datasheet here:
https://faitalpro.com/highlights/2025/21XL3000/files/21XL3000 - Preliminary Datasheet.pdf

Should be available towards the summer.
  • Thank You
Reactions: GM

is Beyma TPL 150 sonically identical to TPL 200 ?

is it just two different lengths of same pleat or is the pleat different ?

because the 200 is only 33% longer yet claims 50% higher power handling so i wonder if it may also be wider and / or deeper in the pleat.

they also seem to have slightly different frequency response in the specs, but that may be due to different measurement conditions etc.

and yet the higher power handling of TPL200 is consistent with it having slightly more rolled off top half an octave if that power handling is achieved by having deeper pleat. deeper pleat would mean more conductor surface area and perhaps more power handling as a result.

in other words you would expect a 4" compression driver to have slightly more rolled off top end than a 3" so maybe Beyma felt you would also expect the bigger AMT to be a bit more rolled off and made it not just longer but also ... different ?

i understand even with the same pleat they would sound different due to different radiation patterns but if you were to array them those differences would mostly cancel out ...

so is it the same pleat or not ?

Professional hobbyist

Hello!
I have been building and designing loudspeakers from 2019.
Master's degree in Acoustics and Audio Technology.

I'm an entrepreneur, alltough my business is not (currently) in audio. (It's in building acoustics and noise)
I have my own dedicated and acoustically treated "Studio" where I listen to music while working 🙂

I have been reading this forum for a while, but finally have decided to start contributing with my own projects.

-Jesse
  • Like
Reactions: tmuikku

Tandberg 3000x Transformer diagnosis

Hi all
Just acquired a tandberg 3000x reel to reel player that I knew needed work but didn't check it Powers on, which it doesn't. I've checked fuses and cable into transformer for continuity, still need to check the switch but want to check the transistor as well before getting to capacitors and pcb work. My knowledge of such things is non existent but can someone tell me if this looks right before I give it a good probing?
Thanks all
1000037367.jpg

Back to my System - new to this forum

Hi,
Let me introduce myself.

I'm Revoicer (really John). I'm from Buckinghamshire (Bucks) in England/UK.

I have just gotten round to unearthing my old HiFi set-up. [Stored for approx 30 years.] And am looking to integrate bits with my more modern HiFi components.

I have joined this group to get some advice.
I am looking forward to receiving your pearls of wisdom to my mind taxing (or bored you senseless) questions.

Lateral MOSFET - Capacitor Betwen D and GND

Hi, i im layouting new pcb for lateral mosfet and found this schematic of OPS.

20211206081535_Figure3-DH-220C-MOSFET-PowerAmp-P1.jpg


My question is about decoupling caps on DRAIN of mosfets. On some schematic i see 220nF between DRAIN and GND, and on some 470nF or 100nF value.
Does this mean that i need to add to each lateral mosfet DRAIN pad as close as possible 470nF?

What is recommended value for DRAIN capacitor for each mosfet (i will be using 2 pairs)?

I will be using EXICON lateral mosfets 2pairs in TO264 case.

And i read that MKT 2,2uF at DRAIN improves THD below 10kHz.

And some members suggests adding 1000uF/63V electrolytic cap to DRAIN and GND and solder on legs of this electrolytic cap 100nF MKP1837 capacitor.

I im not sure if this large capacitor will not give trouble to drain of mosfets when mosfet turns on/off as this large capacitor charges/discharges.This idea have description that it improves large currents/demands from fast slewrate audio amplifier and it is close to DRAIN to give accumulated energ to DRAIN for fast slewrate amplifier.

So what is recommended way?

JVC Victor TT-81 Schematic Diagram & PCB

JVC Victor TT-81 Schematic Diagram & P.C.B.

Schemes and boards in two versions for TAS-19A and TAS-19B boards
I redrew it myself from a poor quality scheme, maybe there are errors somewhere,
so let me know and I will redo it.
I put in a lot of work and patience, so I ask for help in checking and making comments.
Thanks in advance!
Victor TT-81 TAS-19A PCB
https://cloud.mail.ru/public/6VXX/eWdCgKCVR
Victor TT-81 TAS-19A Schematic Diagram
https://cloud.mail.ru/public/Tga7/ithR4GWWt
Victor TT-81 TAS-19B PCB
https://cloud.mail.ru/public/CWJ9/h7dtw1CV7
Victor TT-81 TAS-19B Schematic Diagram
https://cloud.mail.ru/public/PQFR/CpwRdUR34

Attachments

Pac AP4 GM61 tuning

I recently put in my sound system in my new truck with Bose, I am using the AP4 GM61, I just pulled everything from my old truck and put in new one and everything seems fine except the bass, it's non existent until I have the bass knob turned 3/4 way up and by then it sounds muddled or too much especially at medium to lower volumes. The module was installed by a stereo shop in the previous truck and I do not know what all configuring they did which might of been none cause the chime volume still about pierced my eardrums. I did set the " minimum volume " and wonder if that messed it up. The chime volume is perfect now after I set it. I am thinking or setting back to factory settings but don't know how good it would sound like that or if too difficult to try and tune with a PC. Is there a certain position to have the knob while tuning the system? Thanks in advance.

Ewidance warmup

Hi all,

Long time occasional reader, but not a big contributor since English is not my native language (French, sorry🙂 ), I’m a musician, maker and audio DIY mainly on musical instruments / devices. I'm occasionally maintaining amps, effects & other devices. I love vintage guitar tube amps and recently built a tube tester ( excellent uTracer 3+ from Ronald Dekker) to help to match and qualify tubes. I also founded and directed the Montpellier FabLab (https://labsud.org) for 6 years. Recently retired from a technical director position, got now time for my passions around music: building/repairing amps and effects, lutherie for building guitar and bass (and tuning them), rehearsal and live performances with those equipment.... But my main goal is continuing to learn & share, as technician, musician and human...

https://civade.com : DIY Blog

Have a nice day!
  • Like
Reactions: krivium

PSB Stratus gold X-over help

Hello all. I need to rebuild the x-over on my beloved Golds after I fried the x-over board, I mean burnt by an overheated resistor to the point I can't follow the board traces to get the schematics! I have called PSB and they would only give me the values of the resistors, but no schematics. Can anyone steer me in the right direction??? I will rebuild them one way or another and will post the outcome.

could this work ? ( planar dipole / CD horn coax )

let's say HYPOTHETICALLY you took the large format B&C ME464 horn:

https://www.bcspeakers.com/en/products/horn/1.4/0/ME464

1.4" BMS midrange

https://www.bmsspeakers.com/index.php-62.html?id=bms_4594nd-mid

and Radian LM8K mid-tweeter configured as dipole

https://radianaudio.com/collections/ribbon/products/lm8k-wide-band-planar-ribbon-transducer

put the planar coaxially inside the horn and crossed them over at 1.5 khz ...

the black line in the chart below is for DIPOLE operation of LM8K WITHOUT BAFFLE

1742266611829.png


so as you see you don't lose any output by going dipole - if anything you get BETTER frequency response from the dipole ( above 1.5 khz )

there is a fairly obvious SLOT shape in the middle of the ME464 horn where the LM8K would fit almost perfectly ... of course i am not talking about blocking the output of the compression driver - there would perhaps be about 3 inches of space left on all sides of the LM8K for the sound from BMS mid can go around it ... but also some of the sound would go THROUGH it as well ...

then to control the back wave from the LM8K you would put some acoustical damping material ( like wool ) right behind it as well as line the sides of the horn with a thin layer of foam or felt ...

the logic being that below 1.5 khz a horn loaded BMS mid is about 110db / watt

1742269398790.png


versus a measly 90 db for the LM8K ... on other hand the BMS will roll off above about 3 khz but the LM8K actually has rising response all the way up to 10 khz ...

so you can create a sort of a co-axial driver, not unlike this radian:

1742267686196.png


https://radianaudio.com/collections/ribbon/products/6crn38lt6-line-array-driver

only instead of a cone there would be a midrange horn and the planar would be a dipole instead of having a very shallow chamber ( like what i assume Radian is using ).

i think dipole could be beneficial here because by allowing some of the BMS output to go THROUGH the dipole it would reduce the diffraction around it ...

also even though a dipole has a rear wave it has no side radiation so as long as the rear wave could be tamed maybe reflections inside the horn wouldn't be so bad.

for example you could even stuff the horn with wool so that the low frequencies can come out of the horn but the HF back wave from the planar is absorbed instead of being reflected ...

again, for the time being i am not suggesting this would make a practical speaker but rather this is a thought exercise - the question is can it be done and would it work ?

and yes i am aware BMS is famous for their COAXIALS so you could just get one of their 1.4" coaxials on the same horn but IMHO those are also not without compromise. it is not theoretically possible to perfectly combine the output of the two BMS diaphragms. the way they combine is merely GOOD ENOUGH not perfect.

the question then is can my proposed planar / horn coax combine the two outputs WELL ENOUGH too ?

the problem with BMS and B&C coax design is that it is immutable - whereas my concept can be CUSTOMIZED for different applications ...

you may not see the value in crossing over a midrange compression horn to planar and that's understandable in which case just imagine that instead of a horn it's a woofer. and you are placing a dipole planar in front of the woofer. has anybody tried that ?

obviously Radian has done it, but not with a dipole. but i think one benefit of dipole for example is that the diaphragm conductors would be cooled from both sides rather than sealed off with a chamber from behind thus increasing output capacity ...

another benefit is that bass from the woofer won't push the dipole diaphragm into the chamber but put similar pressure on it from both sides the force thus mostly canceling out so that the diaphragm isn't as heavily modulated by the bass ( hopefully )

i already mentioned that diffraction would be reduced ...

what do you think ?

Simple amp to build

Ok, So I am looking around for a simple amp project as a fun build.
And I came across this page https://sound-au.com/project12.htm
And it got me thinking.
The last version on that page could be nice I think. Easy to stabilize and enough loop gain for decent THD specs.
Why is this simple input arrangement not used more often?
I always see the LTP in VFB amps or the basics of a diamond buffer as used on CFB amps.
But why not a single transistor?
The only downside is that it can't be DC coupled.
Using more modern output transistors could make this amp more then fast enough also.

Just some thinking.

Any ideas to improve it further are welcome.
Also any other simple amp projects are welcome.

How to find a replacement/equivalent rectifier diode?

I have a rectifier diode (I think thats what its called?) That I want to replace because it has broken legs, but I cant find the exact model, and am wondering what do I need to know to find a replacement/equivalent?

The diode is a FMU34S, Ill add a photo of it as well.

The diode tests good, at least from what I can tell, but I would like to replace it if possible.

I would like to replace its twin also, a FMU34R, if possible.

Can someone please let me know what I need to know to find a suitable replacement?

Any help is appreciated!



IMG_1336.jpeg

Slatted wood grille speakers - Retrofitting - How to select new full-range drivers?

Hello everyone, I'm starting my first DIY audio project and I have a few questions about the best approach.

I recently picked up two wooden speakers with a slatted grill from a thrift store ("kringloopwinkel") in Amsterdam. I reckon the speakers are from the 1960-1970s and were a DIY build - perhaps the housing was sold as a kit (I believe that was common at the time).

Now this pair of speakers contain two different drivers, one seemingly made by Philips and the other one by Isophon.
I'm thinking about replacing these with modern, full-range drivers that can work with the enclosure. Currently, using two different drivers won't give the best output. But I'm not so sure how to find a possible match for the housing.
My initial idea is to replace the drivers, and install a Raspberry Pi with HiFiBerry AMP2 in one speaker (the speaker which is slightly larger) to make it a TIDAL Connect speaker (a speaker pair that won't break the bank). A friend of mine who is a luthier can help me to repair the damaged wood veneer.

Both speakers have 8 openings/gaps, of which the top and bottom ones are slightly smaller.
The inner dimensions of the housing are as follows.

Speaker 1
230mm x 210mm x 125mm
6.04L
0.213 cubic feet

Speaker 2
230mm x 250mm x 125mm
7.19L
0.254 cubic feet

CAD drawing of front of speakers

vintage-1960s-speaker.png


Front of speakers photo

IMG_0575.jpg


Speaker 1 photos

IMG_0570.jpg

IMG_0572.jpg


Speaker 2 photo

IMG_0578.jpg


Isolation photo

The isolation probably needs replacing.

IMG_0576.jpg


Question about selecting full-range driver
  • How do I select the full-range driver that works with the enclosure?
    • Note that I can make the larger speaker the same inner size by compartmentalizing the Raspberry Pi + HiFiBerry amplifier in a 40mm space.
    • So far I have tried fiddling with WinISD, but I'm just not sure how to take into account the slatted grill/vent situation when modeling.
    • PartsExpress also notes the optimal vented volume. Is it enough to select a driver by matching their vented volume with mine for +/- 10% ?
  • What is the best way to mount the full-range driver?
    • Should I mount the driver in the same way as the current ones, or should I ensure that the speaker covers the six larger middle vents while keeping the top and bottom vents open for bass? If so, this would probably require a 4-inch full-range driver.
Thank you for your time!

The Effects of Damping a Ported Cabinet

I've just lined my desktop speakers with thin egg-box foam. I admit the speakers sound less boxy. There's less bass in the 100-200Hz region. But I remain sceptical. Doesn't the foam simply reduce cabinet volume?

It's confusing. I've taken apart hundreds of speakers. Some manufactures stuff their cabinets with damping and wadding like the they bought it in a fire sale, others use none.

First build, any feedback or questions

Hello Diy audio,

Welcome to my first build, 7 woofers, one horn loaded titanium driver. Rms 600w (but I will never get up to that), db level that is much to loud for my apartment. Some may question this design choices, roast and question away.

About 90% done, just wiring, paint and grills to go. Budget was low but dreams were high. They sound pretty great, with rew, I can eq in a perfect flat line, and phase response is clean. Woofers are suposably somethingb like 98db sensitivity
(6pt-8). One day they will live outside where they have space to breathe.

Thank you to the help diy audio has given me aswell the kind people who responded to my interestingly written questions. This place is my training guide of the ins and outside of speakers.
17422556341748544785647224316926.jpg
1742255672053192466662561607692.jpg
17422557102153952747972011841559.jpg

Change Op-Amp / Caps in CD player

Good Morning, I have a old Luxman D-355 that I like to restore / fresh up.

It have two Op-Amps after DAC chip ( analog out ). The units is NJM2100D ( dual ) from JRC, is it worth change them?

I also thinking of re-cap, the unit is from 1995 so 30 years old. But I like the way it present the music with the 1 bit DAC.

I think of good all around caps for the board except maybe some audio caps for the analog part.

Any thoughts ?

Frank

Taramps DS 1200X4 turns on but sounds distorted

Hello, I have another taramps on my bench. 2 of the traces were burned away I had to repair them, crudely, the 2-10 ohm resistors in the +/- 5 volts circuit, they were reading in the kilo ohms.

I've managed to solder on the IRS 2093 IC, all the relevant voltages are there. VCC 1-2 is 14.2 volts, plus both +/-5 volts and 10.1 volts on the CSD pin.

But with all that the audio sounds distorted so a checked the drive voltages and they're are like 1 volt or .4 volts or even 2 volts.

Attachments

  • IMG_20250312_034729.jpg
    IMG_20250312_034729.jpg
    519.3 KB · Views: 47
  • IMG_20250312_034949.jpg
    IMG_20250312_034949.jpg
    355.4 KB · Views: 44
  • IMG_20250314_160931.jpg
    IMG_20250314_160931.jpg
    466.1 KB · Views: 42
  • IMG_20250314_160956.jpg
    IMG_20250314_160956.jpg
    426.9 KB · Views: 50
  • IMG_20250314_161051.jpg
    IMG_20250314_161051.jpg
    381.8 KB · Views: 42
  • IMG_20250314_161105.jpg
    IMG_20250314_161105.jpg
    238.2 KB · Views: 48
  • IMG_20250314_161118.jpg
    IMG_20250314_161118.jpg
    309.1 KB · Views: 48
  • IMG_20250314_161159.jpg
    IMG_20250314_161159.jpg
    577.9 KB · Views: 46

Auva footers

Recently I installed Auva 50 footers under my speakers. Auva's are products from Stack Audio in UK and only sold dealer direct online. I installed these after a personal recommendation from an audio friend whose recommendations are always spot on. The Stack audio website is full of information on what the products supposedly do, along with numerous reviews and customer comments. Reviews and comments are always on the positive side, otherwise they would not be on the site. The product information is extensive. But then it is about selling a product. They do come with a 30-day money back, so it is a safe purchase. I made the purchase and installed the Auva's.
Once installed I did a slight speaker positioning adjustment, and I started listening.
It did not take too long a time to tell that something was vastly different in the music. And the vast difference is that the Auva's seemed to remove the electronic signature of the audio system. The music came alive and brought a "You are there" sound and feel to the music. This was across the board with every disc that I have played, no exceptions. Of course, the musicians are not there in the room with you quite obviously. But if you close your eyes to remove visual stimuli, the music does sound like the real natural sound of musical instruments and human voices, with nothing really added. I listen to classical and jazz, with some folk music at times. I tend to not listen to electronic popular music very much at all.

The Auva's are the real deal! They are highly recommended. I have found them to be the biggest sound improvement that I have ever made in all my many years of audio listening.

My speakers are in the pic below. They are DIY, designed in 2003 by the late Rick Craig of Selah Audio, and remade in 2018. They are simple DIY speakers nothing too fancy. They are a 2 1/2-way design with Seas W18E and Hiquphon OW,1. They are reasonably similar to Troel's CNO25, though my design predate Troel's by a long time. The spearkers are driven by a Starkrimson Ultra amplifier. All music is DSD64, DSD128 or DSD256 supplied by the Marantz SA KI-Ruby player.

I should add the following disclaimer:
I fully understand that my observations are worthless anecdotal observations and are completely unsupported by any graph, measurement, or any such thing. I only posted because perhaps there are a few odd individual listeners who might read this and investigate for themselves.

Attachments

  • Speaker 1.JPG
    Speaker 1.JPG
    460.3 KB · Views: 88

IRS2092S Pin 2 GND

Working with a IRS2092S based amp and I want to modify it to accept non inverting signals. Pin 2 is marked as the + input and is normally grounded. Can I lift the ground with say a 100 ohm resister to ground and then use pin 2 as a non inverting input or am I missing something. It does state in the datasheet that the COMP and protect circuit reference the ground from pin 2.

Doing the unthinkable. Adding a series resistor to a transformer primary to lower the voltage

A few days ago I suddenly noticed 'something' was humming noticeably in my set up and that something turned out to be my Sony MDS-JE480 Minidisc recorder which was in use at the time. Switch to standby and the noise vanished. Hmmm, well hummmm actually and from the transformer. Examination of the PSU shows that it uses two transformers, one for standby and one to power the main unit. Kind of unbelievable really for a budget model although it was and is ranked as one of the very best sounding of all due to its use of the very last generation of ATRAC which is the data compression system.

For curiosity I measured the mains and it is was high(ish) at 247 to 248 volt. Was that playing a part I wondered. The mains here is often in the 242 volt region but this was right at the top end. We also have something that is a kind of active voltage control system in this area where mains voltage is dynamically controlled and optimised in real time according to overall demand. It obviously has to keep within legal limits but the idea is to reduce voltage when demand is high to reduce loading on the grid which is sort of counter intuitive when you think of all the SMPS as lowering line voltage will increase current draw... however...

Back to the Minidisc. I wondered if doing the unthinkable would be possible... adding a series dropper resistor to the mains transformer.

The power supply is shown here. You can see the little standby supply at the top of the schematic and the larger (but in reality still pretty small) main transformer at the bottom. The PCB layout lends itself well to trying this as there are two wire links (arrowed) are fitted in the feed to the larger transformer primary. I removed these wire links and fitted two 56 ohm flameproof resistors in their place.

PSU 1.jpg


Results seem really good. Operation is now silent and the two resistors drop around 3 volts each and run essentially cold. The drop seems very constant and does not change under any operating conditions such as when the mech is loading. So regulation is essentially unaffected.

I wouldn't normally recommend anything like this but in this instance it seems to be a viable and workable solution. The PSU regulated supplies have a lot of headroom and even at lower mains voltage near the lower legal limit (not that we ever come close to that here) there should not be an issue.

Totally unrelated to this and something I only noticed with studying the PSU diagram is that the analogue opamp stages are fed from a 100% unregulated supply (arrowed). How audiophile is that? Perhaps very! because it certainly works extremely well on these budget machines and which as mentioned are reckoned to be one of the best sounding.

PSU 2.jpg


Also of interest is the 0.0022uF cap shown across the relay contacts in the PSU schematic. I measured an AC voltage of around 17 volts AC across the primary in standby, quite high really and due to the reactance of the cap. Is that of consequence? Maybe it plays a part in muting and silent power on and off as it means some voltage will be present on the rails (particularly the unregulated ones) even in standby. This is one of those times where you have to think that the designers really knew what they were doing.

The analogue stages fed from the unregulated supply:

PSU 3.png

Radian 745NeoBe or 950NeoBe? And Recommended CD Horn?

How might the midrange and HF band presentation change between these 1.4 and 2" models?


https://www.usspeaker.com/radian 745neoBepb-1.htm
https://www.usspeaker.com/radian 950Bepb-1.htm

Alternates https://www.usspeaker.com/radian 951Bepb-1.htm
https://www.usspeaker.com/radian 760neoBepb-1.htm


Which would you choose for what kind of two-way (only) speaker build and why?

Also, please recommend the best constant directivity horn you would suggest for any of those drivers.

Any experience with these?

https://alg-audiodesign.com/pavillons/
https://audiohorn.net/x-shape-horn/
https://audiohorn.net/next-gen-bi-radial-horn/
https://www.usspeaker.com/ciare pr614-1.htm

Help!! 6AQ5 SE amp

1741411681769.jpg
1741411610028.jpg

I want to add this tone control on this circuit.
1742229111640-2.jpg

Can I do it this way?( is the amplification rate too low?)
And I want to add direct function(don't pass the tone control circuit) i know it is unnecessary on this circuit( when the knob is on the middle, it works like direct function) it is for experimental reason, I want to know the effect of the capacitors of the tone control circuit on sound quality. So can I just add the switch on the input(like upper schematic) or should I add one more switch before the grid input(because i thought 12ax7 on the tone control could cause noise)(like lower schematic)
1742229111640-1.jpg

I am sorry for poor English.... can anyone help with these problem?

Which heatsink for a chipamp.com LM3886

I am going to build an LM3886 chipamp on chipamp.com boards. About 30-34VDC on the rails (22vac tranny) with an 8 ohm speaker load.

I have 2 each of these heatsinks but no way to calculate if they are adequate. I will make the enclosure and can make it to fit.

The small heatsinks are 45mm x 50mm x 20mm. One LM3886 on each heatsink.
The large heatsinks are 270mm x 100mm x 50mm. Maybe put both chips on one heatsink?

Do you think the smaller heatsinks will work? If not would both chips on one large heatsink be sufficient?

I know this is somewhat of a guess without know the characteristics of the heatsinks but it is what I have.

Thanks

Small heatsinks:
20250110_143240.jpg


Large
20250110_144732.jpg

Small 3D printed unity horn

For a new topcabinet, I needed a MF/HF (or only HF) section, which is able to handle a crossover around 700Hz. The whole design revolves around maximum output for a given size. The dual 10" LF section is loaded by a short offset driver horn and ported rear chamber. For this to work, I knew from the beginning, that a relative low crossover was necessary (approximately 700-800Hz as an estimate). In total the following demands are set:

-Low XO possibility (700-800Hz)
-Maximum hight and width 29x29cm total
-+- 80 degrees horizontal dispersion
-50/60 degrees vertical dispersion
-Rotatable horn for possible horizontal arraying

This all could be solved by going for a coaxial CD or large 4" VC and for example a RCF HF950 horn. But there is no fun in that.
So I wanted to try a different (and unknown for me) path for this project. After some minor research I decided to just start with some design work and see what can or will come out.


This gave the following:
KyQiV05.jpeg

TTcZ67e.jpeg


The MF drivers are two 4ND34-16 from B&C, and a new RCF NDX595.
The whole horn is for now printed out of 5 parts total due to the size of my current printer. In the end its the idea to bring it down to two and outsource the frontal part.

As I wanted to be flexible with some testing, I made the MF taps removable. This way i can test various placements, sizes and shapes.
By doing so, I am a bit limted in vertical placement eventually though, but priting that whole parts again takes a lot of time and money if I have to do that several times.

As I wanted to try it ASAP, I made some preliminary inside measurements with a calibrated UMIK-1. Its not ideaal, but gives a first impression.
Distance mic to SUT +- 40cm.

2x4NDF34-16 parallel, taps a close as possible with insert, two 22mm round holes (total area 7,6cm^2 per 4NDF34) + NDX595:
bWmWm0j.jpeg

With this measurement, the back of both 4NDF34's was open.

I tried my best by closing the back of the 4NDF34's with gaffa tape and got the following results:
iC56yFU.jpeg

I will be printing some better parts for the drivers to get it better and neatly, I am quite sure its leaking currently.

The 1kHz dip of the NDX595 is due to the current tap design an placement. When covering with some pieces of tape:
Z9LUaZY.jpeg




To try and reduce the dip at 1kHz, I did a quick test with a MF tap inserts which has several small holes compared to the two 22mm diamter ones per driver.
70h1tNs.jpeg

Response evened out quite a bit, but less high end extension from the 4NDF34's:
YIA6bRQ.jpeg



Currently next things I want to do:
-New insert with angled taps (facing away from CD)
-Resolve some resonating parts I still have
-Make a turning table for some off-axis measurements
-Back covers for 4NDF34
-Measure NDX595 on the PH220 I have for some sort of reference
-Get my other measurement mic calibrated again to get some measurements with phase response. This will be crucial to eventually get a nice crossover and if going passive MF tap placement is key from the beginning.
-Do some impedance measurements
-Get a whole day to do various measurements and outside or larger room.
-In the end the HF exit integration can be smoothed out/be better

Its still a work in progress and the outcome could still be that its not viable in the end or does not meet the set criteria. But we will see 🙂

Purifi / Satori, Bench Top 2.1 Speakers

Hello All,

I took the plunge,

Today I received from Madisound a pair of Purifi 6 – ½ inch Aluminum mid-woofers and a pair of Satori Textreme 29MM tweeters.

I also have on hand a pair of Satori Beryllium tweeters to audition.

See the attached links to the data sheets below.

This is planned to be two-way Bench-top speakers with a summed subwoofer tucked on the shelf below.

Look at the Purifi data sheet and you will notice that below ~ 125Hz that the Harmonic Distortion takes a steep climb. The crossover to the sub will remove the long excursion Low Frequencies from the Purify mid-bass driver. With the LF removed there is expected to be reduced intermodulation with the higher frequency voice content of the program material.

Purifi PTT6.5X04-NAA-08 6.5"

Satori TW29TXN-B-4 / TeXtreme Dome Tweeter

https://www.madisoundspeakerstore.c...tori-tw29txn-b-4/textreme-dome-tweeter-4-ohm/

https://www.madisoundspeakerstore.c...tw29bn-b-beryllium-dome-tweeter-black-flange/
https://purifi-audio.com/document/share/16/6b6d90f3-b9db-4eaa-9f36-9d424bd2adba

To start the crossover will be a active Rane Mojo MX23 that Is already on the shelf. There is a switchable factory summed sub-woofer output. The tweeters and mid-woofers will be driven by the XLR “HF” and “mid” outputs on the back side of the active crossover. The crossover frequencies are adjustable.

The Purifi site has a discussion about using series notch filters with the Purifi driver to notch out the 5kHz and ~ 9.5kHz resonate peaks from the Frequency Response. This is worth an effort. I think

A approximate 6 Liter sealed enclosure will used to start the Proto process.

Thanks DT
Projects by fanatics, for fanatics
Get answers and advice for everyone wanting to learn the art of audio.
Join the Community
508,471
Members
7,921,112
Messages

Filter

Forum Statistics

Threads
407,839
Messages
7,921,112
Members
508,471
Latest member
Delrtm