FIR-LADSPA: A LADSPA plugin for FIR filtering

Several weeks ago I started to write code for implementing FIR filters under LADSPA, and mentioned it HERE.

LADSPA plugins were never intended to do intensive processing, or to have highly variable CPU loads between calls to the plugin by the LADSPA host. This means that implementing an FIR convolution inside the plugin itself is probably not advisable, or even possible. So I have taken a different approach:

The FIR convolution is performed in a separate process that is entirely independent of, and is running asynchronously from, the LADSPA plugin and host. I call it an "FIR engine". This separate process is launched using an operating system call as part of the LADSPA plugin setup process. Parameters and other info are passed from the plugin to the FIR engine using a "control" FIFO, and other FIFOs are used to pass unprocessed and processed data.

Performing the FIR convolution in a separate process has several advantages. The LADSPA plugin, once up and running, only needs to put/get data into/out of the FIFOs and this keeps it very computationally lightweight. The FIR convolution can only be performed when the desired length of data has been obtained via LADSPA. Because of the real-time nature of the audio processing, this takes Ndata/sample_rate seconds. For example, at 48kHz if you process 16384 data points per call it will require the FIR engine to collect incoming data for about 0.35 seconds before it can perform one convolution. During that time, the LADSPA plugin might have been called 16 times (given a typical frame of 1024 samples). The speed requirement of the convolution can therefore be as slow as 0.3 seconds or so, by which time the next 16384 samples have been collected and need to be processed. This can be helpful when using low powered Linux hardware such as a Raspberry Pi. Additionally, the OS is free to schedule the FIR engine process around other processes, and there is no need to run it at a high priority, etc.

As part of the set of of the plugin and FIR engine, a uniquely named directory is created in the tmpfs (files in memory) that comes with all Linuxes. The FIR engine collects and writes to this directory a file that lists the mean and longest "cycle time", that is the time to process the data and return it to the LADSPA plugin. The cycle time latency can be obtained in a test run for a given platform and FIR filter set, and then this is supplied to the LADSPA plugin as a parameter during normal use. This latency timing info is used to set up internal buffers so that underruns are prevented.

Because the FIR engine is a separate and independent process and because some LADSPA hosts do not correctly tear down the plugin (by calling deactivate, etc.) I have written the FIR engine to self-terminate and cleanup after itself. This includes deleting the tmpfs directory in which it was operating, and the FIFOs. Once this process is completed there is no sign that the FIR engine was ever there. A pair of error logs are written to the tmpfs and not deleted, however, on reboot the tmpfs filespace is wiped clean. The user can manually delete the error log files anytime. Since these only are used to record fatal error messages, it is not likely they will grow to any appreciable size.

I am currently coding up the FIR convolution using FFTW but have everything else functioning well using a dummy convolution function that simply passes input to output in the FIR engine. I hope to get a test version fully up and running in a week or so and will post updates as I have them.

Beginners question: slightly other geometry

Hi

Apologies for yet another noobie-question, as I have no knowledge of all this but am curious…

I will build a 3w speaker from a kit in the near future. This will be my first ever.
The designer wasn’t too specific about the cabinet except for the front-baffle and the volume.
While contemplating various details, the question came up wether changing the cabinet‘s geometry (making it „conical and a little deeper) without changing the volume or front-baffle would affect the sound?

New AD1862 DAC chips and Miro's DAC PCB

I wanted to try Miro's AD1862 DAC. Apparently the only legitimate source fo thse chips is Digi-Key and I had to purchase 14 of the DAC chips. I've already sold some of these but do have 4 chips left. I paid $25 each for the chips and will sell them for the same amount. I also had 10 of Miro's fine PCBs made by JLCPCB. I will give these for free with the chips. Please send me a PM if interested.
Cheers.
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Denafrips ARES II board repair

I just got a spare Denafrips ARES II board. I bought it used ("in working condition") but I doubt it is the case. The initial plan was to use it in a DIY DAC perhaps with a lithium battery power supply. But first I need to power-up this board, repair it and test it. Can you help me with that ?

I am looking for information about the original transformer and how it is wire on the board (PIN 1 to 8) ?

bottom_ares II.jpg


Here is the top of the board:

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I already see two problems on the board :

input_psu_zoom.jpg
LDO.jpg

The two center PIN of the chip (red arrow) were connected.
Any clue of the chip reference ? I think it is a LDO fix the gate voltage the transistor.
On the LT1763 LDO seems defected (yellow, too much heat)

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Hiss from TPA3116 amp board that dies away when pot is at max???

Hi folks, this is a strange problem I've never come across before. I stuffed a YJ tpa3116 ( black/blue) board in an enclosure with a 2 way input selector switch and a 10k tocos cosmos pot. I've used this setup before without problems but this time I'm getting quite a loud hiss from both channels that isn't coming from a source and dies away to almost nothing when the pot approaches max (nothing playing) in the max position the hiss can only be heard with my ear to the speakers but at normal listening positions I hear the hiss from my chair across the room. The same pot worked fine previously with a buffer so I'm kind of confused why it's not ok with the the amp board. I have a 20k miniature stepped attenuator I could try but I can't see how that would make any difference. Any ideas would be good to hear as this is both annoying and confusing.


Cheers D

DBX 234 /XL 3way crossover

Hi,
I had to sell DBX 234 XL 100% working conditions,use only a few hour only ,
They are not no more using after brought fusion 503 .
Price $50 plus world wide shipping

Best regards
Myint

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Designing ES982x-based ADC - feedback wanted

After looking at all the existing options I can find for adding an audio ADC to a RPi, I've decided to go ahead with making (or at least, trying to make) a RPi hat based around the ES982x. It will be pretty similar to the datasheet circuit, using ESS's own ES9311 regulator to power the ADC chip. The ESS datasheets use OPA1612 op-amps for the input stage, and I allowed for them, however the first prototypes will use ES9820 + NE5532. The main design requirements are:

1. Single-ended connectors onboard, with headers for diff. input
2. Use I2S, not USB
3. Hopefully >100dB THD+N - significantly better than other options available
4. Parts available (not that easy)

It's really for people who want a high(er)-quality I2S input to their Pi, because there are high-quality USB ADCs (Cosmos) and ~90dB THD+N hats available (e.g. HiFiBerry ADC+DAC and this open source WM8731-based one). I also think that a Linux driver for the ESS chips could be useful for someone else.

There will be a few parts of this project that are new for me (especially the driver). Here is the first version of the schematic... I welcome feedback and suggestions. For example, one thing I'm not sure about is whether the resistor values can be left the same when substituting the NE5532 instead of OPA1612.

FYI, my own use-case is to provide analog input for a RPi DSP+xover for 3-way active speakers. Analog input selection is done on a relay board inside the case and the RPi USB ports will be exposed through the back panel, so a USB interface is not gonna work.

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Ultimate Mastering Monitor Speakers

My background is in music production so I know very well all of the commercially available mixing / mastering monitors. I want to explore building my own.

From the monitors I have used I would like to take the general characteristics of:

- the highend from Genelec S30D ribbon tweeters

- the midrange from ATC scm200 soft domes

- the Lowend from Barefoot MM27 but with extended response

For my purpose I would like to be able to go to fairly high SPL's and also frequency linearity is crucial. Budget wise I want to be uncompromising but not needing to go into extreme esoteric territory.

1. What is the most suitable Ribbon Tweeter? (Raal 140-15D / Founteck NeoPro10i / others?)

2. What is the closest available driver to the midrange dome of the scm200's? (Volt VM752 / Morel SCM634 / others?)

3. Not sure what LF drivers would complement the above but would like 2 x 15" drivers in a side firing configuration (a la barefoot MM27) (Volt RV3863 / others?)

Would appreciate thoughts on this configuration with professional audio mastering in mind.

Currently designing a new studio so there are no space limitations - the studio will be big enough to accomodate any design.

Current thinking is to use Hypex amps / power supplies / DSP. But also open to other suitable pairings for the above.

Most likely closed cabinet and not ported design but this also not finalised.

Thanks in advance for any advice for this novice !

Re building Metaxas Solitaire Amplifier

I have a pair of Metaxas Solitaire Amplifiers that drive my Infinity Kappa 9 speakers
The amps were sent to a reputable Perth electronic service outfit ,but were deemed unserviceable by the technician.
The amps in question were revealed on the techs' Hall of Shame
https://liquidaudio.com.au/hall-of-shame/
So I have decided to rebuild myself , but the question is - Is it a Bridge too Far for a novice to undertake?
I have plenty of time at my disposal as I'm retired .
Any comments would be gratefully received.
Thank You.
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Digital level meter for R-R tape deck

The analog level meter has been designed for use as a recording level meter for a reel to reel tape recorder. But it can also be used as a standalone device. The LED bar can contain up to 50 elements per channel. To reduce the level of interference, static indication is used. A feature of the meter is fully digital signal processing. That allows you to make the meter metrologically accurate, with an exact match of the dynamic characteristics. All parameters are programmed using special software on a computer.

Each meter channel contains two signal processing branches, the readings of the first one are displayed as a bar, the second - as a dot. This can be used to display two values, such as average + quasi-peak, or quasi-peak + true peak. For each of the branches, you can individually set the gain, integration time, response time, decay time, hold time, fall time, peak hold time, peak fall time. The scale type is set by assigning an individual threshold level to each LED with an accuracy of 0.01 dB. This allows you to build a linear, logarithmic, S-shaped with a stretch near 0 dB and other scales. A set of all parameters, including the scale type, is stored in EEPROM as a preset. There can be 4 different presets in total. They can be selected with the control button. The second button can turn on the indication of statistics of maximum levels, as well as reset it.

Here we will talk about how simple things can be made difficult 🙂

Block diagram

The block diagram of the level meter is shown in the figure below. This diagram shows one stereo channel of the meter. The input signal is fed to a differential amplifier implemented on an op-amp (Meter50_v4_sch.pdf). It performs two tasks: it shifts the signal by half the ADC scale and allows you to implement a balanced input. The signal shift is necessary because the built-in ADC is unipolar, its input range is from 0 to the reference voltage. A balanced input is desirable even if the signal source is single ended. It allows you to take the signal relative to "analogue" ground and eliminate the interference present on the ground of the meter circuit. If the level meter is made as a separate device, then balanced inputs with XLR connectors can be implemented.

block_diagram.gif


All signal processing is done digitally by the STM32F100/103 microcontroller. To avoid the use of an anti-alias filter at the ADC input, as well as to be able to register short signal peaks, the sampling frequency was chosen quite high - 96 kHz. With DMA, ADC samples are stored in a buffer in RAM. When half of the array is ready, an interrupt occurs, the data is read and processed.

DC removal filter

The first operation on digital data is to remove the DC component from the signal. This must be done before the signal is sent to the detector. With a meter dynamic range of about 60 dB, even such a small DC component as 0.1% of the ADC scale can be equal in magnitude to the useful signal and distort it.

The simplest filter that suppresses the DC component is the differentiator. It has one zero at zero frequency. The DC gain is zero (as required), and as the frequency increases, the gain increases. For a discrete implementation of the differentiator, its difference equation has the following form:

y( n ) = x( n ) – x(n-1)

In fact, this is the simplest FIR filter. When implementing, it must be taken into account that the output value must have a width of at least 1 bit more than the input value. Since the input signal in theory in one cycle can change to the full scale. Increasing the gain with increasing frequency is not satisfactory, since the gain will change in the operating frequency range. Instead, you need to get a linear frequency response above a certain cutoff frequency. Obviously, this requires adding a pole at the desired cutoff frequency. This can be done by connecting the differentiator and the 1st order low-pass filter in series. The overall frequency response will have the desired form. Such a 1st-order low-pass filter in such applications is often called a "leaky integrator". Because in the analog version, such an integrator is implemented by adding a resistor in parallel with the capacitance, through which it will slowly discharge. For a discrete implementation, the difference equation of an ideal integrator is given below:

y( n ) = y(n-1) + x( n )

To add a leakage, you need to multiply the accumulated value by some coefficient less than one:

y( n ) = A*y(n-1) + x( n ), 0 < A < 1

This is the simplest IIR low pass filter. The name "leaky integrator" is used to indicate that this filter is applied in a somewhat unusual way - the operating frequency band lies in the stopband.

The cutoff frequency here should be chosen low compared to the sampling rate, which means that the A coefficient should be close to unity. The cutoff frequency can be calculated using the equation:

f = (1 - A)*Fs/2*pi, or A = 1 – 2*pi*f/Fs, where Fs is the sampling frequency.

The general difference equation for the filter will look like this:

y( n ) = x( n ) – x(n-1) + A*y(n-1)

In fact, this is a first-order IIR high-pass filter. It is equivalent to a differentiating RC chain. Such a filter has one zero and one pole. If you show them on the z-plane, then zero will lie at the point z=1, and a little to the left of it on the axis there will be a pole.

The audio bandwidth starts at 20 Hz. But it is desirable to make the cutoff frequency lower so that a noticeable phase shift does not appear in the audio band. This filter is not phase-linear, so signal components of different frequencies will be delayed for different times. The result of their addition will give a different waveform, the peak level will be incorrect. There are phase-linear FIR high-pass filters, but they require a lot of computing resources to obtain good linearity of the frequency response in the passband at a low cutoff frequency. There is a filter option for removing DC component based on MAF (moving average filter). It doesn't use much computational resources, but it does require a lot of memory to get a high sample rate to cutoff frequency ratio.

In this case, the requirements for the filter are not very strict. It is necessary to remove the DC voltage, which can only change very slowly (for example, due to the temperature drift of the op-amp). A significant part of the compensated bias is generally constant, as it is associated with the error of the bias voltage divider. Therefore, you can simply make the cutoff frequency lower, while minimizing the phase shift. A cutoff frequency of about 5 Hz would be a good choice. The phase shift at a frequency of 20 Hz will be about 17 degrees, which can be considered acceptable.

When implementing a filter in integer arithmetic, one must take care of the ranges of numbers at all stages of the calculation so that overflow does not occur. The leaky integrator and differentiator can be connected in any order. But to avoid overflow, the differentiator should be included first.

Another problem, more complex, is related to rounding errors. For a filter cutoff frequency of 5 Hz at a sampling rate of 96 kHz, the value of the coefficient A = 0.999673. To carry out calculations with such a coefficient with good accuracy, it is required to increase the bit depth. The reverse transition will be a quantization with a larger step, which leads to the appearance of an error, or quantization noise. As a result of such an error in the filter, a parasitic constant component may appear at the output, which is even greater than the one that is suppressed. This error may cause the filter to fail. As a practical test has shown, the filter does not actually work without requantization error correction.

In this case, the ADC is 12-bit, the samples are placed in 16-bit signed integers. Some margin is needed here to protect against overflow. Multiplication is done in 32-bit format. Then, when moving from a 32-bit intermediate result to a 16-bit result, a quantization error will occur. In one of the publications it is proposed to add a quantization noise spectrum shaper to the filter by introducing error feedback. The implementation of this method is very simple. I made my own implementation which uses the same error correction principle.

static const double POLE = 0.999673;
static const int32_t A = (int32_t)(UINT16_MAX * POLE);
static int32_t acc = 0;
static int16_t xx = 0;
static int16_t yy = 0;

x = Input;
acc = LO_W(acc) + A * yy;
yy = x - xx + HI_W(acc);
xx = x;
Output = yy;

Quantization occurs by discarding the lower half of the 32-bit number. The discarded value is the quantization error. The original number is signed, but the error is always negative. For example, if the least significant bits of a positive number are replaced with zeros, the number will decrease, and some positive number will have to be added to correct the error. If the least significant bits of a negative number are replaced by zeros, the number will become large in absolute value, while remaining negative. This means that in order to correct the error, you will again need to add some positive number. Therefore, the error can be extracted from the original 32-bit number by simply zeroing its high half along with the sign bit.

Detector

Next, a full-wave rectification of the signal is performed. This is the easiest part of processing. When the DC component is removed from the signal, the task is to calculate the absolute value.

Filters

Next are several filters that define the dynamic characteristics of the meter.

The main characteristics of level meters are integration time and return time. The integration time determines how quickly the meter reacts to changes in signal level. If the integration time is large (about 300 ms), the so-called VU-meter is obtained. Such a meter will not respond to short signal peaks that can overload the recording path. When pointer instruments were used as the indication device, the integration time could not be made small due to the inertia of the moving system. Therefore, often the VU-meter was combined with a faster LED peak indicator. The use of high-speed information display devices, such as gas discharge tubes or LEDs, removed the problem of obtaining short integration times and made it possible to build peak or quasi-peak meters.

Most often, quasi-peak level meters are used to measure the signal level in audio paths. Unlike true peak meters, which in theory have zero integration time, for quasi-peak meters this time is defined by the standards. It would seem that it is necessary to achieve the minimum possible integration time so that the indicator can register the shortest signal peaks without overloading the path. This approach is used in digital paths, where even a short-term overload leads to undesirable consequences. For analog magnetic recording, a short overload may not be audible at all. If you want to completely eliminate the overload at the peaks of the signal, you will have to reduce the average recording level, which will lead to a more audible problem - a decrease in the signal-to-noise ratio. Therefore, for analog magnetic recording, it makes sense to choose some optimal value for the integration time of the level meter so that it allows short-term overloads.

The integration time for quasi-peak meters is defined by IEC 60268-10 as "...the duration of a burst of sinusoidal voltage of 5000 Hz at reference level which results in an indication 2 dB below reference indication". This standard defines an integration time value of 5 ms for quasi-peak meters.

The integration time is not numerically equal to the charging time constant of the smoothing RC circuit. If an RC circuit with a charge time constant of 5 ms is installed at the output of the peak detector, then the level of -2 dB (about 0.8) will be reached in about 20 ms. To reach -2dB in 5ms, the charging time constant of the RC circuit would need to be around 1.25ms. Thus, the integration time is approximately 4 tau RC circuit.

In addition to the integration time, quasi-peak meters have a return time that is much longer. This time determines how fast the reading will fall after the signal is removed. If this time is made small (for example, equal to the integration time), then the indicator readings will change too quickly, making them difficult to read. For VU meters, such a problem did not arise due to their low speed, one time constant could be dispensed with. In high-speed quasi-peak meters, it is necessary to artificially slow down the fall in readings so that the operator can read the information.

The return time determines how quickly the meter reading will fall by 20 dB after the signal is removed. It is also not equal to the discharge time constant of the RC circuit, but is approximately 2.3 tau. The return time value is also set by the standards. It differs for indicators of different purposes. For indicators of the first type, which serve to check the signal level during its operational adjustment (this is just the case of adjusting the recording level in a tape recorder), the return time should be 1.7±0.3 sec. Accordingly, the discharge time constant of the RC circuit should be approximately 740 ms.

Level meters have another dynamic characteristic - response time. It may not be entirely clear. For inertialess devices, ballistics is artificially formed, similar to pointer devices. Increasing the response time in this case should not increase the integration time of the meter. The short signal pulses should still display correctly. To do this, the detector must "wait" until the reading reaches the measured value, and only then start the fall. In amateur designs of quasi-peak meters, with a fast increase in the signal level, the bar immediately jumps from one length to another. This is not the case with professional quasi-peak meters, because the increase in response time has a positive effect on the perception of readings by the operator. In VU meters, this is also not the case, there the smoothness of movement is obtained due to the large integration time. The response time is defined as the interval from the moment the test tone burst begins until the reading reaches -1 dB.

Integration time

To obtain a given integration time, the detector output must be smoothed with a filter. The smoothing filter is not a simple low-pass filter, but a more complex filter with different charge and discharge time constants. The charge must be fast to provide the specified meter integration time. And the discharge must occur slowly to provide a given return time, which is much longer. Therefore, the integration time filter is turned on only when the signal rises; when the signal decreases, it is kept at the same level. A separate filter is used for signal decay.

The difference filter equation has the following form:

y( n ) = y(n-1) + A*(x( n ) - y(n-1)), where A = 0.0083 for Fs = 96 kHz

A 32-bit accumulator is used to obtain the required calculation accuracy. Filter coefficients are 16-bit unsigned. They are defined as A = 65536 / 96000 / tau.

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Are you really fine with IC voltage regulators ?

Hi ! looking for some information about dual voltage power supplies for preamps in an interview with the Conrad Johnson designer he states that they prefer to realize voltage regulators with discrete parts and not using single chip voltage regulators. In their opinion they can have an impact on the sound ... and not for the good.
Looking then at some preamps schematic i see that in the late '70s and early '80s many of the TOTL solid state preamps actually used voltage regulators made out of discrete parts.
I guess the reason was that this IC regulators were not available ? 🙄
They are clearly handy ... but are they also completely fine to use in preamps ?
i would like to get some opinions ... i have a feeling that i was looking at the wrong part of the story ... the preamp circuit ... instead the secret of good sound could lie in the power supply design and construction 😕
I am attaching one schematic just to explain what i have in mind ... more or less

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Why no flat piston subs?

Except for the KEF B139 (and discounting the few odd Japanese novelty offerings) no piston-type drivers seem to have ever caught on, and apart from expense - which shouldn't be excessive anyway - are there any reasons not to make 'cones' this way for LF applications, and any reasons they should be? I could see a Kevlar/Nomex honeycomb/Kevlar piston as having some merit.

Rotel RD-30F tape deck rewind/ff not engaging

Hello!

I just picked up a Rotel RD-30f and it’s need of some repairs and could use some help pointing in the right direction.

First up is the eject mechanism is super sluggish and almost doesn’t open on its own. I have a video of that here:
Video

Second is the rewind/fast forward doesn’t work. I press down to engage it but then it immediately stops but the button remains pressed down. If I hold down the button more it will run.

Dali Concept 8 woofer configuration

Good evening all,

I got a pair of these Dalis as part of a recent rubbish collection, ie for free.

First thing I noticed was there are 2 ports, front and back, for the 2 woofers and reading online seemed to suggest this Dali series used separate enclosures for each woofer.

However removing the woofers showed that there is only one woofer enclosure with 2 ports. Why would Dali do this? A first inspection shows the 2 woofers are wired in parallel. Is it just for looks?

Cheers, sp

PPI 2300AM

What would cause this on Q16 ?

Is it from stress or the legs being bent and then bent back ?

Everything appears to be factory ampnhas never been repaired as far as I can tell .

The client bought the amp off someone and wanted it checked over before install .

I cleaned the wire strands from the board that I found so far

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Measuring bias (Cyrus amp)

From the diagram below, there appears to be a connector that allows for bias to be measured.

Am I correct that to do that I need to remove the jumper (I will check the board tomorrow morning) and connect an DVM between those point (set to amps) to measure the current?
The VR allows the bias current to be set - can anyone tell me what this should be?

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PeeCeeBee V4 GB!

Hello guys!

A fresh new PeeCeeBee is here, the V4! In this thread group-buys for the printed circuit boards are arranged.

It's a Lateral MOSFET based Class-AB amplifier with symmetrical "current feedback", evolved from from the diverse PeeCeeBee series, and now with excellent sonics coupled with excellent looks. 🙂

The new design is electrically and sonically a step forward from the "stubbornly" simple but very good sounding designs released in the original PeeCeeBee thread. I say a step forward partly because it incorporates some extra well-known techniques to handle some of the problem areas in the old designs and allows a quicker and safer mode of set-up and operation, but mostly because while doing so it didn't become a fatigue-inducing clinical sounding amplifier. The layout is designed by myself and the amplifier is very stable, both thermally and electrically. The amplifier has similar sonic signature to the previous peeceebees in the mid and high frequency band, but the low-frequency performance has been substantially improved so that it can be used as a full-range amplifier with effortless bass as well as beautiful midrange and highs.

It uses a simple but effective diode-clamp based soft-clipping so that occasional overloads don't bother listening enjoyment. The two jumpers, when open, remove the chance of overloading Q9/Q10 while setting-up. The constant current sources keep the input/VAS bias much less affected by power supply variation during loud playback and/or mains fluctuation.

The following two screenshots show the design:

NmP6djgiQemEEJ6ulRHneov_72vVjiTZ9m4BI3cpO2QfjeALR5SlxKm8w0Gi1XCOE4CK5j0wXDgvFAs=w640

79M-Sjnh5trf3F_kybX_kqhk2f0YDxv8R5qIQNxPcSLeHsp_RCwc-LA8Y6ekX462vT1igjCUtzWiah4=w640

This is how the PCB looks in real world:

A fully assembled channel:


Some specs (measured with generic components/passives and basic unregulated linear PSU):
Maximum Power - 50WRMS/90WRMS into 8R/4R load with +/-35V PSU
Maximum allowed PSU voltage for 4R operation - +/-35V (with default clamp diodes and BD BJTs for Q9/Q10) and +/-42V (with higher voltage rated clamp diodes and MJE BJTs for Q9/Q10)
Frequency response(simulated) - ~2Hz to ~800KHz (-3dB)
AC gain - 23 (27dB)
DC gain - Unity
Input level for 50WRMS into 8R - 1VRMS
THD 100Hz 10W - 0.0005%, 1KHz 10W - 0.001%, 10KHz 10W - 0.002%
Slew Rate - 100V/uS
SNR - >100dB
Offset variation - +/-10mV

Screenshot of 1KHz THD spectrum (10W into 8R resistive load, 0.002% source THD):
nBqxTvAn3leWaiaw0BxzZnGnoMSmcwOihGFiLyx2lDkHB9puaaCMxlO8C8rfYXBP7BjLFiJmCNPjWg8=w640
Screenshot of 1KHz + 5.5KHz IMD spectrum (1.7W into 8R load):
Sk9ZuWGXEtFqEJcykNQKpl9qe03Hl58IhLL5OHWrQAbtGe4UA3Jl92Pv-YsnkN_uQd1ZxPH-D6iPxC4=w640

__________________________________________

Group Buy Information:

The PCBs are 92mmx64mm 2-Layer and I plan to batch-print 50, 75 or 100 PCBs depending on interest and how many people join in the GB list. The boards are made with 2.4mm FR4, 60micron copper, HASL finish, white silkscreen and black solder mask, as shown in the above pictures.

> The PCBs cost US$10 a piece.
> After the GB list is full, each user will be contacted and sent the first invoice for PCB cost, and the rest (including shipping charge, packing and processing charge and PayPal charge) is to be paid before shipping the boards, for which the second invoice will be sent.
> International payments are accepted through PayPal and payments from within India are accepted through bank account transfer/deposit, NEFT etc.
> Delay between placing the batch order to PCB plant and delivery of the boards to me will be 18-20 days.


Shipping Information:

Two forms of shipping available. International Air Parcel and Registered Post. Former takes about 15-20 days to deliver and the latter takes about 30-40 days. Both support tracking.

_____________________________________________________

GB1 (April 26 2017)
--------------------------------
Update: Pending members and anyone wishing to join, Please notice:
PM/invoices will be sent up-to 5th May.
Last day of confirming order is 7th May.
PCBs will be ordered on 8th May.

Update(May 9 2017): Total 120 boards ordered. There are still 10 spare PCBs so join in if interested in a couple.
Update(May 11 2017): GB list full. No new entries are being taken until June 1 2017.
GB1 Ended.

--------------------------------

GB2 (June 15 2017)
-------------------------------
Second group buy has started.
GB2 Ended.

-------------------------------

GB3 (Sept 12 2017)
-------------------------------
Third group buy is now open for entry. Join in if interested!
GB3 Ended
-------------------------------

GB4 (Dec 23 2017)
-------------------------------
Fourth group buy has started.
GB4 Ended.
-------------------------------

GB5 (Aug 17 2018)
-------------------------------
Fifth Group Buy arranged. Boards available.

_____________________________________________________


Important Links:

Download Latest BOM for PeeCeeBee V4 >Here<

Download Setup Guide And Component Indication >Here< (Follow this to populate and start the PCBs)

Heat Sink Drill Template >Here<

Thanks.
shaan
🙂

Attachments

For Sale FS Scak Speak Sonus Faber Cremona units in KIT

UPDATED SALE.
Here for sale some Sonus Faber Cremona units.
Some units came from my Cremonas (damaged when my son pushed the dust cap of a woofer with his finger). A pair of woofers came from SF SAT. I will explain:

2x SF Cremona Tweeters. Perfect condition. No damages. They are used and in perfect working and cosmetic condition.
Asking 100€ +pp fees + shipping for the pair




2x SF Cremona Scan Speak 18W8531/G03 Woofers from my Original Cremonas.
One was repaired because the dust cap was damaged by my son. Speakers leads changed as well. Everything was repaired in a Professional service. I can provide invoice.
They measure the same. Perfect working condition. Some minor cosmetic issues (7/10).
Asking 190€ +pp fees + shipping for the pair





2x SF Cremona Scan Speak 18W8531/G03 Woofers
This pair of woofers came from one friend from SF SAT.
The owner coated them. They measure the same than original ones.
Perfect working condition. Average looking condition due to coating (5/10)
Asking 160€ +pp fees + shipping for the pair






I can make lots at discounted price and I can include for free a pair of front baffle for woofers
Each (mm) 758H x 210W x 19D Capable for 2x Scan Speak 18W8531 woofers and a jantzen Ø100mm Bass Refflex port (Ø138,50mm hole)
  • Like
Reactions: vassilis1984

1978 Rotel RA-1412 - slight hissing and pops in the headphone port

Hello!

I just bought this 1978 Rotel RA-1412. Its my first vintage amp and am pretty excited. So far so good aside from a couple things. I noticed while listening through one of the headphone ports im hearing a slight hiss and some crackle pops but its not consistent... only seems to happen once and awhile for a few minutes. I dont hear it when I source isnt playing but it could be by chance.

The way im getting the source is like this: hi-res audio from Windows PC> out through an optical cable> into a newer Rotel RA12 > Pre-Out > to RA-1412 Aux in.

Any place I should look to improve the noise?

The buttons on the right side of the unit (which picks the source) are also kind of finnicky and stick. The bottom button pops completely off if I push it in all the way and I push another button above in. Anyone know where I can find the parts to replace the switch?

Thanks again for everyone's time!

TDA1541A DAC Project Discussion

Hi,

I am sharing the file I got from my friend, it's a TDA1541A DAC with multiple input interface including I2S.
Also, for convenience there is a InterActive BOM that helps a lot while soldering components.

The attached .Zip file contains
A) GERBER files in Zip, ready for fabrication.
B) BOM in HTML.

Please let me know if there is problems.

Regards,
J


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Need Some Advice on Caps to use in Nikko Alpha ii Filter Caps

Hello, I have a set of two different brands of capacitors to install in my Nikko alpha ii power amp and can't decide which are better to use.
When the amp was new, it came with 10,000uf 63V 85 degree ones that had been changed out to others before I got the amp.
The ones that were in it when I got it were Jamico 10,000uf 100V 85 degree units.
I am doing a total re-cap of the entire amp and figured I would replace the filter caps as well while I have it all apart.
I purchased a kit with all caps for the amp, and have no issues with any of them except the filter caps..... and reading all of the forums has got me a bit confused.
Have new caps from Mouser here or new caps that came with the kit.
The choices are: EKMH630VSN103MA50S Nippon United Chemi-Con 10,000uf 63v 105 degrees or Kemet ALC70A103EH100 10,000uf 100v 85 degrees.
I have read some bad things about the Kemets used as power supply filter caps, but not sure if it pertains to these particular ones. Specs are not too bad on either....
So hoping I can get some suggestions from those who know more about this than I do.

Thanks!!!
Russ

Sonos/IKEA Symfonisk hack

Hi there,

I read this article about hacking the Sonos / IKEA Symfonisk, and thought I'd try it with a nice old radio I have in my kitchen. It earns its keep as a decoration alone, but it sure would be nice to get it playing as well.

Now, the first issue that I've run in to (I'm sure there will be more), is that the radio only has one single woofer, while the Sonos looks like some sort of bi-amp setup. I'm not familiar with such. Would it be possible to wire both sets of cables to the same woofer?

From makezine.com:
image031.jpg

Project Madness

Hello all

Everyone who know me understand that I am a big Audio Physic fan, so when I see this one launched at hi end München show, I am getting really stunned... at every point. There are many things that tick boxes here

https://www.audiophysic.com/en/medeos/
  • Point source, or really close to point source, the whole passband down to around 300 Hz is managed by the central 3 drivers really close together, the other drivers symmetrically around this point.
  • Lot of cone area in the mid-bass up to 300 Hz, must leave us with unreal dynamics, and there is nothing like lots of cone area
  • Woofers for midbass located low and high which means it triggers different room modes and so should be more even mid-bass and less room influence with mid-bass, the same gows for the lower bass
  • Integrated subwoofers, an all in one solution that makes it feasible to place even in a medium size flat (like mine)
    With four large / tall towers my place would look like Stonehenge, I don´t want that.... this is also going into the main living room
  • Really really really cool design, the way it looks, IMHO 😍
Somehow this ticks all the boxes for me, except for the price of €160.000, no matter how I look upon it, this is out of reach for me, so it makes me thinking, is it possible to build something with this geometry?
This is a really complex thing to do, 5 way with 3 separate cabinets each side.... Call me absolutely insanely crazy but I am thinking, is this a project I would like to consider?, maybe 😱

It could be a setup with Bliesma beryllium tweeters and hi mids, Purify mids and woofers, several possibilities with subwoofers. But to have any chance it must be active with digital crossovers, this would easily be a €20K to €30K project, so there is pretty high risk here.... I have a working title of "project madness" 🤣


I have not been doing any simulations or calculations on cabinets sizes and/or setup yet.....
Any thoughts guys

Login to view embedded media
Audio_Physic_Medeos-2048x1318.png

Stax DA100 Link For Download The Schematic

Hi to all diyAudio friends
Here is a link for downloading the STAX DA-100 schematic.

http://fileshare.eshop.bg
and enter da100

This is are a rare and hard to find original japanese schematic.

Some years ago i built this amp,whit 2SC3381 and 2SA1349 double transistors.
Currently a friend has this unit,the amp work perfectly from 8 years.
The listen features ,he said,are very good.

If someone download this schematic a little discussion can be started 😱 😱 😱

ESL diaphragm film

Hi whoever's reading this.

I've been planning to build a pair of ESLs for a while now, and I've figured out where I'm buying everything from, except for the mylar film.
I've been looking for mylar film which is about 5 microns thick for a few months now, and the closest thing I can find is 12 micron mylar or 6 micron composite film for RC planes.
Does anyone know of an online vendor who sells 5-6 micron mylar film?

Sorry that it's such a boring question.

Thanks in advance,
Jo

Pam8610 and 4ohm 20w speaker driver (jbl xtreme 2)

Hi all, I'm a newbie in this area, and I want to learn more about it. So anyway, I'm trying to make a portable speaker with a Bluetooth module and Pam8610, a speaker driver (JBL Xtreme 2, 4 ohms - 20w). I power up the Pam86010 with an 8v power supply and the Bluetooth module with a 5v external power source. After playing around with the volume, there are a few problems:

1. When the volume reaches 3/4 of the volume bar, the sound drops at a certain part of the song.

2. Because of "1", I thought maybe power was the problem, so I increased the power from 8v to 9v, and the volume was more stable than before. But the amp got heated up.

3. After 5 songs, the amp was fried.

What should I do? Is it possible to create a portable speaker with my setup?

Thank all

Anyone intimately familiar with the Threshold CAS-1?

Got one for a fine price and am now restoring it internally/electrically, as well as cosmetically, when possible. The fat electrolytes chilling in the center will be replaced, but I'm still afraid that there are more troubles up ahead (the tantalums will be taken care of as well). I actually have very little knowledge about this creation so I would like to ask another question; is this product made to be calibrated/finetuning transistors working current easily by the customer?

Also, how do I send PMs to members?

If "wputera" sees this, please send me a PM regarding the 2sd188 and the 2sa627 I posted as WTB in the market with my old account (Been a member here for a very long time) which unknowingly were linked to my old email adress that's not in use anymore, and in order for me to do anything here with that account, I need to answer that email. Impossible. Happened to me exactly the same way but with Facebook (Old phone number as well) but didn't give the ***** about that as I give about this fantastic place. Cheers.

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Running the Scan-Speak 15w8530k00 full-range with no x-over

Any opinions/ suggestions on running the Scan-Speak 15w8530k00 full-range with no x-over?

http://www.scan-speak.dk/datasheet/pdf/15w-8530k00.pdf


I am currently using the Seas W12CY003 E0044 in a .25 ft3 Parts-Express cabinet with a tweeter/L-Pad added above 12kHz.
As you can see, the bass drops off much quicker than the Scan-Speak. It does have a phase plug so maybe better H-F than the Scan-Speak?

http://www.seas.no/images/stories/excel/pdfdataheet/e0044_w12cy003_datasheet.pdf


Madisound recommends trying a passive radiator to eliminate the 40hz bump but I like the coherency of a single driver. Wouldn't the P-R be like a separate bass driver and not be in-phase? That is 2 sound sources, not one so there will be sound wave interactions between the two.
That is why I am looking at the Scan-Speak for the flatter bass response - IF you can trust the published curves.

Thanks

Speaker mismatch and how it effects the output transformer

Hi everyone, this is from a discussion I had with a friend the other day. I was hoping you could clear something out for me.
So basically what had happened was he had 8 ohm selected from the selector switch of his amp and he connected a 16 ohm speaker. Hearing that something was not right he checked the switch and corrected it. Anyway he was concerned that it might have had a detrimental effect on the output transformer.
My idea was that from what he has done no harm would be caused for the output transformer, but if it was the other way round 16 ohm selected and connected an 8 ohm speaker the output transformer would have been toasted. Then we consulted one of the best tube techs we know and he did mention that from both situations the output transformer could have been fried and that my friend was just lucky that he didn't have his amp fried.
I'll quote what he said "that way around an ohm mismatch is not healthy because tubes hate to be idle.. so it could indeed blow the OT.. but for this amp it does not seem likely because the components and the sheer amount of power available".
Could you please explain why this way round could be a problem for the output transformer.
Thanks in advance.

Passive crossover 2 way - 4 inductors ideal layout, please advise

Hello all,
here is my DCX464 crossover layout - it contains 4 coils. The bottom part is low pass for the mid part (LCL) and the upper part has a highpass (LC) on the left and a notch filter inductor on the right. The board is now 240 x 240 mm. The C-C distance of the inductors as shown is 170 mm along the edges. The squares are other components and terminals. Can I get the board smaller by rotating the inductors better? I could follow the orientation rules for two or three, but what would be the preference when having 4 with these functions? I think I can leave the boards this large, maybe shift the smaller inductors further away a bit closer to the edges if there is nothing much to be gained from different orientation of the inductors. Any opinion is welcome.

1664896870866.png

OK nerds: How to add dynamic damping to a cartridge/arm that does not have it?

The ability to retrofit dynamic damping to a turntable would make more cartridges work in more tone arms. But is there a way to do it?

A million years ago the Shure / Stanton / Pickering brushes were considered a valid solution. Also, remember the paddles on the tone arm that sat in a trough of silicone goo?

Two specific questions for the brain trust:
1- Is there some other approach I am unaware of?
2- Any thoughts on retrofitting any kind of damper are most appreciated. I would love to hear of something off the shelf.

Thanks everyone.

Woofers, Tweeters and mid-range drivers for sale

These drivers for sale have mostly seen little to no use at all. As I'm currently satisfied with the speaker systems that I have built for myself, I have no use for them. I have spec sheets for all of these drivers with a few exceptions. My asking price where possible is 50% of the current new retail price from for example Parts Express or Newark. These prices don't include shipping,

Unless you place a large order and insist on shipping by United Parcel Service (UPS), I will be having the US Postal Service (USPS) doing it. .This is because a Post Office is fairly close to me and this is not the case with a UPS store.

Please see the listing below. .There are a few other speaker building accessories also for sale.

WOOFERS

1. Selenium 12 inch 300W public address, 12PW-5-SLF, one for sale, $45.
2. Newark (MCM) 55-1465, dual voice coils 12 inch w. rubber surround, four for sale, $23 each.
3. Newark (MCM) 55-2185, 8 inch dual voice coils metal cone, no specs available; one for sale for $15.
4. Pioneer 8 inch dual voice coil with foam surround in good condition, B20FU20-52D; four for sale at $15 each.
5. Newark (MCM) 8 inch with rubber surround, 55-1190; two for sale at $10. each.
6. Newark (MCM) 6.5 inch mid-bass with rubber surround, 55-1186HT; four for sale at $10 each.
7. Removed from NHT SB-2 speaker system, 6.5 inch with rubber surround, 4 Ohm, two for sale at $6 each.

MID-RANGE.

1. Aura 5.25 inch, 9 Ohm, foam surround in good condition, 89 dB 1W/ 1M, NS 525-255-8A; sixteen for sale at $2 each.
2. Swan Speaker 4 inch 8 Ohm, Hi-Vi B4N;.3 five for sale, $8 each.
3. Visaton 3.3 inch (8 cm), 4 Ohm, Art No. 2003; ten for sale at $6 each.
4. Newark (MCM) 3 inch paper cone with Santoprene surround, 8 Ohm, 54-605; one for sale at $8.
5. Audax 3 inch with damaged (foam) surround, HT080M03; fixable?, four for sale at $1 each.

TWEETERS

1. Morel 1.25 inch silk dome MDT-12; six available at $23 each.
2. Audax 1 inch silk dome, TM025F1 (discontinued); four available at $7 each.
3. Newark (MCM) PEI 3/4 inch metal dome, 53-440; two for $5 each.

SPEAKER BUILDING SUPPLIES AND PARTS

1. Mahogany Sound Acousta-Stuf 1 pound bag of nylon fiber, $5.
2. Metal grill for 8 inch driver with clamps, two available for $3 each.
3. Round metal grill for 4 inch driver, no clamps, four with one pair slightly larger than the other; $2 each.
4. Square metal grill for 4 inch driver, includes a metal plate that the grill attaches to by press fit; four for sale at $3 each.
5. Plastic vent tube, two inch diameter by 2.5 inch length, Newark 50-14152; six for sale at 25 cents each.
6. Parts Express grill guides, 260-367 ball and socket for 1/2 inch grill, 260-366 press fit for smaller than 1/2 inch thick grill; lot of about 10 sets of 4 guides of each of the two sizes of guides, $4.

Help modeling cabinet for Altec 421-8H

I'm looking for help modeling a bass reflex cabinet an Altec 421-8H? I want to build a pair of high efficiecy 2 way cabinets for my outdoor deck. I will paint the cabinets in deck paint and waterproof the cones. I know it's not good to spray old Altecs but I got these drivers years ago for very little money so I'm ok realizing they will have a limited life outdoors. I will also cover with waterproof covers when not in use. The cabinets need to vent to front.

Thank you,
Milkduds

Needed: Resistor value for Phoenix Gold RX600.5; Schematic would be ideal

I am looking for the value of R16 on the PG RX600.5. It was the resistor near one set of the rectifiers. It is no longer in circuit (long story).

It would be great if someone had a schematic, but based on my research, PG diagrams are not readily available.

Attachments

Calculate PSU

Hello, I would like to ask the following:
It is about a 100-watt amplifier, with an 8-ohm load and a +-40v power supply. To get the current of the power source, we divide 40v by 8 ohms, and get 5 amps. It is not clear to me if this 5 amps is common to both /+ and –/ halves, because if it is, then multiplying the current 5 amps by the voltage 40 volts, we get 200 watts. If we divide these 5 amps by 2 /for the + and – halves/, then 2.5 amps at 40 volts will give us 100 watts of power for each arm. In this regard, am I to understand that the power supply will be 2x40V at 2x2.5 amps? I'm sorry for the bad English, but I use Google translator.

Can it be a sound upgrade to go from a 3-way system to a woofer and compression driver system?

Hello,

I am currently considering switching from a 3-way system with woofer, midrange and tweeter to a system with one woofer, but the tweeter and midrange are being practically replaced by a compression driver (with a waveguide). For me it has to do with the efficiency of the woofer, which is why I came up with the idea. But I just have the feeling that it will always be a downgrade for the sound if I don't have a real midrange/tweeter driver anymore.

(2) Focal woofers 7K011 DBL 7" speakers vintage pair dual voice coil 7 inch -- NH USA --

$200 shipped USA

Work well.
Please see pics for condition.

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  • Like
Reactions: bonjonno

Well this is different!

Functional, organized, aesthetically appealing, consistent, pages' sections clearly defined
is how this forum used to be...

I was gone for a bit, and I've returned to the opposite! Well, there's still consistency.

Anyway, whatever junk this is looks bubbly and has no dividing lines. While browsing the experience is not enjoyable. In fact, it's frustrating. I find myself being put into an angry mood, and I'm normally a pretty happy person - it takes a lot.

I want diyAudio to stay! Functional and organized and fun.

I don't want diyaudio to become useless!!!


Why change the forum so drastically? Sure there are more mobile devices around now, but aren't most people who type a lot doing so (at minimum) on a device with a keyboard? The thing now organizing all our good content, should go back to whence it came 🔲

  • Poll Poll
Order parts from Jacmusic

Please give me suggestion

  • To Everyone

    Votes: 0 0.0%
  • Positive Solution

    Votes: 0 0.0%

I ordered few lundahll transformers and socket teflon tube Yamamoto from Jacmusic. Before the money wired, he answers my emails almost immediately.
Unfortunately after the money was wired, he does not answers to my emails anymore. I waited 15 weeks, and few times called him with no answer.
Until today I did not receive any email whatsoever.

2way crossover

I have two drivers available from RCF, the High Frequency Transducer is ND1411-M and the MiD Bass MB8N251.

I would like to do same test and combine them in a 2 way systems
To have a starting point, can someone tell me the best way to crossover this to drivers.

I appreciate your help.

The HF as Frequency Range : 1500 -20000 Hz
Sensitivity: 109 dB
the MID-BASS: Frequency Range 60 - 3500 Hz
Sensitivity: 96 dB

Attachments

Question for Aleph schematic

Hi all

I am designing the pcb to use with aleph 2 ,3 and 5. after the review the schematic I found that the R C order of the lint that feed the ac signal from output to CCS in Aleph 2 and Aleph 5 are different in order. as I attached the picture of the schematic.

Does the order of RC in this line make any difference on the circuit operation or any difference in sonic character?
I think it was no difference since this line work with AC signal. If it has no affect, I can make a pcb just for one arrangement of RC to use with both Aleph 2 and Aleph 5.

Any suggestion?

Thank you
Tanwa

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Sub tuned incorrectly?

  • TLDR: Subs tuned wrong? Sound kind of boomy and you don't feel the bass quite as much as expected, as there are 4x 18 Inch woofers (4 sub boxes in total, 1 woofer per box). Default tuning frequency is 50Hz, I calculated it for 35Hz and added some ports for that frequency, but that sounds a lot less loud. I'm wondering what the best solution would be for feeling the bass and sounding the best. Subs are hooked up in 2x2 parallel, that makes 4 Ohms. They're hooked up to QSC Powerlight 4.0's (not bridged)

Hi All!

So first of all, I'm quite new to this so I'll most definitely be making some mistakes 😅

Now onto my question,
I have 4 subwoofer cabinets which (I think) are tuned incorrectly. I'm thinking this because they sound very, boomy? And you don't really feel the bass quite as much as I would be expecting, especially since there are 4 18 inch woofers. They're hooked up in 2x2 parallel, which makes 4 ohms, to some QSC Powerlight 4.0's (not bridged)

Specs of the cabinets:
I've attached picture of them for a bit more clarification. Right port has the port calculated for 35Hz and the left port is how they are right now, tuned to 50Hz (just for example)
Woofer: B&C 18pzb46 (https://www.bcspeakers.com/media/W1siZiIsIjIwMTMvMDMvMTEvMTUvNTIvMzAvNTQyL2ZpbGUiXV0)
Volume (internal): 185L (calculated BEFORE the woofer was inserted, don't know if that's correct?)
outer dimensions: 64x52x72cm
ports: 2; slotted; depth 2.8cm ;9x12cm; rounded eges; surface area = 90.6cm2, same as 10.74cm diam. Circle

Now I've been playing around with these specs on some calculator websites to try and tune these subwoofer boxes

Tuning frequency calculation:
First I calculated the tuning frequency, which I thought was necessary, using this website:
https://www.vcalc.com/wiki/subwoofer-tuning-frequency
Values I used:
Air compliance: 155L (Vas in woofer datasheet, is that correct value to use?)
Enclosure volume: 185L
Speaker resonant frequency: 37Hz

These values returned a tuning frequency of ~35Hz

Port length calculation:
I then entered these values in the following website:
http://www.mh-audio.nl/Calculators/WVC.html
I used the top most calculator to calculate the porth lenght.
values I used are:
Dv: 10.24cm (i'm going to 3D print the port, which means that the port will be a little smaller because of the 3D prints wall)
Np: 2
Vb: 185L
fb: 35Hz
k: 0.732

this resulted in a porth lengt of 14.31cm, I 3D printed this, installed it into the subwoofer and sure enough, It sounds quite a lot better on the low end!
But then I noticed that around 50Hz, the sub is quite a lot less loud.

WinISD:
So I went looking for speaker response curve programms, and found one called WinISD. I entered the subs paramaters here and tuned it to 50Hz, 40Hz (in between 50Hz and 35) and 35Hz

The 50 Hz is how it's tuned withouth any additional port, since the front wood is 2.8cm thick.

1664790259627.png


These are the 3 waveforms in WinISD
Red = 50Hz
Green = 40Hz
Blue = 35Hz

The 50Hz has a way higher peak than the other 2 waveforms, which explains why it's such a lot louder at 50Hz, but under 45 Hz it sounds just terrible.

Best advice?
What would your advice be? I'd like to feel the bass as I said, and also for it not to be boomy. Tune it to 40Hz so It's the best of both worlds? Or tune it to 35Hz and would that be loud enough? Or don't tune them at all so they're at 50Hz and just have a really steep high pass filter at around 35-40Hz, so those bad sounding and power consuming low frequencies are cut off, which also would make it so that the amp has a bit more power to spare?

I'd like to hear from you!

Thanks in advance,
-Daan

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New Open Baffle Project

Hi Guys,

I made an open baffle speaker in the spring using the Lii Audio F-15. It was fun, but other builds caught my very short attention span🙂

So I sold them and moved on. But now I'm sorta coming back to wanting to play around with that sound.

I need your advice: This time I want to go two or three way. I will use a passive crossover. As ever, how do I get punchy bass that goes below, say, 30 hz?

I have narrowed it down to three bass drivers:

  • The W-15 from Lii Audio (one on each side)
  • The SB Audience Bianco 15OB (two on each side (if it makes sense))
  • Something from Acoustic Elegance (one on each side)

I hope you can point me in right direction.

Kind regards,
Mads

Feastrex NF5 beta Naturflux

Feastrex 16 ohm 94db 5 inch driver including cabinet up for sale. The cabinet for this driver is an improved version of the F60.

http://www.feastrex.com/products.html
https://6moons.com/audioreviews/miketang/1.html

Was recone by Feastrex japan just about 2years ago to beta cone. well taken care by by me. the speaker cover has plastics and cardboard to protect the driver against water and accident. currently driven by a 15w solid-state amplifier. Very beautiful sounding, sweet and transparent. Nelson pass, Pass Lab and firstwatt founder has their 9inch version, showing how highly regarded it is. Feastrex may not be as well known as hifi speaker companies like Tannoy · Bose · JBL · Klipsch · KEF · Definitive Technology · Bowers & Wilkins · Focal · Dynaudio · Mcintosh · Proac ·KEF · Magico, etc. Perhaps due to marketing fund But its sound will not disappoint and surely surpass at this price.

Looking at USD4000 for the pair of speakers without the tweeter (which acts a supertweeter for me).
Or US$ 3600 for just the drivers.

I am located in Singapore, a country in the Asia.

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Internal Amp Speaker Wiring - Board to Post

Hi Guys, I have a question for the experts here, one which even stumped a guy I know who does this for a living.

I've had great results upgrading amp internal speaker and signal wire. I've used some specialty stuff with impressive SQ outcomes, trouble is, stuff is expensive. The negative post just goes to ground most times - question is, when upgrading the wire, would leaving the stock wire for negative/ground be alright? Would I gain the sonic benefits of upgraded wire (in my case pure silver) with just changing out the positive lead? Assuming the gauges are the same.

What do you guys think?

Thanks!

GFA 565, Help needed. (Used Hoppe's Brain board)

So I have been working on a GFA-565, used Hoppe's Brain board and matched transistors. After getting everything put together and running it, at first I was getting some DC on the output which started out trying to correct itself then went negative and started to climb to a couple volts negative. When the board was built, I ended up with a mistake that likely did some damage on first test (Q111 was in backwards <Cringe> so that is the tip as to damage that might have been done to the circuit - it also got turned on once by another party while I was away before Q116 came in so it was empty - hope that did not hurt anything) I checked all the matched pairs and found one was not reading as a transistor at all anymore at Q101/105 - A13 (using a cheap component checker). Tried using an old pair that still tested alright and it was still off (out of 2 old boards, 3 of 4 were ok but one was 5k hfe, other pair were 45k hfe) so I ordered 30 more from mouser and tested all, they were in 100k-120k hfe range. Picked the most consistent pair at 110k hfe and threw them in and thermally bonded them. Still had -1v offset.

I removed R116 and made a quick DC offset balancer (used 4x D batteries and put a ground in the middle, + and - to a pot going to 1k resistor so I would have -3v to +3v to work with), put that line to the transistor side of R116 and was able to balance the amp to 0 DC on output with -2.2v. So that left me under the impression that the problem was in the servo section of the board. (Had read here that people often saw -4v to +4v from servo). So I start checking the servo parts and found that the C107 and C108 (100nf Wima) were at 92nf and 108nf - so put two of the old babies in and that brought me to -0.120 DC. Started testing and swapping things from old board to see if anything would clean up. Was swapping the opamps at times to see if that was a problem - one of the old Adcom opamps got best result. At some point I found the Feedback from the output ports to wire point 7 on the board had come loose. When I reconnected that and started testing, I was able to get it to zero out! Rechecked bias at 25mv and everything seemed great at that point. Had two rebuilt 565's playing and blasting the hell out of the place. At some point I was checking the temperatures of the amp with a thermal cam and noticed the whole left side of the problematic amp outputs was stone cold. The good amp sounded exactly the same yet it was hot on both sides.

Immediately turned it off and did not have any time left to work on it. Next day I check and it's back to 1v DC on output. Both sides of output banks had rail voltage. Checked the signal outputs to each side. The DRV+ leaving wire from 3 on the board was showing 2.8v and DRV- leaving 4 was showing -0.666v. So I did a quick check of the entire left side - Everything tested fine - all resistors, the couple caps and transistors all tested fine. Tested all the opamps and none of them helped.

Put in my DC offset jig and it was balanced at 0 DC with the same -2.2v I had it set to prior, zero change (Servo was showing -13.48v on pin6/output - what was it trying to correct if it was at 0?). Swapped opamps, no luck. Now I am discouraged and no idea what to go after next, figure my lack of knowledge needed some help.

I am at a loss. I have no idea how the amp goes from 0v DC to suddenly dumping DC again after testing them and having zero issues playing at volume levels that sounded like I was at a concert in front of the speakers. Also how did it sound exactly like the other amp without the PNP side of the outputs not working at all? I figure the speakers would have been toast if that happened during the testing. They were running at rock concert sounding volume for a couple of songs and then all day long playing Star Wars music on May 4th.

Any suggestions and help is greatly appreciated. Thanks for your time!

Dyna-Quik 500 rookie questions.

Hi folks. Short time lurker, first time poster here. Love all the great info here!

To the point, I just started working for a guy who has a fully refurbished Dyna-Quik 500. I'm definitely in need of adapters, so I'm on the hunt for those, but I pulled fuses out of an old Philco radio to test them. But the tube types aren't in the chart. As a rookie who hasn't used more than one other tube tester, this is new to me.

For example, I have a 7B7 which needs the L55a adapter, so I will have to buy that, but I also have a 7J7 that isn't in the chart. This goes for the 7C6 and 7G7 types. Do I just need an updated chart or should I just be way more technical and know what the alternatives should be?

Thanks in advance for any feedback. I really want to make the most of this opportunity and tester!

Genuine or Fake parts

Goodday All

i recently purchased 2sc5200 and 2sa1943 from a local supplier. i have some doubts on the authenticity of them given that the TTC and TTA versions are a better version of these parts. With the uploaded images, can we say these are genuine or fake parts? i need assistance because i have a class h amp where i need to use them on.
please note that the physical dimensions and weight of these are the same with the ttc and tta trannys i already have.
Thanks in advance.
IMG_20220906_113124.jpg

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trying to understand relationship of Vsupply to Vout (graph and datasheet link inside)

maybe it's best explain with an example:

This is my circuit:

Noninverting-op-amp.png


  • Vin is connected to a 5V reference (via a zener diode or a chip like a REF02)
  • Vsupply is +/- 16-18V (controlled via an LM317 or equivalent chip so it can be anything within the LM317 limits)
  • Vout is calculated at around 12V (Rf=24Kohms | Rin=10Kohms ) (5V * 24/10 = 12V)

The way I read the graph is that if I require Iout=2amps then difference between Vsupply and Vout must be at around 3V. (though I'm unsure what Vsupply means in the case since it's a +/- supply)

Here's the graph (There's a high chance I'm looking at the wrong graph but I cannot find anything else suitable in the datasheet)

Screenshot 2022-10-01 133924.jpg


Here's the full datasheet: https://www.ti.com/lit/ds/symlink/opa548.pdf

Fisher xp 7b

I bought the drivers and cross-overs from these speakers in an attempt to re-create the sound of the system on which I first heard Duane Allman and Jimi Hendrix' guitars. I would like to know what the dimensions of the cabinets were, or at least how to figure out what an appropriate size would be. In searching this forum, I realize you folks would dissuade me from re-building these, but that is truly my aim! Any help will be appreciated. Thank you in advance, Mark

  • Poll Poll
JBL Control SB5 modifications

original or modify

  • modify

    Votes: 0 0.0%
  • original

    Votes: 1 100.0%

Recently having bought home a used JBL control SB5 subwoofer ... after 1 day use and getting hand itchy decided to Mod it by swapping in china made "JBL" knock off woofers into the original enclosure .... after modifications as shown below ... bass performance is about the same as the original except the original has slightly higher power handling ... i want to hear opinions from members in this forum should i use this subwoofer in the Modded form or put everything back to the original (the original drivers were removed and well kept in storage box) .... originally it has 4 drivers which could be difficult load for amplifiers but now after mod only using 2 drivers ... thanks
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Audera sub-woofer amplifier, used, amplifier only

Used Audera HD800 0.1 Class D amplifier. Supposedly 650W at 0.1%. "LFE" (no filter) RCA input on small interface board with volume control, power indicator. "Usable" response 10Hz out to 1kHz. Tested, operational. No power supply, only what's pictured. Power input connector clearly designated.

Originally, this amplifier sends a signal to a switching power supply such that it varies its +/- rails in response to the input amplitude. I connected it to a +/- 30VDC, 60 Hz iron transformer based PSU (ignoring the control wire) and it played. I'll guess the max PSU voltage for this amp is +/- 40V, based in its claimed bridged output max p-p voltage of "160V p-p".

I'll assume one could make a powered sub using this, or a bass guitar amplifier - with lots of headroom - by adding an appropriate power supply. I neither need a 650W sub, nor a 650W bass amp...

I'd like $40 for it, plus shipping - USA / Canada only unless you can convince me otherwise. Thanks,

Audera.jpg
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