Single-stage active RIAA correction with second- or third-order Butterworth high-pass included
- By MarcelvdG
- Analogue Source
- 179 Replies
Hi all,
While thinking about ways to speed up the settling of a single-supply single-op-amp RIAA amplifier, see https://www.diyaudio.com/community/...upply-phono-preamp-design.413571/post-7702435 , I found a way to include a second- or third-order Butterworth high-pass filter. As it may be useful outside the context of single-supply circuits, I give it a separate thread.
When you just look at the topology and ignore the component values, this is a rather conventional RIAA amplifier (you could make it even more conventional by connecting R7 in parallel with C5, that doesn't matter much for the principle):
Normally, C8 is used to cause roll-off in the subsonic region and the network R7...R9, C5, C6 realizes the RIAA poles and zero. In this case, however, I use C8 to realize the lowest RIAA pole at -1/(3.18 ms) and R7 to get subsonic roll-off.
Note that C8/C5 = 1000, meaning that without the subsonic roll-off, the DC gain would be 1001, a very ordinary value for a moving-magnet amplifier (1 kHz gain roughly 40 dB).
With everything ideal, at the value of the Laplace variable s where the impedance of C8 cancels the impedance of R12, the feedback disappears and the gain goes to infinity. This means that there is a pole at exactly -1/(R12C8), so if this has to be the first RIAA pole, one needs R12C8 = 3.18 ms. It's actually 3.196 ms in the schematic, pretty close.
The disadvantage of using C8 for the first RIAA pole is that C8, which has a relatively large value, needs to be accurate to get an accurate first RIAA pole. (C8 has practically no effect on the gain at frequencies much greater than 50 Hz, so its tolerance affects deep bass, but not channel balance.) The advantage is that you can include better subsonic filtering in the loop by adding two more resistors and a capacitor.
As an intermediate step, suppose you could add an ideal inductor with a huge value between the output and the negative input of the op-amp, chosen such that it resonates with C5 at the desired subsonic roll-off frequency, and that you chose R7 such that it damps the LC circuit to a quality factor of 1/2 √2. The subsonic response would then be very close to second-order Butterworth. That's because the gain of the RIAA correction amplifier is one plus the ratio of the feedback impedance to the impedance from the negative op-amp input to ground, and that "one" is quite negligible at low frequencies. Mind you, R8 and R9 contribute to the damping of the LC circuit, but not by much. You could also choose a quality factor of 1 and design the AC coupling at the input for the same cut-off frequency. The combined response is then third-order Butterworth.
Such an ideal inductor is totally impractical, but it can be approximated with a T network consisting of two resistors with values much smaller than R7 and a capacitor to ground at the point where they are connected, see this figure:
The transfer from the voltage going into R11 to the current coming out of R10 rolls off at a first-order rate from some very low frequency onwards, like would be the case with an inductor.
I haven't found any simple exact equations for any of the values except R12C8 = 3.18 ms. In fact, I've been very lazy and just calculated approximate values for the other components, and then used a pole-zero extraction program to fine-tune the values.
Regarding those approximate calculations:
R12C8 = 3.18 ms to get the first RIAA pole at the right spot.
The DC gain would be 1 + C8/C5 without subsonic roll-off, so C8/C5 = 1000 gives you a midband gain of roughly 40 dB.
At s = -1/((R8 + R9)(C5 + C6)), the impedance of the network R8, R9, C5, C6 goes to zero and the gain of the circuit becomes 1. As a gain of 1 is pretty close to 0, this must be close to the location of the RIAA zero. That is,
(R8 + R9)(C5 + C6) ≈ 318 μs
At s = -1/((R8 + R9)C6), the impedance of the parallel connection of C6 and R8 & R9 goes to infinity. The impedance of the whole feedback network remains finite due to the other branches R7 and R11, C7, R10, but it does get pretty large. That means the second RIAA pole must be close, so we get the extra criterion
(R8 + R9)C6 ≈ 75 μs
The (theoretical) inductance L is chosen to resonate with C5 at the required subsonic roll-off frequency and R7 is chosen to get the desired quality factor. R10 and R11 get convenient values much smaller than R7 with R10 also much greater than R12. We then have
C7 = L/(R10R11)
Best regards,
Marcel
Edits:
Input RC coupling
The RCR T-network that approximates an inductor actually approximates an inductor with inductance L = R10 R11 C7 and a series resistance of R10 + R11. At low frequencies, it stops behaving inductively, it just turns into the series connection of the two resistors.
As a result, one of the zeros of the high-pass filter that are supposed to lie at s = 0 actually lies somewhere around s = -(R10 + R11)/(R10 R11 C7). For the second-order cases, I have used the first-order high-pass at the input to cover this zero by making the input RC time constant approximately equal to R10 R11 C7/(R10 + R11), or actually to a more accurate value for the displaced zero found by the LINDA pole-zero extraction program.
For the third-order case, I have used the input RC coupling to make the real pole of the third-order Butterworth response, so I couldn't use it to cover the displaced zero. I used the output RC circuit in that case, or simply did not cover the zero. The effect of the zero not being in the origin is typically only seen below 1.something Hz anyway.
There is another zero not exactly in the origin, this is related to the + 1 term in the gain expression of a non-inverting op-amp amplifier. It is so close to 0 that I decided not to bother correcting for it.
Regarding the single-supply versions, the filter R1, C1 is meant to provide some power supply ripple and noise rejection. (Even though the single supply has to be regulated, some extra rejection is very useful at the input of an amplifier that amplifies hum frequencies hundreds of times.) R2 and C2 in series with the cartridge impedance also help to suppress supply ripple. The second-order versions have greater values of C2 than the third-order versions and therefore have better ripple rejection at the lower audio frequencies (such as 100 Hz or 120 Hz).
16 Hz split-supply versions
This is a version for split supply and 16 Hz cut-off frequency, see post #58, https://www.diyaudio.com/community/...rworth-high-pass-included.413649/post-7927611
For the second-order version (values in parenthesis), the input coupling capacitor C2 can be replaced with a short circuit if you don't mind when the roll-off reduces to first order below 1.3 Hz.
This is a variant with 46 dB rather than 40 dB midband gain:
Thanks to having R6 split into R6 and R0, the time constant of the input RC coupling network can be set more accurately without needing awkward values for C2. This was implicitly suggested by hbtaudio on another thread. Because of the high midband gain, the op-amp needs to have a fairly high gain-bandwidth product to get accurate RIAA correction (16 MHz gain-bandwidth product will give about -2 % error of the location of the second RIAA pole).
Finite gain-bandwidth product
See post #100, https://www.diyaudio.com/community/...rworth-high-pass-included.413649/post-7964481 , for some rough calculations on the effect of finite gain-bandwidth product of the op-amp.
From post #101 onward, Nick Sukhov points out that an amplifier with a high open-loop output impedance would result in a loop gain that depends much less on the RIAA correction circuit impedance. That's something to keep in mind when designing a discrete amplifier, you don't have the ability to choose a high open-loop output impedance when using op-amps.
Applying the subsonic filter to a discrete preamplifier based on the Hoeffelman and Meys configuration
The discussion with Nick and Chris about open-loop output impedances made me realize that the subsonic filter of this thread could be combined with a low-noise ("electrically cold") input termination resistance realized with a special feedback configuration that Dual already used in the late 1960's (CV40 phono section, see https://www.diyaudio.com/community/...o-input-load-modification.424717/post-7947176 ) and that was advocated by Hoeffelman and Meys in a 1978 AES article (Jean M. Hoeffelman and René P. Meys, "Improvements of the noise characteristics of amplifiers for magnetic transducers", Journal of the Audio Engineering Society, vol. 26, no. 12, December 1978, pages 935...939, see also Ernst H. Nordholt, "Comments on "Improvement of the noise characteristics of amplifiers for magnetic transducers"", Journal of the Audio Engineering Society, vol. 27, no. 9, September 1979, pages 680...681). The very first electrically cold resistance was made by William Spencer Percival and W. L. Horwood in 1939 as far as I know, but they used a different configuration and did not apply it to phono preamplifiers. See W. S. Percival, "An electrically "cold" resistance", The wireless engineer, vol. 16, May 1939, pages 237...240.
The schematics below show the resulting configurations. They are identical, but the left schematic is for people familiar with nullators and norators, the right schematic for people who feel more comfortable with high-gain twoports and op-amps. At frequencies well above 50 Hz, the input impedance approaches (R13 + R14)/(1 + R13/R12 + R13/R10). You can make this equal to 47 kΩ while using an R14 that is much greater than 47 kΩ, thereby reducing the thermal noise current √(4kT∆f/R) that gets injected into the input.
You can't do this with op-amps (not without floating supplies anyway) because op-amps lack the negative output that conducts a (signal) current equal but opposite to the current through the positive output. That is, you can make electrically "cold" resistances with op-amps, but not as shown here.
Document about dimensioning the circuit
The attached zip file contains a pdf document that explains step-by-step how the component values were found (section 2) and that presents a more accurate method than I have used (one that doesn't need fine-tuning with a pole-zero extraction program, section 3). It also contains a spreadsheet for the more accurate way to calculate the component values.
Deriving the expressions was a nice exercise, but I'm not at all convinced that my more accurate calculation is of any practical use. It can very easily lead to negative or complex resistances.
While thinking about ways to speed up the settling of a single-supply single-op-amp RIAA amplifier, see https://www.diyaudio.com/community/...upply-phono-preamp-design.413571/post-7702435 , I found a way to include a second- or third-order Butterworth high-pass filter. As it may be useful outside the context of single-supply circuits, I give it a separate thread.
When you just look at the topology and ignore the component values, this is a rather conventional RIAA amplifier (you could make it even more conventional by connecting R7 in parallel with C5, that doesn't matter much for the principle):
Normally, C8 is used to cause roll-off in the subsonic region and the network R7...R9, C5, C6 realizes the RIAA poles and zero. In this case, however, I use C8 to realize the lowest RIAA pole at -1/(3.18 ms) and R7 to get subsonic roll-off.
Note that C8/C5 = 1000, meaning that without the subsonic roll-off, the DC gain would be 1001, a very ordinary value for a moving-magnet amplifier (1 kHz gain roughly 40 dB).
With everything ideal, at the value of the Laplace variable s where the impedance of C8 cancels the impedance of R12, the feedback disappears and the gain goes to infinity. This means that there is a pole at exactly -1/(R12C8), so if this has to be the first RIAA pole, one needs R12C8 = 3.18 ms. It's actually 3.196 ms in the schematic, pretty close.
The disadvantage of using C8 for the first RIAA pole is that C8, which has a relatively large value, needs to be accurate to get an accurate first RIAA pole. (C8 has practically no effect on the gain at frequencies much greater than 50 Hz, so its tolerance affects deep bass, but not channel balance.) The advantage is that you can include better subsonic filtering in the loop by adding two more resistors and a capacitor.
As an intermediate step, suppose you could add an ideal inductor with a huge value between the output and the negative input of the op-amp, chosen such that it resonates with C5 at the desired subsonic roll-off frequency, and that you chose R7 such that it damps the LC circuit to a quality factor of 1/2 √2. The subsonic response would then be very close to second-order Butterworth. That's because the gain of the RIAA correction amplifier is one plus the ratio of the feedback impedance to the impedance from the negative op-amp input to ground, and that "one" is quite negligible at low frequencies. Mind you, R8 and R9 contribute to the damping of the LC circuit, but not by much. You could also choose a quality factor of 1 and design the AC coupling at the input for the same cut-off frequency. The combined response is then third-order Butterworth.
Such an ideal inductor is totally impractical, but it can be approximated with a T network consisting of two resistors with values much smaller than R7 and a capacitor to ground at the point where they are connected, see this figure:
The transfer from the voltage going into R11 to the current coming out of R10 rolls off at a first-order rate from some very low frequency onwards, like would be the case with an inductor.
I haven't found any simple exact equations for any of the values except R12C8 = 3.18 ms. In fact, I've been very lazy and just calculated approximate values for the other components, and then used a pole-zero extraction program to fine-tune the values.
Regarding those approximate calculations:
R12C8 = 3.18 ms to get the first RIAA pole at the right spot.
The DC gain would be 1 + C8/C5 without subsonic roll-off, so C8/C5 = 1000 gives you a midband gain of roughly 40 dB.
At s = -1/((R8 + R9)(C5 + C6)), the impedance of the network R8, R9, C5, C6 goes to zero and the gain of the circuit becomes 1. As a gain of 1 is pretty close to 0, this must be close to the location of the RIAA zero. That is,
(R8 + R9)(C5 + C6) ≈ 318 μs
At s = -1/((R8 + R9)C6), the impedance of the parallel connection of C6 and R8 & R9 goes to infinity. The impedance of the whole feedback network remains finite due to the other branches R7 and R11, C7, R10, but it does get pretty large. That means the second RIAA pole must be close, so we get the extra criterion
(R8 + R9)C6 ≈ 75 μs
The (theoretical) inductance L is chosen to resonate with C5 at the required subsonic roll-off frequency and R7 is chosen to get the desired quality factor. R10 and R11 get convenient values much smaller than R7 with R10 also much greater than R12. We then have
C7 = L/(R10R11)
Best regards,
Marcel
Edits:
Input RC coupling
The RCR T-network that approximates an inductor actually approximates an inductor with inductance L = R10 R11 C7 and a series resistance of R10 + R11. At low frequencies, it stops behaving inductively, it just turns into the series connection of the two resistors.
As a result, one of the zeros of the high-pass filter that are supposed to lie at s = 0 actually lies somewhere around s = -(R10 + R11)/(R10 R11 C7). For the second-order cases, I have used the first-order high-pass at the input to cover this zero by making the input RC time constant approximately equal to R10 R11 C7/(R10 + R11), or actually to a more accurate value for the displaced zero found by the LINDA pole-zero extraction program.
For the third-order case, I have used the input RC coupling to make the real pole of the third-order Butterworth response, so I couldn't use it to cover the displaced zero. I used the output RC circuit in that case, or simply did not cover the zero. The effect of the zero not being in the origin is typically only seen below 1.something Hz anyway.
There is another zero not exactly in the origin, this is related to the + 1 term in the gain expression of a non-inverting op-amp amplifier. It is so close to 0 that I decided not to bother correcting for it.
Regarding the single-supply versions, the filter R1, C1 is meant to provide some power supply ripple and noise rejection. (Even though the single supply has to be regulated, some extra rejection is very useful at the input of an amplifier that amplifies hum frequencies hundreds of times.) R2 and C2 in series with the cartridge impedance also help to suppress supply ripple. The second-order versions have greater values of C2 than the third-order versions and therefore have better ripple rejection at the lower audio frequencies (such as 100 Hz or 120 Hz).
16 Hz split-supply versions
This is a version for split supply and 16 Hz cut-off frequency, see post #58, https://www.diyaudio.com/community/...rworth-high-pass-included.413649/post-7927611
For the second-order version (values in parenthesis), the input coupling capacitor C2 can be replaced with a short circuit if you don't mind when the roll-off reduces to first order below 1.3 Hz.
This is a variant with 46 dB rather than 40 dB midband gain:
Thanks to having R6 split into R6 and R0, the time constant of the input RC coupling network can be set more accurately without needing awkward values for C2. This was implicitly suggested by hbtaudio on another thread. Because of the high midband gain, the op-amp needs to have a fairly high gain-bandwidth product to get accurate RIAA correction (16 MHz gain-bandwidth product will give about -2 % error of the location of the second RIAA pole).
Finite gain-bandwidth product
See post #100, https://www.diyaudio.com/community/...rworth-high-pass-included.413649/post-7964481 , for some rough calculations on the effect of finite gain-bandwidth product of the op-amp.
From post #101 onward, Nick Sukhov points out that an amplifier with a high open-loop output impedance would result in a loop gain that depends much less on the RIAA correction circuit impedance. That's something to keep in mind when designing a discrete amplifier, you don't have the ability to choose a high open-loop output impedance when using op-amps.
Applying the subsonic filter to a discrete preamplifier based on the Hoeffelman and Meys configuration
The discussion with Nick and Chris about open-loop output impedances made me realize that the subsonic filter of this thread could be combined with a low-noise ("electrically cold") input termination resistance realized with a special feedback configuration that Dual already used in the late 1960's (CV40 phono section, see https://www.diyaudio.com/community/...o-input-load-modification.424717/post-7947176 ) and that was advocated by Hoeffelman and Meys in a 1978 AES article (Jean M. Hoeffelman and René P. Meys, "Improvements of the noise characteristics of amplifiers for magnetic transducers", Journal of the Audio Engineering Society, vol. 26, no. 12, December 1978, pages 935...939, see also Ernst H. Nordholt, "Comments on "Improvement of the noise characteristics of amplifiers for magnetic transducers"", Journal of the Audio Engineering Society, vol. 27, no. 9, September 1979, pages 680...681). The very first electrically cold resistance was made by William Spencer Percival and W. L. Horwood in 1939 as far as I know, but they used a different configuration and did not apply it to phono preamplifiers. See W. S. Percival, "An electrically "cold" resistance", The wireless engineer, vol. 16, May 1939, pages 237...240.
The schematics below show the resulting configurations. They are identical, but the left schematic is for people familiar with nullators and norators, the right schematic for people who feel more comfortable with high-gain twoports and op-amps. At frequencies well above 50 Hz, the input impedance approaches (R13 + R14)/(1 + R13/R12 + R13/R10). You can make this equal to 47 kΩ while using an R14 that is much greater than 47 kΩ, thereby reducing the thermal noise current √(4kT∆f/R) that gets injected into the input.
You can't do this with op-amps (not without floating supplies anyway) because op-amps lack the negative output that conducts a (signal) current equal but opposite to the current through the positive output. That is, you can make electrically "cold" resistances with op-amps, but not as shown here.
Document about dimensioning the circuit
The attached zip file contains a pdf document that explains step-by-step how the component values were found (section 2) and that presents a more accurate method than I have used (one that doesn't need fine-tuning with a pole-zero extraction program, section 3). It also contains a spreadsheet for the more accurate way to calculate the component values.
Deriving the expressions was a nice exercise, but I'm not at all convinced that my more accurate calculation is of any practical use. It can very easily lead to negative or complex resistances.
PCB advice, Please and THANK you
- By Trzalica
- Electronic Design
- 13 Replies
I want to make an input selector, using Raspberry Zero and phono pre-amp, both jammed on a single board. Since I am IT guy, not an electrical engineer, can you PLEASE review my PCB design and let me know is this going to make it really bad and what should I change to make it work nicely?
This is a RIAA pre-amp part:
This one is an input selector, controlled with Raspberry, with connectors for switch and display:
Here is the entire board, top side first:
...and bottom side
THANK YOU IN ADVANCE!!!!
Marin
This is a RIAA pre-amp part:
This one is an input selector, controlled with Raspberry, with connectors for switch and display:
Here is the entire board, top side first:
...and bottom side
THANK YOU IN ADVANCE!!!!
Marin
For Sale VOLT VM527 2inch dome pair NEW
abandoning a project , i have a brand new pair of VOLT VM527 2inch dome driver to sell , never mounted , never used.
my loss is your gain .
wanting 450€ for the pair including shipping to E.U only
thanks for looking.
my loss is your gain .
wanting 450€ for the pair including shipping to E.U only
thanks for looking.
For Sale Tripath evaluation boards RB-TA3020;RB-TA0105
- By brunomiguel
- Swap Meet
- 15 Replies
I have for sale the following evaluation boards from Tripath.
RB-TA3020-96€
RB-TA0105-160€
I do have more Tripath amplifier boards that you can check here:
https://www.ebay.de/usr/rocksandsound
Payment by Paypal tracked shipment,shipping to US on request.
For more information PM
Attachments
Stands for QUAD ESL-63 speakers
- By MarkRehorst
- Planars & Exotics
- 8 Replies
I recently rebuilt a pair of ESL-63s, and printed some 5 degree tilt back feet, but I don't like having the sound beamed up at my ears from close to the floor, so I designed and built some stands using 1"x1" and 1"x2" t-slot. The stands lift the speaker about 350 mm off the floor, and they are sturdy and center the weight so they aren't easy to knock over.
Details and link to CAD file here. Extra bonus: they are high enough that my cat will probably stop using them as a scratching post.
Details and link to CAD file here. Extra bonus: they are high enough that my cat will probably stop using them as a scratching post.
Cosmos APU a notch+LNA $70 to outperform APx555b for $30,000
- By IVX
- Vendor's Bazaar
- 952 Replies
Yeah, looks like a clickbait but I had such a design target.
Analog Processing Unit contains two devices in the same housing. THAT1510 based preamp-LNA 34/60db gain, 10Vrms max output level, and 48VDC phantom power, just in case if need to work with a condenser mic. The input impedance is 47kOhm 100pF in case it needs to use as a MM phono-preamp without RIAA(Cosmos ADC has internal DSP and may handle RIAA EQ online accurately). The input noise of the LNA is about 130nVrms(A) i.e.
5-6x times fewer than APx555 which, for instance, is not capable to measure the dynamic range of $13 Meizu HiFi DAC, APx555 result is 123-124db(A). Cirrus promises AES17 DR 130db(A) for CS43131, Cosmos APU+Cosmos ADC result is 130.5db(A) with 12db room to APU's residual AES17 DR for 2mV. The LNA's balanced inputs are clamped at 5-6V by TVS thru the 10ohm 0603 resistors in serial to each input i.e. an overvoltage with a good enough current rather may burn the resistors than TVS but that's safe for THAT1510.
The second and biggest half of Cosmos APU is the 1/10kHz active notch filter with -30db ratio(Q about 50) at the fundamental frequency and nearly 0db at 2nd and 3rd harmonics i.e. you can simply divide by 30(or subtract 30db) the THD/THD+N result to get a normal value. Also, with REW or Arta, you can use a calibration file to normalize the notch's frequency response and observe FFT results directly. The max balanced input level is 10Vrms, it is also clamped but by the active clamp with opamp's rail voltage-1V, hence, 10ohm 0603 again may be burned like a fuse. The residual harmonics of the distortions are <-150db@1kHz or <-130db@10kHz, that's probably isn't too spectacular but still better than APx555. The residual THD+N, practically achieved one, -132db@1kHz@10Vrms(DAC+LPF approach), calculations based one is -134db but I can't confirm that so far because of can't find a perfect-zero-noise sine for that 😉 It is clear that APU is kinda a microscope for your ADC, even any laptop audio input with APU gets the ability to measure a tiny noise of DACs or LDOs, and distortion levels are perhaps lower than AP SYS2722 or even Cosmos ADC.
The best way to power Cosmos APU with a power-bank(or a good smartphone charger), is to minimize GND loops issues.
The device is quite simple and inexpensive, strange that's not on the market already. Especially inexpensive(~$70) would be a version without the case as a PCBA which any DIYer can adjust by 4pcs trim-pots(2pcs for the 1kHz Twin-T -30db or -40db, another pair for 10kHz -30db).
REW with the calibration file DAC+LPF->CosmosAPU->CosmosADC
The residual THD+N of 5 preproduction samples, -101db of reading need to subtract 30db i.e. THD+N -131db@9.5Vrms the same about harmonics -127db means -157db
APU Notch frequency responses:
AES17 Dynamic Range test performs at -60dbfs level by measuring the THD+N A-weighted. That plot shows the residual THD+N (A) of the LNA at 1-10mV level sweep. 2mV corresponds 2V 0dbfs DAC like CS43131, and -82db needs to subtract 60db to get AES17 DR = 142db. A typical high-performance DAC with 5Vrms 0dbfs could be measured down to 150db and so on.
The residual noise of 5 preproduction samples(the S2 using SSM2019 instead of THAT1510), the reading need to divide by 1000 due to 60db of LNA i.e. a uV is actually nV.
The gain of Cosmos APU preamp is +60db(1000x times), hence the scale units are nVrms(A). After the APU unit turns On, you can see 5s of settling,
from 6s the input noise reaches 130nV, about 9s 48V power was turned On, and after 1s reached 142nS. Hence, APU's phantom power adds just 12nV to the preamp noise, that's a nearly ideal result as I think 😉 The test was performed with a shorted preamp's inputs(In+ to In-) and biased to the GND by 2x6.8kOhm resistor as a dummy condenser mic.
and the spectrum 20Hz-130kHz of the same test, units need to divide by 1000 as well, so uV becomes nV. This is a good illustration to killing the myth that SMPS is always dirty 😉 Cosmos APU contains no linear voltage regulators at all, only SMPS with Fswitching > 1MHz.
Finally, I got working Cosmos ADC with internal DSP and RIAA EQ implemented for all Fs from 44.1 to 384. A pair of Cosmos APU was used as a dual-mono phono preamp +34db gain with no analog RIAA, which was applied in Cosmos ADC digitally. The video is a comparison of LP vs CD sound, recorded this way.
The formal specs of such a combo:
THD+N@5mV-40ohm(AP output) -83db, and suddenly it places Cosmos APU to the 2nd position of the phono-preamps ASR rating(and yes, I know his tests are silly) 😉
I remeasured Cosmos APU+Cosmos ADC+RIAA max accurate, AC line hum virtually zero.
5mV 10ohm source APU gain 34db Cosmos ADC 1.7V sensitivity, THD+N -89db or -97.7db(A). Due to H2 and H3 being very low, SNR is also 97db(A).
250uV 10ohm source APU gain 60db Cosmos ADC 1.7V sensitivity, THD+N -66db or -78db(A). Due to H2 and H3 being lower than the noise floor, SNR is also 78db(A).
The real SNR i.e. APU source is a cheap MM cartridge AT3600L, the LP record is digitized by Cosmos ADC@32/384, normalized to fit max peak to the -6dbfs, next, the stylus is up to the air, Pioneer PL-300(made in Japan 1980) turntable keeps 33.33rpm. This way the SNR = 91,2dbfs-6db=85.2db(A), that's looks decent if the LP "silence" between the tracks shows SNR = 65db(A) and less.
Of course, both these functions, mic and phono preamp, are features just for the feature list, but I think both are decently implemented and virtually for free so I couldn't call that a marketing 😉
https://e1dashz.wixsite.com/index/cosmos-apu
https://archimago.blogspot.com/2022/05/early-look-cosmos-apu-high-performance.html#more
https://www.l7audiolab.com/f/e1da-cosmos-apu-2/ APx555b measurements
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Analog Processing Unit contains two devices in the same housing. THAT1510 based preamp-LNA 34/60db gain, 10Vrms max output level, and 48VDC phantom power, just in case if need to work with a condenser mic. The input impedance is 47kOhm 100pF in case it needs to use as a MM phono-preamp without RIAA(Cosmos ADC has internal DSP and may handle RIAA EQ online accurately). The input noise of the LNA is about 130nVrms(A) i.e.
5-6x times fewer than APx555 which, for instance, is not capable to measure the dynamic range of $13 Meizu HiFi DAC, APx555 result is 123-124db(A). Cirrus promises AES17 DR 130db(A) for CS43131, Cosmos APU+Cosmos ADC result is 130.5db(A) with 12db room to APU's residual AES17 DR for 2mV. The LNA's balanced inputs are clamped at 5-6V by TVS thru the 10ohm 0603 resistors in serial to each input i.e. an overvoltage with a good enough current rather may burn the resistors than TVS but that's safe for THAT1510.
The second and biggest half of Cosmos APU is the 1/10kHz active notch filter with -30db ratio(Q about 50) at the fundamental frequency and nearly 0db at 2nd and 3rd harmonics i.e. you can simply divide by 30(or subtract 30db) the THD/THD+N result to get a normal value. Also, with REW or Arta, you can use a calibration file to normalize the notch's frequency response and observe FFT results directly. The max balanced input level is 10Vrms, it is also clamped but by the active clamp with opamp's rail voltage-1V, hence, 10ohm 0603 again may be burned like a fuse. The residual harmonics of the distortions are <-150db@1kHz or <-130db@10kHz, that's probably isn't too spectacular but still better than APx555. The residual THD+N, practically achieved one, -132db@1kHz@10Vrms(DAC+LPF approach), calculations based one is -134db but I can't confirm that so far because of can't find a perfect-zero-noise sine for that 😉 It is clear that APU is kinda a microscope for your ADC, even any laptop audio input with APU gets the ability to measure a tiny noise of DACs or LDOs, and distortion levels are perhaps lower than AP SYS2722 or even Cosmos ADC.
The best way to power Cosmos APU with a power-bank(or a good smartphone charger), is to minimize GND loops issues.
The device is quite simple and inexpensive, strange that's not on the market already. Especially inexpensive(~$70) would be a version without the case as a PCBA which any DIYer can adjust by 4pcs trim-pots(2pcs for the 1kHz Twin-T -30db or -40db, another pair for 10kHz -30db).
REW with the calibration file DAC+LPF->CosmosAPU->CosmosADC
The residual THD+N of 5 preproduction samples, -101db of reading need to subtract 30db i.e. THD+N -131db@9.5Vrms the same about harmonics -127db means -157db
APU Notch frequency responses:
AES17 Dynamic Range test performs at -60dbfs level by measuring the THD+N A-weighted. That plot shows the residual THD+N (A) of the LNA at 1-10mV level sweep. 2mV corresponds 2V 0dbfs DAC like CS43131, and -82db needs to subtract 60db to get AES17 DR = 142db. A typical high-performance DAC with 5Vrms 0dbfs could be measured down to 150db and so on.
The residual noise of 5 preproduction samples(the S2 using SSM2019 instead of THAT1510), the reading need to divide by 1000 due to 60db of LNA i.e. a uV is actually nV.
The gain of Cosmos APU preamp is +60db(1000x times), hence the scale units are nVrms(A). After the APU unit turns On, you can see 5s of settling,
from 6s the input noise reaches 130nV, about 9s 48V power was turned On, and after 1s reached 142nS. Hence, APU's phantom power adds just 12nV to the preamp noise, that's a nearly ideal result as I think 😉 The test was performed with a shorted preamp's inputs(In+ to In-) and biased to the GND by 2x6.8kOhm resistor as a dummy condenser mic.
and the spectrum 20Hz-130kHz of the same test, units need to divide by 1000 as well, so uV becomes nV. This is a good illustration to killing the myth that SMPS is always dirty 😉 Cosmos APU contains no linear voltage regulators at all, only SMPS with Fswitching > 1MHz.
Finally, I got working Cosmos ADC with internal DSP and RIAA EQ implemented for all Fs from 44.1 to 384. A pair of Cosmos APU was used as a dual-mono phono preamp +34db gain with no analog RIAA, which was applied in Cosmos ADC digitally. The video is a comparison of LP vs CD sound, recorded this way.
The formal specs of such a combo:
THD+N@5mV-40ohm(AP output) -83db, and suddenly it places Cosmos APU to the 2nd position of the phono-preamps ASR rating(and yes, I know his tests are silly) 😉
I remeasured Cosmos APU+Cosmos ADC+RIAA max accurate, AC line hum virtually zero.
5mV 10ohm source APU gain 34db Cosmos ADC 1.7V sensitivity, THD+N -89db or -97.7db(A). Due to H2 and H3 being very low, SNR is also 97db(A).
250uV 10ohm source APU gain 60db Cosmos ADC 1.7V sensitivity, THD+N -66db or -78db(A). Due to H2 and H3 being lower than the noise floor, SNR is also 78db(A).
The real SNR i.e. APU source is a cheap MM cartridge AT3600L, the LP record is digitized by Cosmos ADC@32/384, normalized to fit max peak to the -6dbfs, next, the stylus is up to the air, Pioneer PL-300(made in Japan 1980) turntable keeps 33.33rpm. This way the SNR = 91,2dbfs-6db=85.2db(A), that's looks decent if the LP "silence" between the tracks shows SNR = 65db(A) and less.
Of course, both these functions, mic and phono preamp, are features just for the feature list, but I think both are decently implemented and virtually for free so I couldn't call that a marketing 😉
https://e1dashz.wixsite.com/index/cosmos-apu
https://archimago.blogspot.com/2022/05/early-look-cosmos-apu-high-performance.html#more
https://www.l7audiolab.com/f/e1da-cosmos-apu-2/ APx555b measurements
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Large Electrolytic Capacitors FS (new)
I have twelve (12) brand new 22,000 uF, 80V capacitors. Ordering error from Mouser and I'm stuck with them. I'm trying to recover some funds. Asking $35 each, plus USPS shipping from zip 17402.
4.125 inches tall and 2.5 inches in diameter. Great for all your power amp projects! Send me a PM if you're interested.
Mouser part number: 598-CGS223U080W4C
4.125 inches tall and 2.5 inches in diameter. Great for all your power amp projects! Send me a PM if you're interested.
Mouser part number: 598-CGS223U080W4C
Tweeter off-axis patterns
- By lehmanhill
- Multi-Way
- 14 Replies
Background – I’ve built speakers in the past but my most recent build was in 1999. So I am trying to read and catch up with current speaker building trends.
A lot more attention is being paid to off-axis performance these days. I gather that consistent and as wide as possible directivity are said to give a more realistic sound. That said, I am seeing two different approaches to off axis in tweeters. I have borrowed a couple of normalized off-axis curves from hificompass.com to demonstrate. Many thanks to them for their fine measurements.
Some folks are trying to keep the widest directivity to as high a frequency as possible followed by a smooth narrowing at higher frequencies. An example of this is the BlieSma T25S-6 where there is no narrowing of the off-axis curves until about 6.5k Hz. Then it narrows linearly until a break-up in the 17k – 20k region.
Others prefer an early but smooth narrowing, often accomplished with a wave guide. An example of this philosophy is the Satori TW29BNWG-4 which uses a wave guide and starts narrowing about 1300 Hz. As it happens, hificompass.com offers 3D printer plans for a the BlieSma T25S-6 which changes the T25S-6 into something like the Satori.
My question is which kind of directivity pattern is preferred and why? If it’s just preference, what are the advantages and disadvantages of each.
Thanks.
A lot more attention is being paid to off-axis performance these days. I gather that consistent and as wide as possible directivity are said to give a more realistic sound. That said, I am seeing two different approaches to off axis in tweeters. I have borrowed a couple of normalized off-axis curves from hificompass.com to demonstrate. Many thanks to them for their fine measurements.
Some folks are trying to keep the widest directivity to as high a frequency as possible followed by a smooth narrowing at higher frequencies. An example of this is the BlieSma T25S-6 where there is no narrowing of the off-axis curves until about 6.5k Hz. Then it narrows linearly until a break-up in the 17k – 20k region.
Others prefer an early but smooth narrowing, often accomplished with a wave guide. An example of this philosophy is the Satori TW29BNWG-4 which uses a wave guide and starts narrowing about 1300 Hz. As it happens, hificompass.com offers 3D printer plans for a the BlieSma T25S-6 which changes the T25S-6 into something like the Satori.
My question is which kind of directivity pattern is preferred and why? If it’s just preference, what are the advantages and disadvantages of each.
Thanks.
A guide to building the Pass F4 amplifier
Building the Pass/Firstwatt F4
This is a fantastically good sounding amp - read more about it here before staring;
https://www.firstwatt.com/wp-content/uploads/2023/08/prod_f4_man.pdf
The thread at DIY audio -
http://www.diyaudio.com/forums/pass-labs/97540-f4-power-amplifier.html
And this the the corrected schematic (The schematic in the Firstwatt article has a typo, also this one agrees with the PCB)
Here you will find a build guide for the Pass / Firstwatt F4 power amplifier using PCBs and chassis from the DIYaudio store.
The 5U 'BIG Amp Chassis' is shown, because that's the one I have. It will fit comfortably in a 4U 'Jack of all chassis' and have enough heatsink as well.
~~~~~
There are plenty of places that you could start, but for the sake of illustration let's begin with the heatsink assemblies -
This is the heatsink(s) from the 5U 'BIG Amp Chassis' It has a mirror-imaged set of pre-drilled heatsinks and brackets to hole them together and make a mounting point for the rest of the enclosure. The 4U is similar, but the heatsink is a single piece.
This build will also utilize the 'DIY friendly' baseplate, here shown with the feet and hardware, and also the heatsink's brackets.
There is a hardware package available for the pre-drilled back and heatsinks, including input and output jacks, IEC module, and hardware for the PCB and heatsinks.
The contents of the hardware bag.
Using the brass PCB standoffs, install them into the PCB mount holes as shown, to get the following pattern;
Now there is a place to mount the amplifier PCB
Speaking of PCB, it's a very nice layout, plenty of room, and the ability to use many sizes of resistors and caps. This is the front.
Here is the back.
Stuffing the PCB should be done in the usual order, from smallest device to biggest - so that would be diodes and resistors first.
Of note, I got the bigger (Dale/Vishay RN60) resistors to see how they would fit on the PCB. They are the size of the PRP resistors that are quite popular amongst the fancy parts crowd. They are great everywhere except the row flanking the small transistors right in the middle. They don't fit there side by side. You could mount them soldier style, or just mix in a few smaller resistors like I did. Or just get RN55's. They are the smaller size.
Pots and transistors next. (yes, I didn't stuff the input pair when the photo was taken…)
Ah, there they are.
The ziptie is just to help their thermal tracking.
And finally the capacitors.
There is a top and bottom to the Universal Mounting Spec holes, the board mounts as shown
I find it helpful to bend the leads of the transistors first, and mount them (a little bit loose) to the heatsink.
Like this
And then mount the PCB. You can snug all the screws down and then solder and trim.
Ok, now lets move on to the Power Supply.
Here is a photo of the PSU board, I am going to use integrated bridge rectifier blocks, so you need to remove the part of the PCB that mounts the diodes. The new PCB, not quite yet available at the time of this writing, will have a similar feature with the diodes, as well as room for more/bigger capacitors.
As always, stuff the small components first - the light blue resistors are the filter resistors, the darker ones with the teflon are the bleeder resistors, and the small ones are for the LEDs.
This PSU board is the exact same DIYaudio PSU board, just without the top blue soldermask.
This shows the INPUT edge (from the diode bridges)
The capacitors are Panasonic T-UP 33,000uf 35V
Connecting the bridges to the PCB
This is the OUTPUT edge of the PSU - the colors are
Red V+
White GND
Green V-
The black connects the PSU GND to the CL-60 to the chassis.
The wiring from the PSU to the amp PCB is clearly shown.
Here you can see the bridges with the wires attached from the transformer secondary. Remember that the green attached to a bridge must have continuity with the blue attached to the same bridge. (As it's the 2 ends of the same piece of wire)
As long as were are tailing about the transformer, here is a photo of the terminal block shown wired for 120v. The Blur lead is the AC Live, and the clear the AC neutral. The reds and blacks are the transformer primaries.
Transformer is an Antek 400VA 18v+18v, part number AN-4218.
The last bit of the PSU wiring is the chassis connection, the black comes from the PSU GND, and the green is the AC safety earth.
The AC to primary wiring confuses everybody, so;
Let's look at the PSU schematic just to make sure everything is OK... Remember that I am wiring it for 120v operation, so the transformer primaries are in parallel. People wiring for 240 with a transformer like this, please ignore.
Notes in red are mine.
Look at the connections of the transformer primary, through the thermistors and line cap, to the mains.
Hot AC is connected to the "120" (which in my case is the red leads on the primaries) One red primary is connected to AC hot through a thermistor.
Neutral AC is connected to the black "0" leads, one of which is connected to the AC through a thermistor.
AC Hot and Neutral have a cap across the leads.
So, yes, the AC will be connected to the center 2 posts, which is across the cap.
(This photo lifted from my F5 thread, but it's the same PSU…)
Left to right we have post 1, 2, 3, 4
POST 1 - Transformer primary 'B 0' which will be connected to AC Neutral at post 2, through the thermistor between post 1 and 2.
POST 2 - AC Neutral in (not shown in photo), connected to Transformer primary 'A 0" , a thermistor to post 1, and a line cap to post 3
POST 3 - AC Hot in, connected to transformer primary "B 120", thermistor to post 4, and line cap to post 2
POST 4 - Transformer primary "A 120", connected to AC Hot through the thermistor to post 3
If you look at the red and black wires in the photo you will see that the Mains AC must to pass through a thermistor to connect to each of the 2 primaries. And that is the point of them, to keep inrush under control during powerup.
Ok!
Now we need to put everything together -
But first a bit more mechanical assembly.
This is the pre-cut back plate and the thick front plate.
Gather and mount the IEC plug.
The speaker posts.
And the RCA jacks.
Note that the shoulder washer goes on the inside, so the metal of the chassis doesn't touch the metal of the jack. There is a similar washer on the speaker posts.
The inside of the back panel.
And the outside. Looks good, yes?
The IEC module is wired as shown. This will switch both the Live and Neutral. The blue (live) and clear (neutral) go the the wiring block with the thermistors, cap and transformer primaries.
The amp PCB completely wired.
A bit closer.
The top connections labeled.
This happens to be the other channel, but the connections are all the same.
Remember that V- / GND / V+ is always left to right as you are looking at the format of the PCB
I'm not entirely sure what I was trying to show here, other than the screw and washer. It looks cool. I will keep the photo in the guide.
A few notes on bias -
P1 controls the bias, measured across any of the 3W source resistors. Adjust for 0.13v when it's cold, and watch that it doesn't get higher than 0.2v once it's up to temperature in about an hour. Adjust for 0.2v when hot.
P2 is used to adjust the DC offset on the output to zero.
Attach a DC voltmeter across the speaker outputs to measure offset.
If you find that the P1 doesn't have enough range, I.E., you can't turn it up enough, replace R9 with a smaller resistor, I used 4.75K and it works well.
Connect a voltmeter across any of the source resistors. The outboard ones are easier to clip across.
Adjust for about 0.13v cold, and once the amp is up to operating temperature, trim for 0.20v - It takes a long time to warm up, take your time.
Adjust P2 for zero offset, then re-trim P1
Here is a photo of it all connected and working -
I'm driving the F4 with an O2 Headphone amp sourced from an iPod; Driving 85.5db speakers. Although it is a small room, it gets louder than I want to listen. It still can't drive it to clipping, but it does get really, really loud.
One thing worth mentioning, and it speaks very highly to the quality of the amp, is that it is completely non-fatiguing, and more interestingly, very easy to listen to turned up too loud… I don't realize how loud it actually is sometimes. Complete transparency is a word used a lot when describing this amp - but I have to agree. It's fantastic!
Please comment away if you desire.
Also please feel free to ask any F4 questions here, and if you would like to post photos of your F4 completions, old or new, please do!
This is a fantastically good sounding amp - read more about it here before staring;
https://www.firstwatt.com/wp-content/uploads/2023/08/prod_f4_man.pdf
The thread at DIY audio -
http://www.diyaudio.com/forums/pass-labs/97540-f4-power-amplifier.html
And this the the corrected schematic (The schematic in the Firstwatt article has a typo, also this one agrees with the PCB)
Here you will find a build guide for the Pass / Firstwatt F4 power amplifier using PCBs and chassis from the DIYaudio store.
The 5U 'BIG Amp Chassis' is shown, because that's the one I have. It will fit comfortably in a 4U 'Jack of all chassis' and have enough heatsink as well.
~~~~~
There are plenty of places that you could start, but for the sake of illustration let's begin with the heatsink assemblies -
This is the heatsink(s) from the 5U 'BIG Amp Chassis' It has a mirror-imaged set of pre-drilled heatsinks and brackets to hole them together and make a mounting point for the rest of the enclosure. The 4U is similar, but the heatsink is a single piece.
This build will also utilize the 'DIY friendly' baseplate, here shown with the feet and hardware, and also the heatsink's brackets.
An externally hosted image should be here but it was not working when we last tested it.
There is a hardware package available for the pre-drilled back and heatsinks, including input and output jacks, IEC module, and hardware for the PCB and heatsinks.
The contents of the hardware bag.
Using the brass PCB standoffs, install them into the PCB mount holes as shown, to get the following pattern;
Now there is a place to mount the amplifier PCB
Speaking of PCB, it's a very nice layout, plenty of room, and the ability to use many sizes of resistors and caps. This is the front.
Here is the back.
Stuffing the PCB should be done in the usual order, from smallest device to biggest - so that would be diodes and resistors first.
Of note, I got the bigger (Dale/Vishay RN60) resistors to see how they would fit on the PCB. They are the size of the PRP resistors that are quite popular amongst the fancy parts crowd. They are great everywhere except the row flanking the small transistors right in the middle. They don't fit there side by side. You could mount them soldier style, or just mix in a few smaller resistors like I did. Or just get RN55's. They are the smaller size.
Pots and transistors next. (yes, I didn't stuff the input pair when the photo was taken…)
Ah, there they are.
The ziptie is just to help their thermal tracking.
And finally the capacitors.
There is a top and bottom to the Universal Mounting Spec holes, the board mounts as shown
I find it helpful to bend the leads of the transistors first, and mount them (a little bit loose) to the heatsink.
Like this
And then mount the PCB. You can snug all the screws down and then solder and trim.
Ok, now lets move on to the Power Supply.
Here is a photo of the PSU board, I am going to use integrated bridge rectifier blocks, so you need to remove the part of the PCB that mounts the diodes. The new PCB, not quite yet available at the time of this writing, will have a similar feature with the diodes, as well as room for more/bigger capacitors.
As always, stuff the small components first - the light blue resistors are the filter resistors, the darker ones with the teflon are the bleeder resistors, and the small ones are for the LEDs.
This PSU board is the exact same DIYaudio PSU board, just without the top blue soldermask.
This shows the INPUT edge (from the diode bridges)
The capacitors are Panasonic T-UP 33,000uf 35V
Connecting the bridges to the PCB
This is the OUTPUT edge of the PSU - the colors are
Red V+
White GND
Green V-
The black connects the PSU GND to the CL-60 to the chassis.
The wiring from the PSU to the amp PCB is clearly shown.
Here you can see the bridges with the wires attached from the transformer secondary. Remember that the green attached to a bridge must have continuity with the blue attached to the same bridge. (As it's the 2 ends of the same piece of wire)
As long as were are tailing about the transformer, here is a photo of the terminal block shown wired for 120v. The Blur lead is the AC Live, and the clear the AC neutral. The reds and blacks are the transformer primaries.
Transformer is an Antek 400VA 18v+18v, part number AN-4218.
The last bit of the PSU wiring is the chassis connection, the black comes from the PSU GND, and the green is the AC safety earth.
The AC to primary wiring confuses everybody, so;
Let's look at the PSU schematic just to make sure everything is OK... Remember that I am wiring it for 120v operation, so the transformer primaries are in parallel. People wiring for 240 with a transformer like this, please ignore.
Notes in red are mine.
Look at the connections of the transformer primary, through the thermistors and line cap, to the mains.
Hot AC is connected to the "120" (which in my case is the red leads on the primaries) One red primary is connected to AC hot through a thermistor.
Neutral AC is connected to the black "0" leads, one of which is connected to the AC through a thermistor.
AC Hot and Neutral have a cap across the leads.
So, yes, the AC will be connected to the center 2 posts, which is across the cap.
(This photo lifted from my F5 thread, but it's the same PSU…)
Left to right we have post 1, 2, 3, 4
POST 1 - Transformer primary 'B 0' which will be connected to AC Neutral at post 2, through the thermistor between post 1 and 2.
POST 2 - AC Neutral in (not shown in photo), connected to Transformer primary 'A 0" , a thermistor to post 1, and a line cap to post 3
POST 3 - AC Hot in, connected to transformer primary "B 120", thermistor to post 4, and line cap to post 2
POST 4 - Transformer primary "A 120", connected to AC Hot through the thermistor to post 3
If you look at the red and black wires in the photo you will see that the Mains AC must to pass through a thermistor to connect to each of the 2 primaries. And that is the point of them, to keep inrush under control during powerup.
Ok!
Now we need to put everything together -
But first a bit more mechanical assembly.
This is the pre-cut back plate and the thick front plate.
Gather and mount the IEC plug.
The speaker posts.
And the RCA jacks.
Note that the shoulder washer goes on the inside, so the metal of the chassis doesn't touch the metal of the jack. There is a similar washer on the speaker posts.
The inside of the back panel.
And the outside. Looks good, yes?
The IEC module is wired as shown. This will switch both the Live and Neutral. The blue (live) and clear (neutral) go the the wiring block with the thermistors, cap and transformer primaries.
The amp PCB completely wired.
A bit closer.
The top connections labeled.
This happens to be the other channel, but the connections are all the same.
Remember that V- / GND / V+ is always left to right as you are looking at the format of the PCB
I'm not entirely sure what I was trying to show here, other than the screw and washer. It looks cool. I will keep the photo in the guide.
A few notes on bias -
P1 controls the bias, measured across any of the 3W source resistors. Adjust for 0.13v when it's cold, and watch that it doesn't get higher than 0.2v once it's up to temperature in about an hour. Adjust for 0.2v when hot.
P2 is used to adjust the DC offset on the output to zero.
Attach a DC voltmeter across the speaker outputs to measure offset.
If you find that the P1 doesn't have enough range, I.E., you can't turn it up enough, replace R9 with a smaller resistor, I used 4.75K and it works well.
Connect a voltmeter across any of the source resistors. The outboard ones are easier to clip across.
Adjust for about 0.13v cold, and once the amp is up to operating temperature, trim for 0.20v - It takes a long time to warm up, take your time.
Adjust P2 for zero offset, then re-trim P1
Here is a photo of it all connected and working -
I'm driving the F4 with an O2 Headphone amp sourced from an iPod; Driving 85.5db speakers. Although it is a small room, it gets louder than I want to listen. It still can't drive it to clipping, but it does get really, really loud.
One thing worth mentioning, and it speaks very highly to the quality of the amp, is that it is completely non-fatiguing, and more interestingly, very easy to listen to turned up too loud… I don't realize how loud it actually is sometimes. Complete transparency is a word used a lot when describing this amp - but I have to agree. It's fantastic!
Please comment away if you desire.
Also please feel free to ask any F4 questions here, and if you would like to post photos of your F4 completions, old or new, please do!
The beautiful design of SB Acoustics SB17
This are my assumptions, and i am not an audio engineer.
This midrange is not designed for bass even if has lower fs. 30hz
The low fs come from the very lose suspension, believe it or not a loose suspension means eficiency, in a way, because it lowers the qes! Also making it good for ported, making it even more efficient.
Thats why they have such low qes qts.
This means that the coil doesnt have to be denser to achieve lower qes, instead the coil can have thinner wires to achieve a bigger bl (more total coil in the gap) better motor force controll. And with the loose suspension mechanical looses are very low.
This gives it very low distortions
SMD components
Hiya,
Looking at SMD components to minimize area on PCBs, but kind of lost in what components to use.
In particular I want to add some caps/ferrites to stop noise/oscillations and wanted to know if I should go for PPS caps for the 10nF/100nF, and what to look for for ferrites. In through-hole components I am using MKP for the caps, ferrite beads on some wire and 150R tantanlum resistors but this takes up more space than I would like (even when using resistors/ferrites vertically.
Not sure what to look for in SMD equivalents. Anyone with experience in SMD components?
Looking at SMD components to minimize area on PCBs, but kind of lost in what components to use.
In particular I want to add some caps/ferrites to stop noise/oscillations and wanted to know if I should go for PPS caps for the 10nF/100nF, and what to look for for ferrites. In through-hole components I am using MKP for the caps, ferrite beads on some wire and 150R tantanlum resistors but this takes up more space than I would like (even when using resistors/ferrites vertically.
Not sure what to look for in SMD equivalents. Anyone with experience in SMD components?
For Sale Vifa D27TG-45-06 Tweeters ( pair ) £60
- By TheFinisher
- Swap Meet
- 7 Replies
Working perfectly and in excellent cosmetic condition .
Bought to compare in my speakers
£60 + shipping ( I’m in the UK )
Bought to compare in my speakers
£60 + shipping ( I’m in the UK )
Attachments
One of the best RIAA Network capacitor - for me unknown Brand
- Analogue Source
- 40 Replies
Attachments
Help Sought: Completing My Late Father’s DIY Audio Builds
- By LEVAYEV93
- Construction Tips
- 8 Replies
I'm sharing a slightly different kind of post here, hoping it will find its way to the right people. I’d truly appreciate any shares in relevant places.With great sadness, my family and I recently said goodbye to my father after his battle with cancer. My dad was a very special person, and there’s so much I could say about him—but in the context of this group, there’s one thing in particular that stands out: his deep passion for sound and audio.
Over many years, he acquired an impressive amount of knowledge, technical skill, and understanding, which he used to restore, upgrade, and build a variety of equipment—from amplifiers to speakers, turntables, and more. Before he became ill, he had started working on several projects, but sadly had to stop midway, and now they sit unfinished with an uncertain future.
I’ll be the first to admit—I don’t have the expertise my father had. What I do share with him is a deep love for music, which was one of the strongest bonds between us. But unfortunately, he didn’t pass down his technical skills to me.I’m posting here because I feel that my father left me with a mission—his legacy, if you will. And I’ve made it my goal to try and complete the work he couldn’t finish himself.
I’m looking for a mentor, a guide—someone who can help me understand where to even begin with these projects that meant so much to my father. I’ve attached some photos of what he left behind, in the hope that someone might see the value in this and, out of love for this beautiful world of stereo and sound, help me breathe life back into a dream that’s been paused.
(Just a note: all the items shown in the photos, except for the QUAD speakers, are unfinished projects. The QUADs are functional—I included them to give a sense of the level of investment and dedication my father had for this craft.)
*A pair of Quad electrostatic speakers — rebuilt and fully functional
*Sony's VFET R0 Diy Kit
*JVC TT-101
Over many years, he acquired an impressive amount of knowledge, technical skill, and understanding, which he used to restore, upgrade, and build a variety of equipment—from amplifiers to speakers, turntables, and more. Before he became ill, he had started working on several projects, but sadly had to stop midway, and now they sit unfinished with an uncertain future.
I’ll be the first to admit—I don’t have the expertise my father had. What I do share with him is a deep love for music, which was one of the strongest bonds between us. But unfortunately, he didn’t pass down his technical skills to me.I’m posting here because I feel that my father left me with a mission—his legacy, if you will. And I’ve made it my goal to try and complete the work he couldn’t finish himself.
I’m looking for a mentor, a guide—someone who can help me understand where to even begin with these projects that meant so much to my father. I’ve attached some photos of what he left behind, in the hope that someone might see the value in this and, out of love for this beautiful world of stereo and sound, help me breathe life back into a dream that’s been paused.
(Just a note: all the items shown in the photos, except for the QUAD speakers, are unfinished projects. The QUADs are functional—I included them to give a sense of the level of investment and dedication my father had for this craft.)
*A pair of Quad electrostatic speakers — rebuilt and fully functional
*Sony's VFET R0 Diy Kit
*JVC TT-101
Dipole Bass for the Quad ESL-63 (Hi-Fi World 1995/1996 DIY Supplement)
- By bolserst
- Planars & Exotics
- 4 Replies
While watching a video on the history of Quad, a magazine cover showing an ESL63 atop an SL-6000 dipole sub caught my eye.
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I was able to locate the articles online and thought other ESL63 fans might enjoy reading them...if nothing else, for historical perspective.
Enjoy!
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I was able to locate the articles online and thought other ESL63 fans might enjoy reading them...if nothing else, for historical perspective.
Enjoy!
The new DDDAC1794MK3 (DAC and PSU) - Technical and support thread
- By dddac
- Vendor's Bazaar
- 99 Replies
As I just released the DDDAC1794MK3 series (Dac and PSU) I thought it might be a good idea to set up a support thread for technical and background question, discussion and support.
The release post is here: http://blog.dddac.com/the-new-dddac1794mk3-dac-and-psu/
For those already interested in the construction and installation manuals:
DAC: http://blog.dddac.com/wp-content/up...tion-DDDAC1794MK3-DAC-Version-2.6-rev-1.4.pdf
PSU: http://blog.dddac.com/wp-content/up...ction-DDDAC1794MK3-PSU-Version-2.5-rev1.4.pdf
I am not a great photographer, nor do I have a product photobooth, so here is a somewhat lousy picture of my test set-up:
looking forward to some good discussions 🙂
Doede
http://blog.dddac.com/
The release post is here: http://blog.dddac.com/the-new-dddac1794mk3-dac-and-psu/
For those already interested in the construction and installation manuals:
DAC: http://blog.dddac.com/wp-content/up...tion-DDDAC1794MK3-DAC-Version-2.6-rev-1.4.pdf
PSU: http://blog.dddac.com/wp-content/up...ction-DDDAC1794MK3-PSU-Version-2.5-rev1.4.pdf
I am not a great photographer, nor do I have a product photobooth, so here is a somewhat lousy picture of my test set-up:
- Power transformer 2x12 Vac
- double DDDAC PSU
- two AE Chokes (my specs: Amorphe 1H 4 Ohm)
- DAC Board DDDAC1794MK3
- Sowter Output Transformers
looking forward to some good discussions 🙂
Doede
http://blog.dddac.com/
Roederstein (ROE) EK Series capacitors datasheet
- By KleinesRotes
- Parts
- 0 Replies
Greetings!
Here is a link to a German language catalog containing the datasheet for the Roederstein EK Series capacitors. See pages 32-37:
https://archive.org/download/katalo...ektrolyt-Kondensatoren - Gesamtkatalog_v3.pdf
I would have added this to the following thread, but it is closed for replies:
To demystify the detail of its body construction: 'The capacitor winding is mounted in a pot-like plastic cup and sealed hermetically with cast resin.' (original text: "Der Kondensatorwickel ist in einem topfartigen Kunststoffbecher montiert und luftdicht mit Gießharz verschlossen.")
Here is a link to a German language catalog containing the datasheet for the Roederstein EK Series capacitors. See pages 32-37:
https://archive.org/download/katalo...ektrolyt-Kondensatoren - Gesamtkatalog_v3.pdf
I would have added this to the following thread, but it is closed for replies:
To demystify the detail of its body construction: 'The capacitor winding is mounted in a pot-like plastic cup and sealed hermetically with cast resin.' (original text: "Der Kondensatorwickel ist in einem topfartigen Kunststoffbecher montiert und luftdicht mit Gießharz verschlossen.")
Need help increasing input sensitivity of Harman Kardon Citation B power amp
- By crooner
- Solid State
- 18 Replies
Attachments
15W Poweramp Folded Cascode and Laterals
- By lineup
- Solid State
- 24 Replies
What is unique is the use of folded cascode.
Simulation shows THD 0.00032% but what the real value is we can not know.
Let us say the distortion should be low.
Folded cascode makes the input pair drive the output.
Because EXICON Laterals are easy to drive, it works.
Simulation shows THD 0.00032% but what the real value is we can not know.
Let us say the distortion should be low.
Folded cascode makes the input pair drive the output.
Because EXICON Laterals are easy to drive, it works.
Simple, no-math transformer snubber using Quasimodo test-jig
- By Mark Johnson
- Power Supplies
- 2722 Replies
Attachments
Home grown digital sub filter
- By njswede
- Digital Line Level
- 14 Replies
I need some brain gymnastics this weekend, so I was thinking of designing (from scratch, of course) a digital sub filter for the sub woofer I'm currently working on. I have a Teensy4.1 with an audio shield that should be up to the job.
My DSP-fu is a bit rusty, though, so maybe someone can do a sanity check?
Filter. I want to use a FIR due to its linear pass band phase (and therefore constant group delay). Unfortunately, very low frequency LP filters require extremely long FIRs, so I'm going to use the old decimate-filter-interpolate trick. I first decimate by 5, apply my filter, but with a 5 times higher relative cutoff frequency. Then I filter it again to interpolate. I experimented a bit it Octave, and it looks like 100 taps for the first filter and 50 for the second one yields decent results without using too much compute power. I'm not too worried about the aliasing bumps in the stop band, as they sit at about -55dB.
Next I need some kind of phase correction. And correct (no pun intended) me if I'm wrong, but it's really the group delay and not the phase I want to correct. So instead of fiddling with all-pass filters, I should just use a simple digital delay, right?
And yes, I know I can buy a perfectly good module for this at AliExpress for less than a the price of a cup of coffee. But what's the fun and learning experience in that?
My DSP-fu is a bit rusty, though, so maybe someone can do a sanity check?
Filter. I want to use a FIR due to its linear pass band phase (and therefore constant group delay). Unfortunately, very low frequency LP filters require extremely long FIRs, so I'm going to use the old decimate-filter-interpolate trick. I first decimate by 5, apply my filter, but with a 5 times higher relative cutoff frequency. Then I filter it again to interpolate. I experimented a bit it Octave, and it looks like 100 taps for the first filter and 50 for the second one yields decent results without using too much compute power. I'm not too worried about the aliasing bumps in the stop band, as they sit at about -55dB.
Next I need some kind of phase correction. And correct (no pun intended) me if I'm wrong, but it's really the group delay and not the phase I want to correct. So instead of fiddling with all-pass filters, I should just use a simple digital delay, right?
And yes, I know I can buy a perfectly good module for this at AliExpress for less than a the price of a cup of coffee. But what's the fun and learning experience in that?
Harvesting CDM9/44 laser assemblies...
- By dowser
- Digital Source
- 3 Replies
** Cross-posted on Pink Fish Media **
Just picked up another 2 of these Kodak PCD-860 Photo/CD players to add to my collection (I now have 4). A perfect supply for CDM9/44 laser assemblies, which the laser assembly is identical to the CDM9/Pro laser fitted to my 2 Naim CDIs (and many other high end players).
They are a perfect source of low mileage lasers - a photo cd player was something that never really caught on, so any out there are almost bound to be little used. Three of the 4 I have all displayed the exact same faults;
- Tray belt stretches so doesn't close tray fully
- This means laser mech is not in right position and it slams the upper disc clamp down onto the disc platter off centre, pushing the shitty plastic platter down onto the mech
For the 2 I've just bought I have just forced platter up .2-.4mm, fited a temp belt to test laser - both working perfect. Now I strip the lasers and throw the rest away (a little guilty for the landfill, but apart from laser they're a crap machine!).
Look at these belts - proof of lack of use for a few years 🙂
Sharing this info now as I now have 6 suitable laser head donors for my 2 CDIs - should see me out for my life 🙂 If you're after good 9/44s, keep your eye out for these (recent 2 cost me £17 each, they're always cheap).
Richard
Just picked up another 2 of these Kodak PCD-860 Photo/CD players to add to my collection (I now have 4). A perfect supply for CDM9/44 laser assemblies, which the laser assembly is identical to the CDM9/Pro laser fitted to my 2 Naim CDIs (and many other high end players).
They are a perfect source of low mileage lasers - a photo cd player was something that never really caught on, so any out there are almost bound to be little used. Three of the 4 I have all displayed the exact same faults;
- Tray belt stretches so doesn't close tray fully
- This means laser mech is not in right position and it slams the upper disc clamp down onto the disc platter off centre, pushing the shitty plastic platter down onto the mech
For the 2 I've just bought I have just forced platter up .2-.4mm, fited a temp belt to test laser - both working perfect. Now I strip the lasers and throw the rest away (a little guilty for the landfill, but apart from laser they're a crap machine!).
Look at these belts - proof of lack of use for a few years 🙂
Sharing this info now as I now have 6 suitable laser head donors for my 2 CDIs - should see me out for my life 🙂 If you're after good 9/44s, keep your eye out for these (recent 2 cost me £17 each, they're always cheap).
Richard
DAC PCB for DSP
- By Monstercore
- Digital Source
- 0 Replies
After 20 years I want to upgrade my active crossover (dcx2496). I do not need more functionality but upgrade noise and thd specs.
Looking around I find dbx pa2 and venu360 but they have almost the same specs.
So I decided to build an active crossover/dsp myself with some ready made modules(pcb).
DSP I like to use an adau1467 processor using it for 2 channel in and 8 channel out.
For output dacs I have seen boards below, does anyone experience with them?
I think the pcm1798 with OPA 161X opamps has the best specs.
For that price I can not build it myself.
Looking around I find dbx pa2 and venu360 but they have almost the same specs.
So I decided to build an active crossover/dsp myself with some ready made modules(pcb).
DSP I like to use an adau1467 processor using it for 2 channel in and 8 channel out.
For output dacs I have seen boards below, does anyone experience with them?
- CS4398
- PCM1798
I think the pcm1798 with OPA 161X opamps has the best specs.
For that price I can not build it myself.
Easy way to find port (or passive radiator) tuning frequency. Test your subwoofers
- Subwoofers
- 1 Replies
Here is the method
1) Install Spectroid app on your Android phone from Google playstore
2) Ensure there is relative silence while doing this test, we want to keep the noise floor down.
3) Prepare to run frequency sweep from youtube Login to view embedded media This is a sweep from 25Hz to 150Hz. You may use any other way of sweep if you want to go lower than 25Hz. Pause the video at start.
4) Bring your android phone mic as close to the port as possible keep 1 cm of distance but dont touch the speaker or sub with your phone. The idea is that the mic should mostly catch the audio from port only and other sounds should appear as noise coming from far.
5) Start the app
6) Run the sweep from youtube, when it completes, notice the peak in response in bass in the red curve of the app, that's your tuning frequency
7) Repeat above to be sure.
Note:- Android phone mic is not a good way of measurement but for this purpose, it works
Below is my speaker's 60Hz tuning
Hope it helps.
Warm Regards,
WonderfulAudio
1) Install Spectroid app on your Android phone from Google playstore
2) Ensure there is relative silence while doing this test, we want to keep the noise floor down.
3) Prepare to run frequency sweep from youtube Login to view embedded media This is a sweep from 25Hz to 150Hz. You may use any other way of sweep if you want to go lower than 25Hz. Pause the video at start.
4) Bring your android phone mic as close to the port as possible keep 1 cm of distance but dont touch the speaker or sub with your phone. The idea is that the mic should mostly catch the audio from port only and other sounds should appear as noise coming from far.
5) Start the app
6) Run the sweep from youtube, when it completes, notice the peak in response in bass in the red curve of the app, that's your tuning frequency
7) Repeat above to be sure.
Note:- Android phone mic is not a good way of measurement but for this purpose, it works
Below is my speaker's 60Hz tuning
Hope it helps.
Warm Regards,
WonderfulAudio
Adding a tweeter to a full-range
I'm using 4" full-range whizzer cone speakers in my bedroom. They operate normally. No lack of treble. However, one day, I found the NOS tweeters of the same series with those full-range are being sold. I think what if I buy and add them to the 4" full-ranges.
Of course, the HPF for the tweeter is required. But, how about LPF for the full-range, isn't it required?
If so, the x-over point might be set at around 4-5kHz (no problem). But, if not, which means running the full-range (closed) to 20kHz as usual and assigning somewhere upper frequency for the tweeters to handle, then, what frequency should be set for HPF of tweeters?
PS. I have a DSP in my system so I can actively control x-over.
Of course, the HPF for the tweeter is required. But, how about LPF for the full-range, isn't it required?
If so, the x-over point might be set at around 4-5kHz (no problem). But, if not, which means running the full-range (closed) to 20kHz as usual and assigning somewhere upper frequency for the tweeters to handle, then, what frequency should be set for HPF of tweeters?
PS. I have a DSP in my system so I can actively control x-over.
For Sale Universal Power Supply PCB for TO 220 Voltage Regulators
For lower power projects like preamps, headphone amps, etc. I like to use power supplies with TO220 voltage regulators.
I designed a universal board for this.
To power a recently built OREAD, among other things, I had some new supply boards made and now have some spare boards for sale.
The board can be used with TO220 voltage regulators like LM317/337 and 78xx/79xx or others with the same pinout.
It also allows the use of: fuses, RC-snubbers for transformer secondary windings, rectifiers like 1N54xx for full 1.5A current, CRC filtering, suitable heatsink,
soft start for output voltage.
The circuit below shows values for +/-15 volts output. A BOM (example/proposal) will be sent to buyers via email.
this version sold out. new version available, see last post.
If interested: Email/PM me
I designed a universal board for this.
To power a recently built OREAD, among other things, I had some new supply boards made and now have some spare boards for sale.
The board can be used with TO220 voltage regulators like LM317/337 and 78xx/79xx or others with the same pinout.
It also allows the use of: fuses, RC-snubbers for transformer secondary windings, rectifiers like 1N54xx for full 1.5A current, CRC filtering, suitable heatsink,
soft start for output voltage.
The circuit below shows values for +/-15 volts output. A BOM (example/proposal) will be sent to buyers via email.
this version sold out. new version available, see last post.
If interested: Email/PM me
Yet another "opamp + transistor" SE headphone amplifier
- Headphone Systems
- 5 Replies
Not sure that it actually needs the servo, but I was not happy with 18mV worst case DC offset at the output.
Seems to simulate pretty well.
Seems to simulate pretty well.