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Speakers got damaged, can anyone explain the reason plz?

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humblebee

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So, I have Q Acoustics 2020i, Marantz PM5004 amp & MAudio 2496 sound card. The sound card allows for 88.2KHz output.

Some time back I had tried this 88.2 KHz output and had noticed distortion in sound (only in demanding songs like heavy electric guitar & western classical music). Looked to me as if spkrs would get damaged. So I questioned here in the forum and ppl agreed that there should be no distortion on upscaling.
The related thread is here - http://www.hifivision.com/speakers/57629-do-speakers-buzz-when-music-upsampled.html

So I was careful, but a few days back I got the urge and saw that normal bollywood songs played fine. So I set it to 88.2 and off I went. I forgot to switch it back to 44.1 and played classical & rock songs.
Yesterday I noticed that dreaded scratching sound coming from spkrs when volume is changed.

Now, what has happened here? What is your take on the reason for this damage?
Either upsampled music demands higher power and clipping has occurred.
OR as Marantz amp manual says, higher than 44.1 music requires spkrs to be able to handle higher frequency range.
OR maybe spkrs couldn't handle the higher dynamic range.
OR maybe spkrs having paper based drivers (2020i are carbon fibre & ceramic coated paper) were not as sturdy as maybe polypropylene based ones and blew.

Asking the more experienced ones as I will have to buy new ones and I plan to listen to music upscaled to 192khz.

Thanks.
My dear spkrs lasted only a year :-(
 
Last edited:

greenhorn

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my theory is that the sound card/resampling algorithm must have an issue and produced a full RMS power HF signal which would have killed the tweeters.
Upsampling by itself cannot add high frequencies
The tweeters are usually not designed to handle full RMS power output, typically 20-30% of it only.
however, if your speakers are in warranty, and since your amp RMS output (45) is within the speaker recommended RMS range(25-75), you could claim warranty
 
Last edited:

spandan414

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So, I have Q Acoustics 2020i, Marantz PM5004 amp & MAudio 2496 sound card. The sound card allows for 88.2KHz output.

Some time back I had tried this 88.2 KHz output and had noticed distortion in sound (only in demanding songs like heavy electric guitar & western classical music). Looked to me as if spkrs would get damaged. So I questioned here in the forum and ppl agreed that there should be no distortion on upscaling.
The related thread is here - http://www.hifivision.com/speakers/57629-do-speakers-buzz-when-music-upsampled.html

So I was careful, but a few days back I got the urge and saw that normal bollywood songs played fine. So I set it to 88.2 and off I went. I forgot to switch it back to 44.1 and played classical & rock songs.
Yesterday I noticed that dreaded scratching sound coming from spkrs when volume is changed.

Now, what has happened here? What is your take on the reason for this damage?
Either upsampled music demands higher power and clipping has occurred.
OR as Marantz amp manual says, higher than 44.1 music requires spkrs to be able to handle higher frequency range.
OR maybe spkrs couldn't handle the higher dynamic range.
OR maybe spkrs having paper based drivers (2020i are carbon fibre & ceramic coated paper) were not as sturdy as maybe polypropylene based ones and blew.

Asking the more experienced ones as I will have to buy new ones and I plan to listen to music upscaled to 192khz.

Thanks.
My dear spkrs lasted only a year :-(

Are you sure your speakers got damaged? Did you try playing your music through a different source removing your sound card from the chain? Did you try removing everything (Speaker wires, interconnects and power cords) and connecting them back?

Regards,
Sam.
 
Last edited:

humblebee

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my theory is that the sound card/resampling algorithm must have an issue and produced a full RMS power HF signal which would have killed the tweeters.

Now this is a completely new thing and I couldn't have prepared for this.

I also noticed that the scratching sound is coming when 88.2 freq is set and not at 44.1.

This kind of confirms that more power is required at higher frequency.
It makes sense from a technical perspective also.
88.2 freq means more sound waves to reproduce - amp had to reproduce twice the sound waves. Hence it requires more power to operate at the same comfort level.

Something to take care of for everyone who is looking to play music at freq higher than 44.1.
 

arj

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could you answer Sams Q..ie is it really the tweeter which is damaged or is it a soundcard issue ?
IMHO The amp should have no problem with higher sampling as it samples an analogue wave and the speakers will have no problem as well unless there is a dc component or clipping.
 

doors666

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Have you tried updating the sound card drivers. If you have another pc, try it with that, maybe with a linux. if not,, try a different sound card.
 

humblebee

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could you answer Sams Q..ie is it really the tweeter which is damaged or is it a soundcard issue ?
IMHO The amp should have no problem with higher sampling as it samples an analogue wave and the speakers will have no problem as well unless there is a dc component or clipping.

This is what I thought in the beginning as well. But thinking deeper on the problem and reading technical things opened up my mind to this new knowledge which is very relevant. In fact I think I need to make a separate thread and ppl will find it very helpful in many ways.

But b4 that...its not a soundcard issue, drivers are latest, they are time tested rock solid and the main 5" driver is making the noise.

Now onto my point.
Everyone including myself has been thinking that 44.1 makes a perfect wave after recreation from d to a.
BUT
Think in terms of time...it has 44,100 samples per second. So there is no such thing as a perfect wave.
To look at it in another way,
Just imagine that there is one wave of one second duration.
Now we zoom in this wave and if we zoom in very deep we see 44,100 chunks.
But if we zoom in further, gaps will appear.

But the question is why zoom in?
Ans - Because we must zoom in till it is a perfect wave to our mind

This brings us to what you were saying - amp samples an analogue wave by which i assume you mean a perfect wave.

44.1 khz samples mean 0.00002267573sec/sample I.e 22 microsecond for one sample.
But our mind is capable of resolving upto 4 microsecond
Which is roughly 250,000 hz
So a wave is a perfect wave to us if it is sampled at 250khz.

This means I think that amp and thus speakers are putting out waves that are corresponding to 44.1khz input to them.
This also explains why amp is needing more power for 88.2 instead of 44.1.
This mind resolving thing is what makes upsampled music more smooth. And natural.

Excerpt from relevant article - reviewer is talking to Rob Watts of Chord electronics...

If reading up on Watts Transient-aligned Algorithm bends your brain, youre not alone. DAC design is complicated business, especially when it comes to FPGA code. Im capable of explaining only the essence of Watts approach.
I sat down with the man himself flanked by Franks of course during their visit to Sydney in March. In doing his level best to accommodate my rudimentary understanding of matters technical, Watts stressed the importance of time domain accuracy. The rationale here is that our brains are super-sensitive to any inaccuracies in when sounds start and end, errors with which is often what leads some D/A converters to sound dry and tinny.

Redbook/CD-quality audio is sampled at 44.1kHz. Thats 44100 times per second. Dividing a second into 44100 chunks gives us 0.00002267573sec/sample. In other words, Redbook samples are taken at intervals of 22 microseconds (2 x 10^-6 seconds). According to Watts, this falls short of the human brains sampling speed (or inter-aural delay) of 4 microseconds and therefore Redbook data doesnt contain sufficient timing accuracy. What if the transient strikes between two Redbook sample moments?

Watts filter inserts data points between the original samples in order to keep the brain up to speed with important perceptual cues for which proper timing is fundamental. Central to Watts thinking is that the brain contributes as much (if not more) to auditory perception than our ears alone and by reconstructing the original waveform a more natural (and less edgy/harsh) sound presents.

And if Watts interpolation filter sounds a lot like simple up-sampling, it is.


Related article here - Chord Electronics Hugo TT DAC & headphone amplifier review | DAR__KO

Entire Chord DAC range is based on this proposition. (Hugo to Dave)

In fact when we transitioned to red book, and people complained, there were 2 problems technically.
One is this.
Another is limitation of delta sigma which gives max resolution of 5 bits.
Thus ppl love LPs and I miss cassettes.

But if we have a dac that outputs proper 16 bits (LPs give 12bits) at 250khz, we will have gained analogue back.

What do you think?
 
Last edited:

arj

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You are there thinking too much :) test the speaker by feeding sound from Your phone to see if it is the tweeter sampling should not impact it


Sent from my iPhone using Tapatalk
 

MaSh

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This is what I thought in the beginning as well. But thinking deeper on the problem and reading technical things opened up my mind to this new knowledge which is very relevant. In fact I think I need to make a separate thread and ppl will find it very helpful in many ways.

But b4 that...its not a soundcard issue, drivers are latest, they are time tested rock solid and the main 5" driver is making the noise.

Now onto my point.
Everyone including myself has been thinking that 44.1 makes a perfect wave after recreation from d to a.
BUT
Think in terms of time...it has 44,100 samples per second. So there is no such thing as a perfect wave.
To look at it in another way,
Just imagine that there is one wave of one second duration.
Now we zoom in this wave and if we zoom in very deep we see 44,100 chunks.
But if we zoom in further, gaps will appear.

But the question is why zoom in?
Ans - Because we must zoom in till it is a perfect wave to our mind

This brings us to what you were saying - amp samples an analogue wave by which i assume you mean a perfect wave.

44.1 khz samples mean 0.00002267573sec/sample I.e 22 microsecond for one sample.
But our mind is capable of resolving upto 4 microsecond
Which is roughly 250,000 hz
So a wave is a perfect wave to us if it is sampled at 250khz.

This means I think that amp and thus speakers are putting out waves that are corresponding to 44.1khz input to them.
This also explains why amp is needing more power for 88.2 instead of 44.1.
This mind resolving thing is what makes upsampled music more smooth. And natural.

Excerpt from relevant article - reviewer is talking to Rob Watts of Chord electronics...

If reading up on Watts Transient-aligned Algorithm bends your brain, youre not alone. DAC design is complicated business, especially when it comes to FPGA code. Im capable of explaining only the essence of Watts approach.
I sat down with the man himself flanked by Franks of course during their visit to Sydney in March. In doing his level best to accommodate my rudimentary understanding of matters technical, Watts stressed the importance of time domain accuracy. The rationale here is that our brains are super-sensitive to any inaccuracies in when sounds start and end, errors with which is often what leads some D/A converters to sound dry and tinny.

Redbook/CD-quality audio is sampled at 44.1kHz. Thats 44100 times per second. Dividing a second into 44100 chunks gives us 0.00002267573sec/sample. In other words, Redbook samples are taken at intervals of 22 microseconds (2 x 10^-6 seconds). According to Watts, this falls short of the human brains sampling speed (or inter-aural delay) of 4 microseconds and therefore Redbook data doesnt contain sufficient timing accuracy. What if the transient strikes between two Redbook sample moments?

Watts filter inserts data points between the original samples in order to keep the brain up to speed with important perceptual cues for which proper timing is fundamental. Central to Watts thinking is that the brain contributes as much (if not more) to auditory perception than our ears alone and by reconstructing the original waveform a more natural (and less edgy/harsh) sound presents.

And if Watts interpolation filter sounds a lot like simple up-sampling, it is.


Related article here - Chord Electronics Hugo TT DAC & headphone amplifier review | DAR__KO

Entire Chord DAC range is based on this proposition. (Hugo to Dave)

In fact when we transitioned to red book, and people complained, there were 2 problems technically.
One is this.
Another is limitation of delta sigma which gives max resolution of 5 bits.
Thus ppl love LPs and I miss cassettes.

But if we have a dac that outputs proper 16 bits (LPs give 12bits) at 250khz, we will have gained analogue back.

What do you think?


Too much thinking going on. Did you go back to 44.1 and see if there is still disturbance? Did you try another source? Could your sound card be adding Noise while up sampling?

MaSh
 

greenhorn

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Now this is a completely new thing and I couldn't have prepared for this.

I also noticed that the scratching sound is coming when 88.2 freq is set and not at 44.1.

This kind of confirms that more power is required at higher frequency.
It makes sense from a technical perspective also.
88.2 freq means more sound waves to reproduce - amp had to reproduce twice the sound waves. Hence it requires more power to operate at the same comfort level.

Something to take care of for everyone who is looking to play music at freq higher than 44.1.

what I am trying to say is, that the soundcard or the resampling software is creating some unwanted signals while creating the 88.2K signal. a 44.1K source will still be limited to 20Khz no matter how high you resample it.

to give an analogy - if you play a VCD on a HDTV at native resolution without resampling, you will get a very blocky picture. If you upsample it, the blocks will go away, but you will not get any additional detail or clarity.

The same goes with sound.
 
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