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Marantz AV8801 Preamp/Processor Official Owner's thread - Page 264

post #7891 of 11327
No problem Kal I'm honored to be your dart board biggrin.gif I also predict in the future that Audyssey and AV companies will address high res with Audyssey engaged. Even if (I said if) there's no sound.quality improvement, the marketing of this will sell more products.
Edited by comfynumb - 8/27/13 at 6:51am
post #7892 of 11327
Quote:
Originally Posted by Kal Rubinson View Post

Indeed and I have said so.  Today, I will have lunch with Audyssey and we will have much to talk about.

I would love to be the fly on the wall smile.gif
Edited by ss9001 - 8/27/13 at 6:40am
post #7893 of 11327
+1
post #7894 of 11327
good info, the only obvious problem is lack of 12V triggers, 2 triggers is very short for 4 zones.thanks ku7v
post #7895 of 11327
Quote:
Originally Posted by Kal Rubinson View Post

Indeed and I have said so.  Today, I will have lunch with Audyssey and we will have much to talk about.

Please let us know how they respond to this issue.
post #7896 of 11327
Quote:
Originally Posted by Influence View Post


Please let us know how they respond to this issue.

Chris' response to this issue is pretty well available here and elsewhere on the 'net.  If there's anything interesting, it will probably be in my column or at stereophile.com to which I owe my primary allegiance.

post #7897 of 11327
I think it would be possible with the current DSPs in the Marantz 8801 to get two channel to 96KHZ with Audyssey. Perhaps you could discuss that with Chris.
post #7898 of 11327
Quote:
Originally Posted by jmschnur View Post

I think it would be possible with the current DSPs in the Marantz 8801 to get two channel to 96KHZ with Audyssey. Perhaps you could discuss that with Chris.

In all truth , even if it did do 192/24 for 2/ch the odds of me using it or slim to none, as it stands for bluray which there's no downsampling going on so all is well there, as for multi/ch sacd and the fact that you're converting to PCM to use it in the first place there's no loss there as well. I find the bypass FR/FL for certain multi/ch sacd's a golden option as with well placed mains that need not be touched in anyway and allowing the rest to be corrected due to less than optimal placement a win win! So while there may be some and I'm sure not going to say no, if say a firmware or other mod becomes available but I'm so swamped in the feature and playback of the 8801 that it would be icing upon icing. Now what pre/pro would be able to stand in its shadow would be the real question after such an upgrade eek.gif
post #7899 of 11327
For stereo music I prefer Audyssey off. For MC music, Audyssey flat, but Im still trying to get used to the FR/FL bypass. For movies, full audyssey is great!
I agree with audiofan, even if audyssey could work on 192/24 for 2ch, i´d probably not use it anyway!

BTW, I found Dynamic EQ great for late night movies...
post #7900 of 11327
Quote:
Originally Posted by M Code View Post

Hmmm.
Let me get this straight..
We spend $3000 for the 8801, internally the 8801 has superb 192kHz DACs, spend even more $ for 192/96kHz lossless AV software....
Then we turn ON Audyssey and now can only process @ 48kHz... rolleyes.gif
Seems to me we are throwing away significant audio resolution...

Just maybe..
I could accept this trade-off in a lower cost AVR but not a high-end AV processor..
We spend the big $ expecting the best sonic performance...
Something doesn't add up here..

Just my $0.02... 👍😉

Number 1, if you spend $3,000 on the Marantz, you got ripped off
Number 2, AVS search this thread and you will find where the computing power was used elsewhere as comfynumb has already pointed out
Number 3, Plenty adds up, and for the $$ I haven't heard a better AVR.

Quote:
Originally Posted by jmschnur View Post

You know I tried really hard to hear the difference between 96khz pure direct with the release of the Koln Concert and then with Audyssey flat. Back and forth over short and then long sequences . I failed to discern any differences on my system. (Montis , Bryston, 8801, Oppo 103).

So even though I was very concerned about this when I learned about it, I cannot come up with a logical reason to be upset for listening in my system.

I did the same for several other HD ablums. Same results, jazz, classical piano and quartets, and operatic vocals. I had the same conclusions for these .

Case in point. There have been MULTIPLE ABX comparisons of folks listening to higher res music and 48khz head to head. The results were 50/50 correct to incorrect AKA they were guessing. If having that high of resolution is that important to you, bypass audyssey and spend some dough on your room treatments and odds are you can have the best of both worlds.

Quote:
Originally Posted by M Code View Post

Then why all the hype for:
96kHz/192kHz lossless source material. stated my argument here already
Displays that can handle 4k.. 4k on a 126" viewing from 12 feet back will certainly yield better PQ.
Loudspeakers that go to 40kHz.. no point here at all to prove. Total crock if you ask me
Amplifiers that can output 300W per channel... 300 watts per channel depending on the application is more than necessary, but in others might be almost required dpeending on the listening levels
Having a higher quantity of channels such as 11.2, doesn't necessarily mean better quality...
I would rather have superb, sonic performance @96/192kHz in 7.1...
Instead of 11.2 @48kHz.. To each his own here really.

Just my $0.02... 👍😉

.

Quote:
Originally Posted by M Code View Post

See my post #7872, we already have turned Audyssey OFF..

Incredible..
How many think their system sounds better with it ON...
To me the bottom line is that many users run Audyssey and simply accept it settings as gospel..
And have little or/no basis of reference or comparison...
I sure would support some creditable publication or test lab to do some face-offs between the various Room EQ schemes, all that make certain claims. A lab could easily run these by using an AP and outputting sweeps through pre-outs. Also they could compare the proprietary target transfer function of each as these can/will vary greatly..

Just my $0.02... 👍😉

Not everyone does, and I for one understand its limitations, but still prefer it at times for certain reasons.

The problem with a lab really doing a head to head comparison of the different DSP's is there are too many othe variables to count, and human perception is one of the most deciding factors in the end, so these AVR manu's give us several options to set our systems up to our own liking. The minute one of them has something that is 100% set and forget, I will be QUITE skeptical...
post #7901 of 11327
Quote:
Originally Posted by audiofan1 View Post

In all truth , even if it did do 192/24 for 2/ch the odds of me using it or slim to none, as it stands for bluray which there's no downsampling going on so all is well there, as for multi/ch sacd and the fact that you're converting to PCM to use it in the first place there's no loss there as well. I find the bypass FR/FL for certain multi/ch sacd's a golden option as with well placed mains that need not be touched in anyway and allowing the rest to be corrected due to less than optimal placement a win win! So while there may be some and I'm sure not going to say no, if say a firmware or other mod becomes available but I'm so swamped in the feature and playback of the 8801 that it would be icing upon icing. Now what pre/pro would be able to stand in its shadow would be the real question after such an upgrade eek.gif
Quote:
Originally Posted by Kal Rubinson View Post

Indeed and I have said so.  Today, I will have lunch with Audyssey and we will have much to talk about.


While it seems the processing power can readily be made available to handle streams of 192/24 of source material at reasonable cost, the real problems with exceeding 48KHz or so of sampling frequency have to do with how to obtain the data to adjust the source steam to the room and what to use for speakers to play it back. Here are few of the factors which complicate the situation:

If we want use the new and improved correction system to sample at 96 KHz we’ll first need a microphone with a fairly flat frequency response to 48 KHz or accurate calibration files. For reference: John Atkinson uses a microphone which goes to 30 KHz for his speaker measurements. Most recording microphones that I found in a quick search are only rated to 20 KHz. I don’t know the realities, including cost, of supplying microphones with predictable response to 50 KHz or so.

A review of loudspeaker tests in Stereophile shows very uneven on axis frequency response for many models starting as early as 15 KHz, most are uneven by 20 KHz and even a Revel Salon 2 with a beryllium tweeter is starting to show indications of the inevitable hi-frequency resonance peak found in dynamic loudspeakers. The peaks in response often are 10 dB to 20 dB or more. Likely the frequency response of the units falls off quickly after the peak unless unit is particularly poorly behaved and has several peaks. Most loudspeaker specifications that I have seen from manufacturers show high frequency response stopping at 20 KHz or less and often this is not accompanied by a tolerance band or off access information.

The one inch dynamic tweeters used in most loudspeakers start to become very directional at 10 KHz even with the help of some sort of waveguide, the Revel Salon 2 again being an example and one with excellent off access response. Referencing the measurements for other speakers will typically show worse or much worse results. By 20 KHz most tweeters are performing like those laser pointers used for presentations, abet with some sharp off axis lobes. Higher frequencies only exacerbate these characteristics.

Even air complicates this situation. A sound at 20 KHz is absorbed 1.6 dB in 10 feet at 50% relative humidity, 30 KHz 2.9 dB in 10 feet, 40 KHz 4.1 dB in 10 feet and 48 KHz 5 dB in 10 feet. This is actual absorption by the air. It isn’t due to the sound energy “spreading out” due to distance from the source.

There may be more complicating factors in obtaining good data to build a set of response curves for the room to use for then adjusting the sound from the speakers, but these seem to be enough to make the task very difficult.

Consistent, quality microphones with predictable response to 50 KHz or so will be required. Placement of the measurement microphones and the angle of the tweeters to the microphones will be very important. There will be one position; typically directly on the center axis of the tweeter where there will be a maximum and who knows about other positions. Picture three people sitting in front of stereo speakers with the speakers position to “point” at the center person. This center position may get a level of say 20 dB or more for a 40 KHz tone than those positions on either side. Do we optimize for this position only?

In a room of any significant size the already minimal frequency content over 20 KHz will be drastically reduced by the mere absorption of sound by the air. We can’t count on the room to distribute the high frequencies by reflections and cover for the uneven off axis response of the loudspeakers. The air has absorbed too much of the high frequency energy by the time reflections reach the listeners.

Once measurements have been taken within the above limitations then decisions must be made on how to use them to make corrections. One, or likely more, notch filters will be required to tame the tweeter resonances if smooth response is desired on the axis of the tweeter. The data gathered will likely have very large variations in sound pressure levels by location, especially since the wavelength of the over 20 KHz sound is so short compared to the diameter of the tweeter. How should this be averaged?

It seems we are in in uncharted territory when attempting to correct room responses for sound frequencies much over 20 KHz. What do we use for the coefficients for all those correction filters given the realities of creating and measuring sound frequencies above 20 KHz.

All of this doesn’t mean that hi-res recordings don’t have advantages in other parts of the reproduction chain. It is just hard to see how the advantages extend to equalizing rooms above 20 KHz.
post #7902 of 11327
^^
confused.gif
you have excellent information in your post and sound like a very knowledgable guy on mics, room acoustics, but...

your thinking seems to be misapplied when thinking of upsampling a digital signal wink.gif

nobody is talking about measuring speaker or room response at above 20 khz, that would be silly. this is about the sampling rate used to make a digital recording from analog. when a CD is made at sampling rate of 44.1 Khz they aren't measuring the sound at 44.1! that is how often the analog signal itself (20-20 Khz) is sampled to capture as much of the analog waveform as needed to re-create it when it's converted back to analog with a DAC.

that has NOTHING to do with microphones, measuring speakers on or off axis or anything at all related to recording the analog source to begin with the room & speaker response! the sampling rate of the digital recording isn't part of how the mics get the audio or measure the room.

we all know 44.1 Khz is the redbook standard for CD's. hi-rez audio is taking the same analog audio signal and instead of sampling it 44, 100 times per second, it's sampled at 96,000 or 192,000 times per second for PCM and or in the case of DSD, approx 2.8 million time per sec as a 1 bit data stream. this doesn't mean the mic is being used to capture sound at 2.8 million hz!

I'm really surprised at the conclusion you reached about the whole digital process and mic's needing to capture higher than 20 Khz (up to 192 Khz) to make a hi-rez file. those kind of mics certainly aren't needed to make an analog recording and analog is the source for digital recording except DSD which is recorded as a digital stream to begin with.

the higher the sampling rate, as the theory goes, the closer the stairstepped digital signal approximates an analog signal. that's not a technically exact explanation but close enough I hope.

freq response in the room or your ears is NOT the same as how fast the analog audio signals are electronically sampled to make or reproduce a digital recording.

if you were right, you'd only need to measure up to 128 Hz for a 128 k mp3 file and clearly that's not the case wink.gif

and you aren't "equalizing a room" above 20K hz either. again, it's the rate of sampling the original signal. not EQ'ing the room. if the original recording was CD at 44.1 Khz, Audyssey would just send it on thru unaltered. however, if you started with a 96Khz sampled (NOT analog response) digital signal, Audyssey would "clip" it down to being samples at 48000 times per second instead of its original 96000 times per second.

if you upsampled a 44.1 CD with a 192 K DAC, it would take the original 44.1 sampling and resample it at 192000 times per second. the advantage is NOT to capture audible signal above 20 Hz (if that were the case, CD's 44.1K sampling would be too much), because 1) there's nothing audible above much over 20 Khz, and 2) 192 sample rate converters can't create new audible data when it's not there to begin with. the advantage is to electronically sample the analog signal so high, that the sampling rate is way far removed from the audible region, allowing a very gentle digital filter can be used to remove the sampling frequency and reconstruct the analog signal. this can result in less phase shifts and better imaging in the audio you hear. theoretically, up to some point, the higher you sample the signal, the digital filter used to remove the sampling frequency can have increasingly gentler slopes and not degrade phase cues, etc. unlike the old brick wall filters we had in the early days of CD which had negative impact on the sound.

there are upsampling DAC's that can take good old PCM and upconvert it all the way up to 2.8 million times per sec. & higher. Anthem's D2V can do combined upsampling & oversampling to 128X which is 44.1 K X 128 = 5,644,800 the equivalent to DXD DSD.

if you remember the days of oversampling, with 2X, 4X, 8X CD players, then upsampling is similar.
Edited by ss9001 - 8/27/13 at 3:52pm
post #7903 of 11327
^^ nice post ss9001 and a good explanation smile.gif
post #7904 of 11327
Quote:
Originally Posted by comfynumb View Post

^^ nice post ss9001 and a good explanation smile.gif

thanks!

I try. the OP has a lot of knowledge and great information but confused in applying it to the idea of upsampling, at least that's what it looked like to me wink.gif

I'm not a Kal Rubinson or John Atkinson in my explanation but I hope is close enough technically wink.gif
post #7905 of 11327
Quote:
Originally Posted by ss9001 View Post

thanks!

I try. the OP has a lot of knowledge and great information but confused in applying it to the idea of upsampling, at least that's what it looked like to me wink.gif

I'm not a Kal Rubinson or John Atkinson in my explanation but hopefully, is close enough technically wink.gif



No problem. I'm a novice and I understood it perfectly biggrin.gif
post #7906 of 11327
Quote:
Originally Posted by M Code View Post

See my post #7872, we already have turned Audyssey OFF..

Incredible..
How many think their system sounds better with it ON...
To me the bottom line is that many users run Audyssey and simply accept it settings as gospel..
And have little or/no basis of reference or comparison...
I sure would support some creditable publication or test lab to do some face-offs between the various Room EQ schemes, all that make certain claims. A lab could easily run these by using an AP and outputting sweeps through pre-outs. Also they could compare the proprietary target transfer function of each as these can/will vary greatly..

Just my $0.02... 👍😉

Count me in for thinking... make that for observing that my system sounds MUCH better with it ON.

When I first learned of the 48 KHz downsampling issue and reported it to the members of this thread, I was quite disappointed. However after seven months of listening in my highly-prone to room-modes square living room, the results have been nothing short of spectacular. Whenever I playback a recording that I haven't listened to since acquiring the 8801, I still feel the urge to do tests with MultEQ XT32 on and off. I find that even HD 96 kHz DVD-Audio or Blu-ray Audio material and even SACD titles (converted to PCM) benefit greatly from the room correction. Not only are the nasty room modes tamed but there is an increased focus to the vocals and instruments, even cymbals sound more realistic and have more sizzle.

Stereophile gave a "Products of 2011 JOINT ACCESSORIES OF THE YEAR" award to Audyssey's MultEQ XT32, no doubt due to the enthusiasm evoked by Kal Rubinson after he reviewed it on the Integra 80.2. Kal, over the years, has tested and reviewed perhaps more room correction solutions than just about any other reviewer I can think of, including ARC on the Anthem D2. How's that for a basis of reference or comparison...
post #7907 of 11327
Quote:
Originally Posted by jam88 View Post

Count me in for thinking... make that for observing that my system sounds MUCH better with it ON.

When I first learned of the 48 KHz downsampling issue and reported it to the members of this thread, I was quite disappointed. However after seven months of listening in my highly-prone to room-modes square living room, the results have been nothing short of spectacular. Whenever I playback a recording that I haven't listened to since acquiring the 8801, I still feel the urge to do tests with MultEQ XT32 on and off. I find that even HD 96 kHz DVD-Audio or Blu-ray Audio material and even SACD titles (converted to PCM) benefit greatly from the room correction. Not only are the nasty room modes tamed but there is an increased focus to the vocals and instruments, even cymbals sound more realistic and have more sizzle.

Stereophile gave a "Products of 2011 JOINT ACCESSORIES OF THE YEAR" award to Audyssey's MultEQ XT32, no doubt due to the enthusiasm evoked by Kal Rubinson after he reviewed it on the Integra 80.2. Kal, over the years, has tested and reviewed perhaps more room correction solutions than just about any other reviewer I can think of, including ARC on the Anthem D2. How's that for a basis of reference or comparison...

Jam..
Thanks for the input..
On my system...
Audyssey disturbs the musical balance especially the low frequencies...
I listen for smooth low frequencies wthout any peaks or resonances..
IMHO..
If the low frequencies are not properly the rest of the audio spectrum will be out of balance.
We are a strong supporter of multi subwoofers, and here Audyssey fails to deliver the right balance....
When toggleing Audyssey ON/OFF we prefer it OFF even for the movies, though we do agree for movies its effects are more subtle..

Final thoughts..
Audyssey has clearly led the AVR category by being implemented within the Onkyo, Integra, Marantz and Denon products..
Also we bought an Insigma flat display for a 3rd system and the display included Audyssey processing but again we prefered it switched OFF. In the end everyone's ears are different and if one prefers it ON go for it..
But again I wonder why no publication or test lab has run a faceoff between each of the Room EQ schemes..
Credos to Chris and Phil for creating the Audyssey S/W plus all their extensive efforts promoting it...
I am likely in the minority but I do appreciate all of their hard work establishing their products starting from those early days, 10 years back.... 👌

Just my $0.02... 👍😉🔊📀
post #7908 of 11327
It would be interesting to see audio plots of your audio response. Have you run REW or Ominimic mic on your system, with and without Audyssey ? I found that Audyssey got the subs distances wrong and I had to tune the distances for flattest response using omni mic. I have two subs.
post #7909 of 11327
Quote:
Originally Posted by ss9001 View Post

^^
confused.gif
you have excellent information in your post and sound like a very knowledgable guy on mics, room acoustics, but...

your thinking seems to be misapplied when thinking of upsampling a digital signal wink.gif

nobody is talking about measuring speaker or room response at above 20 khz, that would be silly. this is about the sampling rate used to make a digital recording from analog. when a CD is made at sampling rate of 44.1 Khz they aren't measuring the sound at 44.1! that is how often the analog signal itself (20-20 Khz) is sampled to capture as much of the analog waveform as needed to re-create it when it's converted back to analog with a DAC.

that has NOTHING to do with microphones, measuring speakers on or off axis or anything at all related to recording the analog source to begin with the room & speaker response! the sampling rate of the digital recording isn't part of how the mics get the audio or measure the room.

we all know 44.1 Khz is the redbook standard for CD's. hi-rez audio is taking the same analog audio signal and instead of sampling it 44, 100 times per second, it's sampled at 96,000 or 192,000 times per second for PCM and or in the case of DSD, approx 2.8 million time per sec as a 1 bit data stream. this doesn't mean the mic is being used to capture sound at 2.8 million hz!

I'm really surprised at the conclusion you reached about the whole digital process and mic's needing to capture higher than 20 Khz (up to 192 Khz) to make a hi-rez file. those kind of mics certainly aren't needed to make an analog recording and analog is the source for digital recording except DSD which is recorded as a digital stream to begin with.

the higher the sampling rate, as the theory goes, the closer the stairstepped digital signal approximates an analog signal. that's not a technically exact explanation but close enough I hope.

freq response in the room or your ears is NOT the same as how fast the analog audio signals are electronically sampled to make or reproduce a digital recording.

if you were right, you'd only need to measure up to 128 Hz for a 128 k mp3 file and clearly that's not the case wink.gif

and you aren't "equalizing a room" above 20K hz either. again, it's the rate of sampling the original signal. not EQ'ing the room. if the original recording was CD at 44.1 Khz, Audyssey would just send it on thru unaltered. however, if you started with a 96Khz sampled (NOT analog response) digital signal, Audyssey would "clip" it down to being samples at 48000 times per second instead of its original 96000 times per second.

if you upsampled a 44.1 CD with a 192 K DAC, it would take the original 44.1 sampling and resample it at 192000 times per second. the advantage is NOT to capture audible signal above 20 Hz (if that were the case, CD's 44.1K sampling would be too much), because 1) there's nothing audible above much over 20 Khz, and 2) 192 sample rate converters can't create new audible data when it's not there to begin with. the advantage is to electronically sample the analog signal so high, that the sampling rate is way far removed from the audible region, allowing a very gentle digital filter can be used to remove the sampling frequency and reconstruct the analog signal. this can result in less phase shifts and better imaging in the audio you hear. theoretically, up to some point, the higher you sample the signal, the digital filter used to remove the sampling frequency can have increasingly gentler slopes and not degrade phase cues, etc. unlike the old brick wall filters we had in the early days of CD which had negative impact on the sound.

there are upsampling DAC's that can take good old PCM and upconvert it all the way up to 2.8 million times per sec. & higher. Anthem's D2V can do combined upsampling & oversampling to 128X which is 44.1 K X 128 = 5,644,800 the equivalent to DXD DSD.

if you remember the days of oversampling, with 2X, 4X, 8X CD players, then upsampling is similar.

I think you might have misunderstood bigguyca. I highly recommend the following:

http://en.wikipedia.org/wiki/Nyquist_rate

https://docs.google.com/viewer?a=v&q=cache:UkZPz9IV-XMJ:lavryengineering.com/pdfs/lavry-sampling-theory.pdf+&hl=en&gl=nz&pid=bl&srcid=ADGEESixyoVgX-xPcoopzrKvXgJSgsKbcgz5p1CvKcQ_OYi68GQk-n2-c03sgkTsjOjzpiQse8apn8n4mCnUTCV1RwVP_QS4q0eVFZxm29RMAxL3ESYOdjiTErQtkHNIPH9yDxLoB3O9&sig=AHIEtbTBUZW5v4t_8zT4x8ms5zp8QivWeA

The article in the second link has a lot of advanced mathematics in it. I quickly read through them but you could go right to the conclusion. Pretty much in line with some of the points bigguyca made.
Edited by avman09 - 8/28/13 at 4:36pm
post #7910 of 11327
Quote:
Originally Posted by M Code View Post

Jam..
Thanks for the input..
On my system...
Audyssey disturbs the musical balance especially the low frequencies...
I listen for smooth low frequencies wthout any peaks or resonances..
IMHO..
If the low frequencies are not properly the rest of the audio spectrum will be out of balance.
We are a strong supporter of multi subwoofers, and here Audyssey fails to deliver the right balance....
When toggleing Audyssey ON/OFF we prefer it OFF even for the movies, though we do agree for movies its effects are more subtle..

Final thoughts..
Audyssey has clearly led the AVR category by being implemented within the Onkyo, Integra, Marantz and Denon products..
Also we bought an Insigma flat display for a 3rd system and the display included Audyssey processing but again we prefered it switched OFF. In the end everyone's ears are different and if one prefers it ON go for it..
But again I wonder why no publication or test lab has run a faceoff between each of the Room EQ schemes..
Credos to Chris and Phil for creating the Audyssey S/W plus all their extensive efforts promoting it...
I am likely in the minority but I do appreciate all of their hard work establishing their products starting from those early days, 10 years back.... 👌

Just my $0.02... 👍😉🔊📀

That is interesting. I do not take anything for granted and I prefer to have Audyssey on. Bypass worked well for me too but I still like it better when Audyssey is on. Did you ask Audyssey (Chris) for suggestions to make it work better for you?
post #7911 of 11327
hi

suppose there's only a very faint sound coming from the SBR channel of my brand-new MM8077 so Audyssey says there's either no speaker or the ambient noise is too high, and suppose I've used three different pre-amps with the same result and swapped speakers and cables in order to prove that the rest of my equipment is ok - is there still a chance that I'm doing something wrong?

there's really not much doubt that the SBR channel is defective, except of course that it seems unlikely that Marantz would deliver a broken power amp. The amp was received directly from the exclusive distributor for Marantz in my country, and according to my dealer, I'd be the very first customer to ever receive a defective unit, so - are there any other tests I can conduct?

thanks for your insights.
post #7912 of 11327
Quote:
Originally Posted by tombeck View Post

hi

suppose there's only a very faint sound coming from the SBR channel of my brand-new MM8077 so Audyssey says there's either no speaker or the ambient noise is too high, and suppose I've used three different pre-amps with the same result and swapped speakers and cables in order to prove that the rest of my equipment is ok - is there still a chance that I'm doing something wrong?

there's really not much doubt that the SBR channel is defective, except of course that it seems unlikely that Marantz would deliver a broken power amp. The amp was received directly from the exclusive distributor for Marantz in my country, and according to my dealer, I'd be the very first customer to ever receive a defective unit, so - are there any other tests I can conduct?

thanks for your insights.
It's seems to me if you've had the same issue with different preamps and its not your speaker cable or interconnects . Then there's only one thing left and that's your SRB speaker "s" are the common denominator to the problem .
post #7913 of 11327
I purchased the mm8077 last week and had the same "faint" sounds coming from 2 channels. No Audessey in the equation. The dealer validated this and exchanged this to a new unit which works fine ;-)
post #7914 of 11327
Quote:
Originally Posted by tombeck View Post

hi

suppose there's only a very faint sound coming from the SBR channel of my brand-new MM8077 so Audyssey says there's either no speaker or the ambient noise is too high, and suppose I've used three different pre-amps with the same result and swapped speakers and cables in order to prove that the rest of my equipment is ok - is there still a chance that I'm doing something wrong?

there's really not much doubt that the SBR channel is defective, except of course that it seems unlikely that Marantz would deliver a broken power amp. The amp was received directly from the exclusive distributor for Marantz in my country, and according to my dealer, I'd be the very first customer to ever receive a defective unit, so - are there any other tests I can conduct?

thanks for your insights.



It sounds like there's no doubt it's the amp and I would see if your dealer would swap it with a working amp, since it was defective from the get go.
I know you have the 8801, we've talked before. Welcome to the 8801 thread tombeck smile.gif
Edited by comfynumb - 8/29/13 at 4:25am
post #7915 of 11327
Quote:
Originally Posted by comfynumb View Post

It sounds like there's no doubt it's the amp and I would see if your dealer would swap it with a working amp, since it was defective from the get go.
I know you have the 8801, we've talked before. Welcome to the 8801 thread tombeck smile.gif
He stated he used 3 different preamps with the same issue ?
post #7916 of 11327
Quote:
Originally Posted by TheFactor View Post

He stated he used 3 different preamps with the same issue ?



I guess in other words there's no doubt it's the amp because the same channel has no output no matter what he does. That's the way I'm interpreting it anyway.
post #7917 of 11327
Quote:
Originally Posted by comfynumb View Post

I guess in other words there's no doubt it's the amp because the same channel has no output no matter what he does. That's the way I'm interpreting it anyway.
LoL The amp totally slipped my mind that's why I thought it was his rear speakers . Had a brain fart and forgot he was talking about a preamp and not a self powered AVR .
post #7918 of 11327
Quote:
Originally Posted by comfynumb View Post

I guess in other words there's no doubt it's the amp because the same channel has no output no matter what he does. That's the way I'm interpreting it anyway.

Exactly. I've made sure my speakers are working by swapping them, same for the cables.

Now that I know that one other person has had two broken channels in a brand new MM8077, the last bit of doubt has vanished smile.gif

Thanks everybody for your confirmation. It was quite clear when I tried my first Audyssey calibration that the amp was broken, I just didn't want to believe, and I didn't want to risk being labeled as a dumb customer, that's why I did extensive testing that confirmed my suspicion. However, I wanted to be absolutely sure there was nothing else I could try and nothing else that could be wrong.
post #7919 of 11327
Quote:
Originally Posted by tombeck View Post

Exactly. I've made sure my speakers are working by swapping them, same for the cables.

Now that I know that one other person has had two broken channels in a brand new MM8077, the last bit of doubt has vanished smile.gif

Thanks everybody for your confirmation. It was quite clear when I tried my first Audyssey calibration that the amp was broken, I just didn't want to believe, and I didn't want to risk being labeled as a dumb customer, that's why I did extensive testing that confirmed my suspicion. However, I wanted to be absolutely sure there was nothing else I could try and nothing else that could be wrong.
Have you tried switching out different channels on the amp to see which amp or channels has the issue ? Then you'll know for sure .
post #7920 of 11327
Quote:
Originally Posted by TheFactor View Post

Have you tried switching out different channels on the amp to see which amp or channels has the issue ? Then you'll know for sure .

Yes, I also did that, it's the SBR channel. No matter what I do, whatever is attached to the SBR output doesn't output sound (at least nothing that could be used). I wanted to make sure I missed absolutely nothing before I returned the unit. Thanks! smile.gif
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