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  1. Hi John Could be as simple as selecting the optical input on your Oppo UDP205 . It looks like you need to select this from the remote control and find a button called Input, which should then display a menu called Input source Menu. Hope that helps.
  2. The CD as you acknowledge, simply reflects what has been recorded. As I will explain below the maximum attainable level of CD - where undistorted sound is reproduced, is rarely, if ever attained. So to understand what is attainable in undistorted level we need to know about a subject called companding. Its history goes back to a failed patent of Murray G Crosby which described emphasis and de-emphasis. In the 1960's two manufacturers developed equipment based on Crosby's work, that was widely used in the recording industry, namely DBX and Dolby. Companding is a portmanteau of compression, and expansion. A recording could be compressed when recording and then played back expanded, thereby capturing the available dynamic range. Prior to companding the maximum dynamic range a recording could offer was limited by the tape format to 55db, which as expressed as bits approximately 9.8 bits In the mid to late 1960's DBX companders could attain 75-85db which was very close to the 16 bit standard., so recordings as at say 1969 were equal to the ability of CD players of 1988 with respect to capturing dynamic range. in fact some of the best recordings are from this era, where large reel to reel recorders and either Dolby or DBX companders were used. When digital recording was first made available, the prior 0db levels used with analog tape, had to be reduced to -7db so a step backward occurred with many recording studios and recordings they could offer. If the recording engineer attempted anything over -7db they were greeted with very harsh overload by the recorder. Indeed some recording engineers worked out that they could in some cases marry the benefits of companding once again with digital equipment. DBX type 1 enables this. But digital had, and still does have very specific needs where the analog to digital conversion takes place in order to regain the 16 bit ability analog previously enjoyed. https://warehousesound.com/r/dbxTYPEIV.pdf I would need to know more about CD manufacturing equipment before proceeding, but I strongly suspect recordings are at least 15 years ahead in capturing dynamic range, than the commercial CD's we purchase. At a guess replay equipment we presently use with possibly 105db ability in some cases matches an available CD, because of dynamic range losses when the CD is manufactured. We can partially blame ourselves too, for not adopting companding in the CD era for domestic use. But there is still opportunity to do that if we begin to show interest.
  3. A electronic test signal, whether it be sine wave triangle wave, or square wave or other variants is just that, a test signal. If we are to get closer for audio equipment to be better at reproducing actual music, a music signal needs to be what is compared to. This raises an interesting observation that certain electronic music can indeed get close to test signals. Lou Reed's Metal Machine music as example is the most audacious attempt I know of at presenting sound that has little if any relationship to conventional music as we know it, therein it has a certain beauty of its own. Audio component testing can certainly use test signals, but should also include comparison to music and indeed live instruments if their budget stretches that far, I am sure we would have better equipment made if the real thing, was being compared to.
  4. You are being asked to accept a different interpretation in this case sine waves - but at the same time being asked to apply that different interpretation once you are happy with it, to.... wait for it.... be somehow applying to music ? No, test signals will always be test signals which to the smallest fraction possible cannot account for, and likely never will account for, the complexities within music. It is better to learn the sound of instruments and all of the intricacies like harmonic structure reverberation etc, by going to live music recitals - and never consider test signals as properly capable of testing audio equipment. A proper testing facility if piano is the instrument chosen, would have one permanently available to measure, and compare against, a bit like this: http://www.classicspeakerpages.net/IP.Board/index.php?/topic/869-live-versus-recorded-truth-in-listening/ As for level, with enough refinement to the electronics used for testing, we may see peaks captured in the recording much higher than the display reveals. Partnering equipment you use should have as little change to the audio available from the source, as is possible, prior to amplification, in order to pass on the same attributes in the recording. Insight into the utmost care needed to initially capture dynamics and then attempt their replay, is described here: https://warehousesound.com/r/dbxTYPEIV.pdf which shows replay equipment, thankfully is behind the capability of some recordings. It would be a worry, if it was the other way around.
  5. The two are though false , What makes easier to test, is not necessarily then properly testing the system, rather it has appealed to another sensibility, that of comprehending different information unrelated to music before you. You are led to believe the test signal and its verification, can equally apply to hearing music, which is entirely false. The second. Indeed yes might be found on test CD's but once again it is unrelated to the complexities of reproducing music. It is similarly false to assume test signals at reference 1khz 0db have relevance to matching to other equipment, simply because such levels and that form of test, do not occur in music CD's Here is what a 1Khz Sine wave looks like Rather you need a reference recording of I would suggest a few minutes on a piano, piano happens to contain some of the most complex ability of interpretation. Firstly make effort to hear piano being played well in a live environment. The environment is critical too because unlike test signals there is added information being presented to you - namely reverberation, decay, harmonic structure, the construction of the piano itself, the way the artist approaches and plays the instrument. In a few weeks you should be able to hear distinct differences between a Steinway https://www.steinway.com/ and a Bosendorfer https://www.boesendorfer.com/en/ as well as differences between artists those who play them well. And here is piece of piano music by Liz story The added information qualifies audio equipment as being faithful in ability, whereas without reverberation decay harmonic structure etc, we are being provided limited information like the first image, that is quite alien to our hearing ability, and being asked to make false assumptions of ability to reproduce music, without that much needed information. A handful of pianists for you to hear on CD are Jon Balke - Book of Velocities , Anat Fort - A long story , George Winston - December, Liz Story - Unaccountable Effect For actual testing of a CD player you will need a reference recording of one of these and audacity will do to begin with, as it displays in graphical form the piano recording - that similarly has captured reverberation, decay and harmonic structure. From the output of the CD player you then will need a line in recording to audacity to compare the CD player to the first graphical representation. You can also unlike test signals that are alien in comparison ,subjectively assess each by listening.
  6. You can arrange various test signals such as sine waves at levels such as ~2.0 V to be recorded - then playing back from a CD player test linearity using a oscilloscope for deviation from ideal. The point I was making though is levels like ~2.0V are NOT available from commercially available CD's where to allow for handling peaks and offering pleasant if not outstanding reproduction of music, levels are deliberately kept much lower. A interesting point to consider is if we use alien levels and signals for testing CD players, when we expect them to also following testing to then reproduce music, would we have been better using music and painstakingly examining linearity of say a single piano note in the first place? I think the latter, as the further we distance ourselves from our hearing and natural perception of musical instruments , the objective measurement using alien wave forms and high levels , loses all credibility to then apply to be capable of comparing to what we hear. To demonstrate this - at low level - listen to the sine wave - then a few minutes later a piano note at natural level, which is a valid test and which is invalid ?
  7. Exactly, the point is the potential of a CD players analog output stage to swing voltage into a given load, is always limited and controlled by the level that is recorded on the CD. The players can state much higher figures as potential to swing into a given load, but it never happens because the CD itself limits attaining such high levels You can begin to see the much lower level CD's really attain by allowing a computer program called audacity to sample a CD track and to display the level. Here is Vangelis 1492 track 12. The RMS scale value shown in light blue is not attributed as voltage, but is exceedingly close when checked with a multimeter playing the same track as a CD via the RCA output jacks on a CD player- which also indicates the flac file is the same level as the CD . The peak level is shown in dark blue. We can also say the CD is well recorded in terms of the peaks never flattening and remaining within the allowed scale.
  8. Hi Stuart Manufacturers presently rely on the simple recipe of power amp and speakers , or amp and speakers continuing for many years to come. Your proposal is far superior though, as it gets much closer to what is needed. Rob's build, shows two way active crossovers and Hypex power amps as an example: Thanks for the offer of the cables, I will be in contact.
  9. Hi Stuart Yes the cables are great, I use them constantly thank you, they provide insight into music other cables cannot reach. Re Zobels their use is an admission of a voltage power amplification system where they are absolutely required. The amplifier end Zobel presents as a High frequency load to the power amp and is required to curb parasitic oscillations that otherwise occur- particularly where a speaker load is not present. The speaker lead would otherwise act as an antenna of sorts. Think of what precedes the speaker binding posts and we see circuit board tracks connecting to various rates of transistor or mosfet switching. All solid state amps are unstable to varying degrees when a loudspeaker lead is connected. A zobel at the amplifier end of the cable assists the amplifier to exist and survive when a speaker cable is attached. The reader might ask what happens with Zobels and valve amps ? A valve amp in 99% of cases uses a transformer as its speaker interface and is to the best of my knowledge stable with or without speaker leads. The zobel rarely used at the speaker lead end, we can see acts as a load for the earlier zobel and further properly assists the proper transfer of full frequency audio information, to arrive intact at the speaker binding posts. We stop here, for a moment to pause and reflect that we have a pretty good system of voltage delivery.... but everything changes ... as what follows is a highly reactive speaker. A third zobel exists to try to account for the speaker impedance located inside the speaker cabinet. The passive crossover is with 99% of such designs begun with a inductor - its purpose is to control the back electromagnetic force EMF of particularly the bass driver(s) so pretty alarming compromises occur because we choose to use voltage amplification - the best a voltage amplification system can do is just stiffen that voltage delivery - but the amplifier knows nothing of the speaker. Hence much needs to be done to change the landscape of power amp design, to begin to know the speaker they attach to.
  10. To use adequate source impedance to arrange the complex load of the loudspeaker as part of the amplifier circuit. A good implementation would enable the amplifier to sense how much current and how much voltage, and at what time, it should amplify, based on that load, with comparison to the input signal. https://www.current-drive.info/
  11. In a unbalanced audio system, perfect silence is achievable, so SNR does not contribute. You just have to consider attenuation methods you might not have known or considered. Reinforcing the voltage amplifiers long history, does little to embrace the next page , which will see the amplifier fully capable to sense the loudspeaker, current driven and long overdue, then having capability to correct for speaker non linearity - based on comparison to the input signal.
  12. No, they should - but sadly DO NOT have a zobel on the speaker end of the lead per the attached diagram, for each channel. parts are 18nf capacitor non polarized 100v, a 68R 1 watt, and a 100R 5 watt non inductive type - so NOT wire wound. No, you will have to do this yourself - DIY
  13. In any audio system there are three Zobels needed, one at the amplifier to assist the amplifier with the speaker cable ( note NOT the speaker itself ) a second at the speaker lead end, and the third you refer to.
  14. Sadly 99 % of amplifiers know nothing of the loudspeaker that they attach to. We need to embrace current amplification, and much less so voltage amplification. http://www.customanalogue.com/various/Current-Amplification.pdf There is need of a second Zobel network at the loudspeaker end of the cable - but this is rarely implemented as outlined by Cyril Bateman on page 20 of the cables at AF article. http://web.archive.org/web/20091027010126/http://uk.geocities.com/cyrilb2@btinternet.com/downloads_5.html
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