Denis31 a écrit:Les ordres de grandeur sont très différents entre l'amortissement d'une salle (1s) et celui d'un couple ampli-enceinte (1ms), et alors les problèmes de temps dus à la propagation de la CR dans 10cm de cuivre n'en parlons même pas (1ns)... c'est vraiment dommage de faire de tels amalgames, ça me fait penser à un moyen-age de la hi-fi.
oui peut etre, je sais aussi que les bon techniciens doivent savoir faire l'effort de se mettre a niveau du peuple et peuple je suis
je suis victime de ma passion, sans pouvoir l'expliquer au plus haut point technique, j'ai cependant énormément essayer et écouter d'appareil, je me place donc comme un technicien candide mais avec une experience de terrain grande. et c'est pour moi en fait assez penible de n'avoir le niveau technique, mais je compense avec l'ouverture d'esprit, chose que je prefere de loin a un technicien qui a ses references et qui ni deroge pas jusqu'a qu'un autre technicien lui prouve le contraire.
si je comprend bien vous dite qu'un ampli doit disposer de l'impedance de sortie la plus faible et ce peut importe l'enceinte qu'il drive, j'ai pour ma part sur le terrain fait des expériences inverses ou l'affaiblissement de celle-ci permettait au systeme de mieux sonner, on appel ca l'effet de compensation, ou un defaut induit peut devenir une qualité au global. Je ne dis rien d'autre, juste que je trouve cela une tres bonne idée de pouvoir regler ce parametre a la main.
je suis aussi peut-être victime du marketing "malin" de constructeurs, voici par exemple ce que dit Spectron Audio (specialiste de la classe D depuis plus de 30 ans) sur les boucles de contre-reaction des amplis "classiques"
How Feedback is implemented is the Key to Outstanding Sound in a Class-D Amplifier: A power amplifier is a voltage control system with sufficient current delivery capability to maintain the output voltage no matter what the speaker load. So let's look at it from that point of view. Power amplifiers simply take an input voltage and amplify it by a factor of 20 to make the output voltage high enough to drive the speaker. When you load the amp with a speaker, the current drawn by the speaker causes the output voltage to fall. So the entire key to low distortion and flat frequency response, in other words the accuracy of the amplifier, is control of the output voltage. This is where feedback comes in. Control systems are implemented by feedback loops.
What is a feedback loop? A feedback loop is a circuit that compares the output to the input and drives the errors in the output towards zero. In general there are three important stages that should be included in the feedback loop. They are the modulator, power section and the output filter. Most class-D amplifiers will have at least some of the stages included in the feedback loop. The Spectron Musician III has all of the stages included in the feedback loop. The stage most often not included in the feedback loop is the output filter. It is by far the most difficult to implement.
Why did Spectron go to the trouble of including the output filter in the feedback loop? The output filter passes the audio signal to the speaker and blocks the high frequency carrier. All filters have group delay errors. Group delay occurs when the various frequencies of an instrument arrive at the listener's ears out of alignment compared with the original recorded sound. Including the output filter in the feedback loop greatly minimizes these group delay errors. All of the harmonics of the music therefore appear at the output of the amplifier with the same time alignment in which they were recorded. Consequently, the four most important advantages of including the output filter in the feedback loop are 1) proper time alignment, 2) flatter frequency response, 3) lower distortion and 4) lower output impedance, which improves speaker damping.
Another very important aspect of a control system is called transit time, the amount of time it takes from the time an error is detected at the input until corrections are made at the output. For example, a typical transistor power amplifier has a three primary sections: a low noise high gain differential input stage, feeding a differential to single ended conversion driven a by a high current output stage. Each of these three stages is designed to have low distortion and noise. These attributes come at the sacrifice of speed. Typical transit time of linear amplifiers will have about 2000 - 3000 nanoseconds which is too slow for effective implementation of global feedback and error correction. This results in ringing artifacts and increased odd-order harmonics which are so detectable to human hearing that even the smallest amount of these distortions are annoying. Delays in feedback also introduce transient and phase discrepancies, susceptibility to transient overload and vulnerability to disturbances at the output, such as reactive speaker interactions. Consequently, feedback has gotten a bad rap in audiophile literature and from magazine writers.
In contrast, the Spectron amplifiers don't use slow distortion circuits, they use very high speed digital logic along with densely packed surface mounted electronics. Consequently, the Musician III transit time is 0.2 uS. Lets illustrate the meaning of this time interval. Since the period of a 20KHz sine wave is 50uS then its ratio to the propagation delays is 50uS/0.2uS = 250. In simple terms, this means that the Spectron control loop is about 1/250 of treble signal period, 1/2500 times midrange signal periods etc. In practical terms, these are near real time speeds which allow the amplifier to correct for smallest errors and the control loop can follow the input with much more precision. It is indispensable to the most accurate reproduction of sound.
la source :
http://www.spectronaudio.com/tech1.htmautre technologie qui me passionne sont les amplis a conversion PCM-PWM (autre dit leurs entrées n'est pas analogique mais numérique) et comment ils ont traité le sujet, concernant le Tact-Equibit ou Lyngdorf , il la simule et l'additionne au signal de sortie, sur le terrain j'ai toujours trouvé que ces amplificateurs étaient vraiment tres bon, supérieur a grossomodo tout les amplis analogique que j'ai eu entre les mains
The very unique thing feature of the Equibit technology is that the PCM to PWM conversion is made without using feedback. Which actually is a necessity since you cannot make a feedback loop taking the analog signal at the speaker terminals and feed it back to the digital PCM signal! That is just not feasible!
So, the Lyngdorf true digital amplifiers are open loop amplifiers - no feedback used at all.
It is quite obvious that developing such an amplifier is a much more complex process. It is simply more expensive since it requires very stable and ripple free power supplies and other special solutions such as the Equibit for the PCM to PWM conversion and extremely linear design of both power supply, output stage and reconstruction filter.
The advantage of this meticulous design is that e.g. the very linear and low distortion simply results in a more musical sounding amplifier. If you consider an acoustic instrument it gives a fundamental tone and a lot of harmonic overtones. For e.g. pianos and violins there is considerable energy in the overtones compared to the fundamental tone. If the distortion versus frequency of the amplifier is not flat (which it rarely the case for a typical switching amplifier) you will add more or less distortion from the fundamental to the natural harmonics and actually destroy the balance of the natural harmonics. However, when the distortion (which, as already mentioned, is very low) is the same at all frequencies you can preserve the natural balance of the music you listen to. We have conducted experiments with this, and actually test persons would prefer higher but linear distortion compared to lower but nonlinear distortion over frequency. So, linear distortion is key to musicality.
la source :
http://www.lyngdorf.com/content/view/64/35/et depuis peu nous avons NAD qui vient de sortir son modele le nouveau M2 :
Most digital amplifiers are open loop and have no correction mechanism, so their performance falls far short of linear amplifiers. They cannot correct for the imperfections that are inevitable in power supplies, or real-life switching waveforms. Attempting to perfect the power supply or switching structures is not a practical approach, so the DDFA technology was developed to implement correction for the problems.Feedback Re-invented An analogue amplifier, whether linear or Class D, can use conventional negative feedback methods to compensate, but the problem is much more difficult for a true digital amplifier. The obvious method of digitizing the analogue output and feeding back to subtract from the input is hampered by the large delays involved and results in an unstable system.
To solve the problem, the Zetex team has developed an entirely new approach, which is best described as noise shaping error correction. Any deviation from the perfectly programmed pulse shape is regarded as an error. This could be caused by the amplitude of the pulse (power supply ripple or sag), the width of the pulse, or even the slope of the edges. Any of these factors will impact the area under the pulse (which is really how the signal amplitude is encoded).
The system operates by comparing the output PWM signal with a high purity
‘Reference PWM’ signal to create an error signal, which is representative of the voltage error at the output. Integration in time provides an indication of the pulse area error, which is digitized at a conversion rate of 108MHz to pass back to the digital domain. The error information is then processed to compensate subsequent modulation cycles. The system can be considered to be constantly adapting to minimize the errors and hence deliver as true a signal as possible to the speaker. The output signal is also monitored at the output of the LC filter, which means the system has amazingly low output impedance. This very tight or direct feedback path gave rise to the name Direct Digital Feedback Amplifier.
source :
http://nadelectronics.com/content/10021 ... EU-Web.pdfa noter que ce nad possède un réglage paramétrique numerique de la "Direct Digital Feedback" pour l'adapter au mieux a son système.
j'ai donc a la lecture de tout cela pensé que la contre-reaction d'un ampli est un paramètre très important pour la qualité d'écoute et mon message original n'etait autre que je trouvais que rendre réglable une boucle de contre-reaction sur un ampli de technologie "classique" était a mes yeux un vrai plus VS la contre-réaction théorique idéale fixée par le concepteur et je n'ai pas compris pourquoi il n'en serait pas ainsi ? et c'est, j'ai l'impression, une des grosse argumentation marketing des amplis Stormaudio ?
et quant a la hifi de moyen-age je crois qu'on a quasiment rien inventé depuis plus de 50 dans l'electro-acoustique (a part les technologies numérique)