LYNGDORF TDAI 2200

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LYNGDORF TDAI 2200 - Amplificador integrado con Room Perfect

The TDAI 2200 true digital integrated amplifier is the most versatile amplifier in our range and in the world. The standard version of the compact 2 x 200W digital control centre can be expanded with analog input module and the outstanding RoomPerfect room correction system.

The TDAI 2200 is a fully digital amplifier from Lyngdorf based on the experience gained during the design of the new Millennium platform. The TDAI 2200 is much more than an amplifier, it is a complete digital signal processing control centre for designing advanced audio systems based around active loudspeakers. Active meaning use of active cross-overs instead of ordinary passive filters.

And don't let the actual size of the product deceive you. The compact design of the TDAI 2200 features a powerful amplifier capable of driving difficult loudspeaker loads effortlessly. Also integrated within is a DSP section with equalization and correction possibilities for speaker position and delay, combined with extreme versatility with regard to processing, making this amplifier an obvious choice for the critical audiophile.

This product replaces simultaneously a D/A converter, a pre-amplifier and a power amplifier, requiring only a CD transport as a source. The TDAI 2200 is full of remarkable features like the adjustable power supply voltage which actually works as the volume control. This ensures you full dynamic range - even when playing at very low volume levels.

The TDAI 2200 is the ultimate combination of tools, toys and audiophile amplification.

True Digital Amplifier
In principle, the output stage of most switching amplifiers has the same working principle.
PWM = Pulse Width Modulation is simply a matter of variable open/close time of the transistors.
Simplified, you can compare the transistors in the output stage to two switches. Pls. see fig 1.

Fig. 1

The longer switch 1 is "ON" (and switch 2 is "OFF") the longer the "positive" excursion of the loudspeaker will be and the longer switch 2 is "ON" (and switch 1 is "OFF") the longer the "negative" excursion will be.
And if the two switches are on for the same period of time within one switching cycle there will be no excursion of the driver (no sound). Pls see fig 2.

Fig. 2

Most PWM amplifiers today are self oscillating which means that the amplitude of the ANALOG input signal decides the switch frequency. This means that the switch frequency is high (> 300k Hz) when the audio input signal level is low, and that the switch frequency is low (= moving towards the audible frequency range) when the audio input level is high = when the output from the amplifier is high.
The advantage of this topology is that it is cheap and simple to design since it uses well proven feedback from the analog output to the analog input partly to create the oscillation and partly to create a low THD amplifier.
The disadvantages are that it requires an analog input signal (= it is really an analog amplifier!). So, the PCM (Pulse Code Modulated) signal from your CD player is converted to analog and then the analog signal is converted to the PWM signal.
Also, the feedback loop typically isn't linear which is why distortion often increases towards higher frequencies.
Finally the fact that the switch frequency moves closer to the audible frequency range (typically it is limited not to go lower than 80 - 100 kHz) when you turn up the volume results in risk for creating an offset working point for the tweeter. Even though you cannot hear it directly, harmonics of a 80 or 100k Hz switch frequency can bias the tweeter.

The Lyngdorf true digital amplifiers use a fixed switch frequency (@400k Hz). Furthermore, we convert the PCM signal directly to the PWM signal for the output stage in a PCM-to-PWM EquibitTM modulator.

Below in fig 3, is a sine wave, in terms of a PCM signal, in the upper panel of the figure. In a PCM signal, each discrete sample represents a specific amplitude. The corresponding pulse-width-modulated signal at the same sampling rate (frequency) is shown in the lower panel.
The magnitude of each PWM sample is described in terms of the pulse width, as opposed to the pulse height in a PCM signal. So, the 24-bit PCM digital audio signal is fed to the modulator where the audio data is up-sampled 4 times. The EquibitTM modulator then translates the up-sampled signal to a PWM signal having the same switching frequency.

Fig. 3

This means that we have an unbroken signal path without sound deteriorating conversions.

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.

The reconstruction filter of a switched amplifier is often a point that is overlooked - partly because the ideal components are expensive and take up board space and partly because the filter is regarded to be way out of the audible frequency range. But even filters placed octaves above audible frequencies affect the linearity within the audible frequencies, so in Lyngdorf amplifiers the output filter is based on a linear ferrite rod construction with low loss components as polypropylene capacitors resulting in a very low distortion and very good linearity in the filter.

Power Supply and Volume Control
The Lyngdorf true digital amplifiers are based on non-feedback construction, which means there are no differences in the THD performance at any frequency when it comes to the output stage. In other words, the distortion is very low and linear.
However, another major influential factor to the THD performance of a fully digital amplifier is the quality of the power supply. With the Lyngdorf power suppliers we have obtained completely identical and very low impedance which gives a completely flat THD at all frequencies. This is simply a necessity for a high performance implementation of a true digital amplifier.

The power supply uses a toroidal transformer and the advantage of this over a conventional C or El core transformer is significantly less magnetic radiation, something which can potentially induce noise (50/60Hz) which would be very audible.
The supplies for the output stage and the microprocessors are kept completely separate since they use separate windings on the transformer.

However, the truly unique aspect of the power supplies in the Lyngdorf true digital amplifiers is that it actually works as the volume control of the product. When you are playing at low volume levels, the voltage supplied to the output stage is low - when you turn up the volume control you increase the voltage, and thereby the output, from the amplifier.
This is implemented as a DC -> DC converter. On the "primary" side full voltage is constantly applied and then the DC -> DC converter switches the voltage needed for a given setting of the volume control to the "secondary" side (= the voltage supplied to the output stage).

In order to be able to turn down the volume fast again, the DC-> DC converter can, quite uniquely, switch the power back to the "primary" side.

The result of this truly unique way of controlling the volume level is that a Lyngdorf true digital amplifier gives full dynamic range from maximum volume and down to the 62dB reading in the volume control (which corresponds to as little as 1V RMS on the speaker terminals). This is due to the fact that when the supply voltage to the output stage is reduced, then you lower both the output signal AND the noise floor - and thus the full dynamic range is maintained - also at low volume levels. Perhaps this is one of the reasons why reviewers often mention that Lyngdorf amplifiers play equally well whether you're playing at high or low volume levels.

DSP
The powerful DSP "engine" allows you to customize a speaker set-up completely to your preferences.
The built-in x-over filter functionality allows you to both control conventional passive speakers, substitute the normal x-over in a 2-way speaker or use it as the x-over filter between main speakers and woofer(s) - e.g. in a Lyngdorf Audio 2+2 set-up.

Also, multi-way systems can be controlled by using daisy-chained Lyngdorf Audio products.
For example you can use a single amplifier as the "master" of the system to handle the high range (treble) and send the low-pass filtered signal to a "slave" amplifier which then powers the midrange but then sends a further low-pass filtered signal to a power amplifier that handles the bass frequencies.

So, when combined with other Lyngdorf products, the DSP allows a wide range of system configuration options. From "simple" rigs to very advanced set-ups!
What's more, the DSP section can also be used for simple but effective equalization since it also features:

1. A global equalizer for overall room equalization
2. Separate equalizers for left/right and main/line channels to compensate for shortcomings in the audio chain.
3. A very powerful "broadband" voicing equalizer allowing you to "tilt" the sonic balance to your personal preferences.
4. Gain control (left/right and main/line channel) for optimum balance adjustment.
5. Delay control (left/right and main/line channel) for optimum time alignment (ensures that the sound from different drivers or speakers will arrive simultaneously at the listening position).

A/D Converter
The TDAI 2200 can be equipped with a state-of-the-art A/D conversion module with 1 balanced and 3 unbalanced inputs. This extends the amplifier from being just a digital control center, and allows it to interface with analog sources. The converter is built to outperform noise levels of most analog sources. Listening to the analog output from a Lyngdorf CD-1 through the A/D converter leaves you asking the question... which is which? This gives an idea of the performance level of the Lyngdorf TDAI 2200.

RoomPerfect
When listening to your audio system you hear the 3 dimensional soundfield - i.e. the direct sound from your speakers as well as all the reflections. The reflections add ambience and thus the sensation of "space" and "air". However, most room corrections systems measure only at the listening position meaning that the correction is based on one-dimensional information.

At Lyngdorf, we solved the problem with getting 3-dimensional room knowledge in a very unique way.

The RoomPerfectTM system is capable of combining the information about the listening position with the information about the energy transport into the sound-field in a wholly new and innovative way.
The measurement at the listening position holds information about your access to the sound-field while the room positions hold information about the 3-dimensional sound-field in the entire listening room.

Thus we can ensure that you achieve perfect sound, irrespective of your listening room, speaker position and listening position. In fact, the vast amount of information gathered about the sound-field allows you to enjoy the benefits of room correction in any position throughout the room.

Amongst other things RoomPerfect can - as the only system in the market - derive information about:

• Room Acoustic properties, modes (peaks and dips in the room)
• Power response throughout the room
• Loudspeaker directivity
• High frequency roll-off
• Characteristics of low frequency roll-off

From the acquired measurements RoomPerfect processes the information and sets up amplitude targets and limiters for the different filters in an all automated process.

RoomPerfect automatically identifies the optimal target curve for your speakers from the information in the measurements and everything is thereafter controlled by the guided setup in the menu system.
A complete calibration typically comprises of 4-6 measurement and can be done in less than 15 minutes!

Also, RoomPerfect allows you to break away from conventional "free space" placement of your speakers.
Normally, you need to place a loudspeaker well away from rear and side walls to secure the flattest possible frequency response. Due to the fact that a traditional box loudspeaker has omni polar dispersion in the bass region, this "free space" placement has a big disadvantage namely that you actually risk "destroying" the impulse response. The reason for this is that you hear both the direct sound from the speaker and later all the reflections from the walls. The reflections are delayed as a consequence of the distance to the walls and will therefore arrive later thus smearing the "attack" of e.g. a drum beat.
If you place the loudspeaker close to the back wall the bass reflections from the wall and the direct sound will arrive simultaneously at the listening position - i.e. the impulse response in the bass region can be improved considerably and RoomPerfect can easily compensate for the uneven frequency response. So, with RoomPerfect, it can actually be an advantage to choose what's normally regarded as a less ideal "close wall" loudspeaker placement as this will not only improve the impulse response. Also, when compensating for the increased efficiency, the load on both amplifier and loudspeaker is decreased whereby less distortion and better headroom is achieved.

The RoomPerfect module comes complete with microphone, cable and microphone stand - i.e. everything you need to make the calibration.

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