Philips DPM7 digital amplifier 1987 review

In this year, digital fever will truly take hold: CD, digital cassette, digital satellite radio - so why not operate amplifiers with bits instead of analog voltages? This opens up entirely new possibilities for shaping sound. We got a glimpse of what's to come with a Philips prototype.

Several hi-fi manufacturers have already announced their first digital amplifiers, including Denon, Kenwood, Sony, and Yamaha. These devices will be available by the time of the Berlin Radio Exhibition at the latest. However, this represents just a tentative step towards digitization: the amplifiers will have digital inputs for CD players and DAT recorders, perhaps even for satellite tuners. Source selection between these inputs will occur at the digital level. Afterward, the signal goes through a digital-to-analog converter - the rest of the amplifier operates using conventional analog technology.

Not much is gained by this approach. However, one thing could be interesting: if the devices are capable of converting between different sampling frequencies at the digital level, then CDs could be transferred directly to DAT tape without an analog detour. Of course, assuming the copy protection is bypassed...

The Philips amplifier we're dealing with here pushes digitization much further: it can manipulate sound in various ways without relying on noisy and distorting analog components. For this purpose, it samples the analog input signals at 44.1 kilohertz and quantizes them with 16 bits. Additionally, there are digital inputs for CD players and DAT recorders.

At the heart of the sound section are five highly integrated circuits, the so-called "audio signal processors." They fulfill three different functions: dynamic control, reverb, and equalization.

Tailored Dynamics

Dynamic control means either compression or expansion. Compression adjusts the original dynamics of, for example, a symphony orchestra to the listening conditions in a living room or even a moving car. The dynamic range in a concert hall reaches about 80 decibels. If one were to reproduce a symphony at home with the original dynamics and volume, the pianissimo passages would be drowned out by ambient noise, while the fortissimo passages would exceed the neighbor's tolerance threshold. The compressor raises quiet passages and lowers loud ones, with the compression ratio selectable in three stages. For copying to a cassette for the car, the strongest compression would be engaged.

However, nearly all commercial music productions already have significantly reduced dynamic range. Therefore, it may be desirable during playback to expand the ratio between loud and soft, which the Philips amplifier also offers in three selectable stages. Even the decay time of the control process can be adjusted. Compander systems are already known from conventional technology, such as Dolby NR, and they haven't earned the best reputation among hi-fi enthusiasts. However, digital technology has an advantage because it can temporarily store the instantaneous values of the music signal and adjust the amplification during this time.

Nevertheless, we weren't particularly convinced by the sound result: during expansion operation, for example, voices swell unnaturally. How would the electronics know at which point the sound engineer adjusted the volume during recording? If the sound engineer's interventions were recorded and transmitted as control signals, then the original dynamics could be accurately restored during playback. Such a method has been proposed by the Institute for Broadcasting Technology and is awaiting implementation. A hardware and software company like Philips could make a start there.

Reverb through Digital Delay

Digital technology is far superior to analog when it comes to generating reverberation. It can delay a signal without disturbing side effects. However, for the reverberation to sound natural, three basic conditions must be met: the first reflection must arrive after a certain delay, subsequent reflections must follow at statistically distributed intervals, and they must be frequency-dependent. Because high frequencies are generally more absorbed in a room than low frequencies. All of this is considered in the Philips device. Additionally, the reverb time, the ratio of reverberation to direct sound, and the frequency response of the reverb can be adjusted. The time until the first reflection arrives is fixed at 80 milliseconds - equivalent to concert hall dimensions. Bathroom effects cannot be simulated.

A disadvantage is that the reverb is mixed into both stereo channels; quadraphonic technology could offer much more here. Nevertheless, the adjustable reverb effects are quite interesting. By changing the reverb intensity, one can - like with a zoom lens - bring the stage closer or push it further away. In the "100%" position, the listener seems to be sitting somewhere far back in a huge hall. Increasing the reverb time, on the other hand, keeps the distance between listener and stage constant but expands the space - especially behind the stage.

Equalization without Phase Errors?

A rumor precedes the third function of the Philips processor, frequency equalization: that digital filters have no phase shift. Unfortunately, we have to deny this. Our measurements show that the DPM-7 rotates the phase just like any analog equalizer! The device divides into ten octave bands with a plus/minus twelve decibel adjustment range. The lower five octaves can be shifted in their center frequency by a third up or down - a step towards parametric equalizers. However, room resonances cannot be specifically compensated for in this way because they are usually much narrower than an octave, and their center frequency would need to be hit exactly.

The Philips processor can remember any combination of dynamic, reverb, and equalizer settings in a quadruple memory. For the future, even automatic equalizer calibration is envisaged, perhaps via a microphone built into the remote control unit. And correct volume adjustment can also be achieved with the digital "audio signal processors." At the current stage of development, the Philips amplifier operates with an analog volume control, but it is controlled in digital steps by a touch button. The rest of the device corresponds to the Marantz amplifier PM64.

Experimental model showcasing the amplifier of tomorrow.

As mentioned: a study object, not the prototype of a production model. A lot of fine-tuning is still required before such a device can go into production. One issue to be resolved is the adjustment of analog inputs to the dynamic range of the analog-to-digital converter: if it is to be used optimally, one cannot operate with arbitrary input voltages. Will future amplifiers have an "input level control"? The first series amplifiers with digital technology have already been announced for testing - it will be exciting to see.