48 kHz or 96 kHz
Sampling Frequency ?
An experience studied in the frequency domain showed that above 20 kHz the loudspeaker, or more precisely the tweeter, creates intermodulation.The speaker is therefore no longer a realistic transducer but more an interpreter of the sound; the phenomena of intermodulation modifies the contents of the audio message and therefore diminishes the fidelity of the system.
In the temporal domain
According to this doctor, digital systems currently in use (in the middle of the ‘80s) were unable to go beyond 48 kHz sampling frequency, whence it seemed that the separation ability of our ears could reach 20 microseconds. At the time A/D and D/A converters only operated up to speeds of 48 kHz and without oversampling, we were unable to satisfy the practitioner.Remembering this, it seems evident that the direction of the temporal domain merits a little more exploration, and the answer could be the justification for recording at 96 kHz rather than 48 kHz.
The repetition speed is set to about half a second, which experimentally speaking seemed optimal in order to evaluate the time discrimination of the ear.
The first operation to come to mind is simple: it is sufficient to amplify this double impulsion by means of an amplifier sufficiently rapid to attack a tweeter normally used in quality speakers.
The listening was done by placing
the auditor at a distance of about 1.5m from the tweeter, with one ear facing
the source. The distance of 1.5m is a compromise: closer, we would find
ourselves in a situation of near-field (pressure/speed balance not established),
further, and bearing in mind the dimensions of our listening room, we could be
disturbed by reflections from the walls.
It was therefore possible to create two delayed pulses by simply moving the position of one of the speakers, 7.5mm corresponding to 20 microseconds delay.
Below 50mm of the relative position of the tweeters, the difference in the level of the two pulses heard at a distance of 1.5 m should not trouble the experiment.
As soon as it was set up, it was tested and our suppositions were confirmed: the tweeter was the source of the phenomena. In this last experiment the progressive modification of the relative positions of the tweeters from 50mm to 0mm made no difference to the sound of the pulses. It is once again in the non-linearity of the materials used to build the tweeter where the explanation of phenomena observed in the first and second experiments lies.
Placing a measurement (B+K) microphone in the place of the auditor, we can clearly see the phenomena which appears when approaching the coincidence of the pulses in the first two experiments (Figure 6A, delay is 120 microsec. approx.; Figure 6B, delay is 50 microsec. approx.). However, we do not see it in the third.
Thanks to the non-linearity of the tweeter, figures 6 A/B show that during T2 appears low frequencies not present in the pulses. It is this trail which makes the "clap" sound muffled.
By integrating and performing an FFT (Fast Fourrier Transformation) on the signal, which represents the impulse response (we should say double impulse response) we should obtain a correlation with the observations noticed in the frequency domain as explained in the previous article (NAGRA NEWS #19)
It seems therefore, that we should wait for an improvement in the transducers in order to benefit fully from the performances of a recording system operating at 96 kHz. As an addition to this conclusion I invite you to re-read the article «96 kHz Recording, a door to the future» which we published in NAGRA NEWS #14 (Dec.1996).
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