Yamaha E1010 E1010 Owners Manual Image - Page 12

given Delay Time pushbutton (6A) can actually produce A CLOSER EXAMINATIONOF THE 11 a wide range of delay times. DELAY CIRCUITRY C. The Clock Rate Circuitry sets the speed at which the audio signal segments move from one storage register to the next. The clock is simply a variable oscillator whose frequency may be changed by adjusting a control voltage (i.e., a VCO - voltage controlled oscillator). Higher settings of the DELAY control slow down the clock which in turn slows down the movement of audio through the BBD's, and hence increases the delay time. However, the BBD's can only be slowed down by a finite amount before they cease to function properly; longer delays then require the use of additional BBD's, which is why the Delay Time pushbuttons are necessary. NOTE: The process is analogous to moving a large water tank (the musical program) from one point (the input) to another (the output) by pouring it along a series of buckets (the BBD's)-hence, the term "delay line." The time it takes to move the water from one tank to the other depends on how many buckets are used (the Delay Time) and how fast the water is transferred from one bucket to the next (the Clock Rate circuitry). D. The Delay Time MODULATION circuitry really does the same thing as the DELAY control; it changes the Clock Rate. The only difference is that the Delay Time Modulation circuitry automatically varies the clock rate. The DEPTH control sets the maximum change in clock rate (and hence the maximum change in delay time). The FREQUENCY control sets the speed at which the clock rate is varied (and hence the rapidity of changes in delay time). Post-Delay Conditioning (7) This circuitry restores the original balance and dynamics by doing the reverse of the Pre-Delay Conditioning (5). The previously boosted mid to high frequencies are cut, thereby duplicating the original frequency response while simultaneously reducing any noise in this range which might have been introduced by the Delay circuitry (de-emphasis). The loud passages are made louder and the soft passages made softer, hence restoring the original dynamic range and simultaneously forcing low-level noise to even lower levels (expansion). Very high frequencies are cut out, thereby avoiding any clock noise which might have been introduced in the Delay circuitry (low pass filtering). At the output of this stage, the audio sounds like it did before being delayed, only it is offset in time. The Feedback Path (8) The FEEDBACK control permits a portion of the delayed signal to be re-applied to the delay circuitry. This creates a loop which is useful for special effects. At longer delay times, the feedback is heard as repetitive echoes. At shorter delay times, the feedback creates a series of signal cancellations which are head as "flanging," "comb filtering," "tunneling," etc. At very high FEEDBACK settings, the entire delay line can go into self-oscillation, even without an input signal, and create unwanted howling. Direct/Delay Signal Mixing (9) The MIXING control affects the front- and rear-panels OUTPUT. It determines the proportion of direct and delayed sound fed to the output. Due to phase cancellations, it is possible to achieve certain effects, effects which could not be achieved by feeding direct and delayed sounds to separate amplifier/speaker systems, by electrically mixing direct and delayed sound together in that one output. General Discussion The bucket brigade devices (BBD's) and the clock (oscillator) constitute the heart of the analog delay line. All the other circuits (tone control, pre- and post-delay conditioning, feedback and mixing) are of secondary importance; in fact, similar circuits to these are also used in digital delay lines. The following paragraphs explain how the signal is delayed inside the E1010, why higher frequencies are lost at longer delay times, and how the analog delay technique differs from digital delay techniques. For the purpose of this discussion, assume the input is one cycle of a simple sine - a brief burst of pure tone. Any signal could be used, but this one is easy to visualize. The original audio program (an analog signal) is nothing more or less than a voltage level which varies up and down. A graph of instantaneous voltage level (vertical axis) versus time (horizontal axis) gives the familiar representation of audio waveform shown in Figure 10. Sample and Hold: The Key to Analog Storage As the signal is applied to the first BBD, it is divided into small segments of equal length, creating a continuous stream of program samples. The exact length of each sample is defined by the E1010's clock pulses, and each sample has an average voltage value that is taken directly from the signal present during the corresponding clock pulse. (Refer to Figure 11) These samples are stored (held) to create the time delay. I-4-- 1 CYCLE ----0.-4-- Peak Voltage (1V) 4;1 RMS Voltage '9 (.707V) • Peak-To -Peak • Voltage (2V) TIME If t 1 If t 1 If t1 If t1 - t = 0.5 milliseconds, frequency = 2,000Hz (2kHz), 0 t0 = 1.0 milliseconds, frequency = 1,000Hz (1kHz), t0 = 2.0 milliseconds, frequency = 500Hz (.5kHz), t = 1 second, frequency = 1Hz, etc. 0 Fig. 10 - A Simple Sine Wave Audio Signal Reconstructingthe Original Waveform After a sufficient number of clock pulses, the samples begin to exit from the last storage register of the BBD. At this point, a waveform is reconstructed. It is very similar to the original input signal, but rather than being a smooth and continuous wave, it moves up and down in steps, corresponding to the average voltages of the samples. The steps can be considered to be a high frequency noise (clock noise) superimposed on the original waveform. Thus, by routing the signal through a low pass filter, the steps are eliminated and the waveform is smoothed out to be very nearly identical to the original input signal, except that it is present at some later time (i.e., it is delayed). (Refer to Figure 11.)