![]() ![]() Interconnecting these devices digitally can remove the latency and potential signal loss of analog interconnects, but digital transmission introduces a new set of problems, such as timing. (See Chapter 8 for more on setting gain structure.) If you practice good gain structure (essentially, controlling amplitude changes from one device to the next) and keep all your sampling rates consistent, the loss of quality is minimal, but it is still something to consider when using analog interconnects. You may also pay for this in a loss of quality in your audio signal as a result of multiple quantization errors and a loss of frequency range if each digital device is using a different sampling rate. As described in Section 5.2.1, you pay for this strategy with increased latency in your audio system. Transmitting in analog involves performing analog-to-digital conversions coming into the devices and digital-to-analog conversions coming out of the devices. If you dig around you can find several papers (the AES is probably a good source) which run through the maths to correlate jitter with distortion of the signal.When transmitting audio signals between digital audio hardware devices, you need to decide whether to transmit in digital or analog format. So we need a word clock with both low jitter and fast edges. The result is yet another source of jitter. As different logic gates it is fed to will have different logic switching levels you can't guarantee which gates will detect the change in logic state when. Now take a master clock with slow rise and fall times. Jitter of the master sampling clock will soon have drastic effects on the output. In an audio system there can be hundreds of sources each of which might have hundreds of mathematical operations performed on them. Now randomise your sampling points and do the same sums. The result should be perfect cancellation. Now phase shift those results by 180 degrees and add it in to the original. Draw a sine wave and sample it in a number of places equally spaced. ![]() Jitter means that the samples we are dealing with are no longer related to the other samples. Delay lines to equalise for delays in one part of the software are common. The digital chain often has many many steps which act in series on a number of samples from different sources. When dealing with digital audio we are dealing with a sampled analogue original signal. Whilst people argue about how much jitter causes audible effects, it's effects are real. it would take a LOT of jitter to make ANY difference in the perceivable quality of recorded audio. ![]() These will do 10V comfortably, and are pushing faster as SMPS move into MHz ranges. (75 ohm terminator + 75 ohm cable in series)Īddit: Worth testing would be modern MOSFET gate drivers. Or, you can go on eBay and chase down a Rubidium Oscillator.Ī series of 5V 75 Ohm signals, would need multiple 75 ohm line drivers - this needs to drive 10V into 150 ohms. They also have a series like ASVTX-09, which adds Voltage Trim to the TCXO, so you can lock this to a global standard like GPS. The SI5328C & SI5351A both need precision reference sources, so a (VC)TCXO is a good start.ĭigikey have FOX924B, which are TCXO, affordable and easy to apply. How would could the device have multiple outputs to feed several devices the same square wave?ĭigikey show SI5328C-C-GM (36QFN) in stock for for 1 : $12.83, or you can buy a Eval Board for $250 ![]() What would be a good, better, best circiut for creating a perfect square wave?ĥ. How would it be implemented with a Propeller?Ĥ. How much would it be for something like the Si5328 ?ģ. ![]()
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