Phase shifting DC+AC audio signal without affecting its DC content

Question

I have a strongly and asymmetrically clipped audio signal (green signal on attached picture), with DC bias content (blue signal).

In order to obtain the exact DC bias level of this composite signal later, first I would like to phase shift the signal with 180 degrees, without affecting its DC content (red signal on the following picture).

The inverting op-amp configuration change the polarity of the DC content too, it can not be used in this case. Is there any possible solution in the analog signal domain? For the time being I don't want to switch to digital domain.

Answer 1

Your method of extracting the bias voltage will only work if you know the nature of the original AC signal and can infer the zero crossing point from its shape. It will not work for an arbitrary waveform which has been distorted by your circuit.

For example, imagine the output is a square wave going from 0 to 1V. How much (if any) of the bottom half of the original wave has been cut off? You don't know! What shape was the cut off part? You don't know! Since that information has been lost there is no way to reconstruct the original AC waveform, and no way to determine its average value (DC bias point).

Answer 2

to obtain the exact DC bias level of this composite signal

Use a low pass filter - it will average the signal and give you the DC level. Of course, you will need to take into account that a low pass filter will take time to settle to the average value and that there will still be some rippley artefacts left from the AC content of your signal but by choosing the correct number of stages you will get a small ripple that will likely be acceptable to use.

Following a discussion it appears that the op wants the dc level representative of the on-off periods being equal so: -

The comparator turns the analogue signal into a square wave. If the duty cycle isn't 50:50, the integrator raises or lowers the comparator threshold until it is. Bingo, there's the output.

Answer 3

What you need is an all pass filter. It will delay the signal without changing the frequency content. They are described here; All pass filters

No, the all pass filter will not average the signal. That is what all pass means. Here is a simple second order all pass;

Here is the output at 100Hz;

The loss in gain can be made up in a following gain stage.

Here is the output at 1kHz;

This is starting to show errors at 3kHz;

So, it can be done. Now you need to generate the specitication for the all pass you need and design that.