What is the native DAX audio sample size, is it 16 bits/sample?
When DAX is selected, are all other digital audio processing functions bypassed (e.g. compression, etc.) on both receive and transmit?
Finally, how does this compare to the VAC connections on the legacy 3000 and 5000 radios? What is the native, internal VAC sample rate and sample size on those radios?
There are two ways to use DAX -- you can connect directly to the API in which case you will be working with 24ksps VITA-49 data streams. You can also connect to the sound cards in the PC which have been upsampled to 48ksps. In the native API, the data is provided as stereo IEEE-754 single precision floating point data with 0dBFS at 1.0F give or take. So this is 64-bits-per-sample with a 23-bit mantissa.
> What is the native DAX audio sample size, is it 16 bits/sample?
A little bit tricky, but the data starts its life as 16-bit data in an ADC with an ENOB of 12.1. As the data is decimated, you swap sample rate for bit-depth. This confuses a lot of people because they look at a 16-bit data converter and assume that it's dynamic range is 16 * 6 = 96dB. And, in fact this is true at the Nyquist bandwidth (122.88 MHz in our case). But through decimation, we achieve processing gain. To understand how this works, imagine a signal right at the LSB of the 16-bits. Each sample it bounces between 0 and 1 at 122.88Msps in a pattern of, let's say 0001 0001 0001 0001. So every fourth sample it is a 1. When we decimate this data by two, we now have samples that are at a 61.44Msps rate, but now we look at two samples for every one we eject from the decimator. With the first two samples (00), we eject a 00. With the next to samples, we see a 01 and so we eject a 01. Notice that we are outputting 17 bits instead of 16 from the decimator -- we tacked on an extra bit and we average the bits we decimate to decide what to output. If you imagine the decimal point in-between the 0 and 1 bit of the 17-bit number, our samples would look like this:
Now we can decimate one more time and add another bit. Now when we look at the two samples from our first decimation, we see one has a 0 at the end and the other has a 1. Since we are half-way in-between, we set the bottom bits to a 01:
In the binary system, this is equivalent to saying "1/4" of our original LSB, which is the average of the four samples that we got in which were 0, 0, 0 and 1 -- the average is 1/4. In this way, we increase the number of bits of precision and the dynamic range, but it is considerably less data -- now we are sampling at 30.72Msps. We went from 64-bits on the input for four samples to 18-bits on the output.
This is not an entirely accurate model for many reasons, but it should give you an intuitive feel for how we use the input data over time to provide more bits of information at a lower speed in the output. By the time we decimate down to 24ksps from 245760000ksps, we have gained log2(245760000/24000) = 13-bits. So we could have 16+13=29 bits of output data, but we're told that only 12.1 are really good so this would be 12.1+13.3 = 25.4 bits of data. The 3000/5000 have 24-bits for reference.
The other way to access the data is after we have put it into a Microsoft Windows audio device. You will see these enumerated in the system as audio channels with DAX RX and TX names. This data has been upsampled to 48ksps, but is otherwise the same data.
For DAX audio, the DAX data replaces the microphone audio so it goes through the compressor and the modulator. If you do not want to use the compressor, you can disable it. To remove the modulator and just perform upconversion, select DIGU which will essentially do what you want. The DAX audio receive data goes through the receive signal chain including all filters, DSP, etc. Again, you can turn any of this off that you want. DIGU also removes several of the pieces that don't make sense for digital data (NR for example).
> Finally, how does this compare to the VAC connections on the legacy 3000 and 5000 radios? What is the native, internal VAC sample rate and sample size on those radios?
It is essentially the same as what is done for the 3000/5000. The 3000 and 5000 do downconversion in the analog domain via a direct conversion and the resulting data is 24-bit per sample real data which is converted to floating point in PowerSDR at different sample rates (48, 96 or 192ksps). What you get in those radios is a wider bandwidth of data initially, but we then filter the data so if you are at 192ksps, you get a lot of data that has been filtered into a small bandwidth.
The fidelity of the data is similar, but the 6000 has both a better receiver and transmitter and more effective bits of output.
I have sort have asked this before but it was not picked up.
Where is the sampling rate and frequency accuracy performance determined for DAX?
The initial ADC sampling clock is TCXO, OCXO or GPSDO derived.
Does this carry through all the way to 3rd party processes?
Andrew, de VK5CV.
Another question... Does DAX input on USB/LSB go through the TX EQ and Processor now? Some previous versions did not. I.e. Do I need to pre-EQ and compress my recordings for Voice Keyer like I have been doing, or will the 6500 do that on the DAX input?