I have my eye on some other codecs for the next project.
There ARE good reasons for recording and processing music at higher bitrates and sample sizes. Effects, especially those with positive feedback loops, can go more unstable and clip and lose information with fewer bits and samples.
There's just no good reason to DISTRIBUTE music at the higher rates.
I wonder if gigabit would make a difference on latency, faster packet transmissions. Could packets drop from 64 to 32b?
4ms is pretty good but I feel like sub 2ms would be nicer.
(emphasis mine)
The latency depends on the device. Hardware implementations of Dante commonly support latencies of 1ms or less, but software implementations are higher. The minimum latency of Dante Virtual Soundcard running on a PC is 4ms.
That's the appropriate number to compare against here (since the PC is using a software driver to interface with the network). However, that 4ms number is one-way latency, and the OP's 3.6ms number is round-trip. So this is already half the latency of DVS. (That being said, it sounds like this latency figure was only achieved in very ideal configurations, and we don't know the reliability/rate of late packets compared to DVS.)
It seems like, at 16000TbaseT anyways, you’re adding an overhead of about 150% on top of copper/fiber latency plus transmit time latency to process data at that bandwidth. I wonder if the same holds true at 100 vs 1000, 2500, 10000? Certainly this is a known tradeoff for DDR performance tuning — if you don’t mind spiking response times greatly, you can get the advertised maximum speeds, else you accept less bandwidth for somewhat less latency — and they’re both effectively using the same strategies to talk over copper.
Um.
"very low latency" in audio is <=1ms. 3.6ms is good but not special.
https://interfacinglinux.com/linux-compatible-audio-interfac...
AFAIK the best one is RME AIO Pro PCIe card. ETH-68 matches the round trip audio latency of this card at 48 kHz and surpasses it be 0.33 ms at 96 kHz.
You might be thinking of latency of the converters, which is normally sub-ms.
> The LATMON pulse width is therefore an accurate measure of the total processing latency of each cycle and is affected by every element in the data path: processing delay in the microcontroller, network transmission delay, host OS delays, signal processing delay in the DAW, etc
One way audio latency is about 3.6 ms divided by two = 1.8 ms. This type of latency is audible.
Processing latency is just the amount of time it takes to complete all processing for each audio cycle. From the histograms, the typical processing latency is about 625 us and of course there is some jitter (almost all of the jitter comes from the Linux host BTW). Processing latency is not audible. However if the processing latency exceeds the deadline on a given audio cycle, there will be an underrun which will cause an audible glitch.
I would love to know why this is hand-wavy and not very cool? Even as not-audiophile, I’ve got ideas of things to use this for.
> RME HDSPe AIO Pro PCIe
> eth68 matches the latency performance of the RME card at 48 kHz and surpasses it by 0.33 ms at 96 kHz.
On Linux, we don’t have much in the way of Thunderbolt support for audio interfaces, so the only way to achieve this sort of latency has traditionally been with PCIe or PCI audio interfaces. Having a low-cost, infinitely more portable solution would be very welcome.
Or is there no such synchronization, in which case there would be long-term drift?