audiohobbit
500+ Head-Fier
If you record a PRIR with a speaker 1 m away, then you will hear the virtual speaker 1 m away. Even if it's recorded in an anechoic room I think.
The speakers in the PRIR have to roughly match the size of your room where you watch your movies and listen to music with the Realiser.
I read the main chapters of the manual, just have to have a closer look at the appendices.
Many things from the A8 are missing. It seems there are no gain and delay settings for the (virtual) speakers at the moment. For gain there are some settings but I'm not sure if they work. For delay they say it doesn't work at the moment. So you have to ensure externally that all the speakers recorded for the PRIR are equally loud at the sweet spot and the sound arrives at the same time (Nearer speakers have to be delayed).
There seems to be no way to "play" with the settings (loudness of the channels, delay) afterwards in the Realiser. When creating a sound room there are also gain settings for each channel but they say nothing about them in the manual at all. There's just this Normalis Loudness switch but this is an automatic and who knows how good this will work.
Totally missing is the diagnostics page after PRIR measurement that gives you some info how good or bad the PRIR has been recorded.
The manual HPEQ procedure(s) and manual tweaking of the Auto HPEQ are better solved than with the A8. But still the first band includes 0-500 Hz, so you can't do any fine tuning in this range. That's annoying. Headphones also vary greatly in this range, e.g. bass response can be totally different between 2 headphones.
And since there are also no more simple bass and treble sliders, as in the A8, there's no way with on-board utilities to influence bass response at around 100 Hz for example. There's something with 40 Hz but I'm not sure if it's an EQ. I hoped for a high pass filter.
With auto EQ with the A8 you could change the compression factors an 3 different bands, at higher bands auto EQ is normally not so heavily applied (only factor 0.3). That's also missing with the A16 and no mentioning what factors are implemented.
What's also not clear to me are the positions of the speakers that will be recorded, e.g. where exactly do they have to be placed? In the appendix there are lists with azimuth and elevation angles but sometimes those angles seem very odd to me. Does this mean that the speakers have to be placed at those angles? You can manually specify angles when setting up PRIR sound rooms (Chapter 5.1) but I don't think that the Atmos decoding algorithm remaps the objects to the speaker layout you entered manually. This would be the functionality of a Trinnov that is x times more expensive.
So I think the Atmos decoder assumes specific angles for the speakers and I clearly want to know them.
Next point: How do I get my A8 PRIR into the A16? And is it even possible with this FW at the moment?
If I use the A16 like my A8 with decoded sound coming from my AVR to the multichannel analog ins of the A16, how does this work?
Do I have to set up a PCM sound room then?
The speakers in the PRIR have to roughly match the size of your room where you watch your movies and listen to music with the Realiser.
I read the main chapters of the manual, just have to have a closer look at the appendices.
Many things from the A8 are missing. It seems there are no gain and delay settings for the (virtual) speakers at the moment. For gain there are some settings but I'm not sure if they work. For delay they say it doesn't work at the moment. So you have to ensure externally that all the speakers recorded for the PRIR are equally loud at the sweet spot and the sound arrives at the same time (Nearer speakers have to be delayed).
There seems to be no way to "play" with the settings (loudness of the channels, delay) afterwards in the Realiser. When creating a sound room there are also gain settings for each channel but they say nothing about them in the manual at all. There's just this Normalis Loudness switch but this is an automatic and who knows how good this will work.
Totally missing is the diagnostics page after PRIR measurement that gives you some info how good or bad the PRIR has been recorded.
The manual HPEQ procedure(s) and manual tweaking of the Auto HPEQ are better solved than with the A8. But still the first band includes 0-500 Hz, so you can't do any fine tuning in this range. That's annoying. Headphones also vary greatly in this range, e.g. bass response can be totally different between 2 headphones.
And since there are also no more simple bass and treble sliders, as in the A8, there's no way with on-board utilities to influence bass response at around 100 Hz for example. There's something with 40 Hz but I'm not sure if it's an EQ. I hoped for a high pass filter.
With auto EQ with the A8 you could change the compression factors an 3 different bands, at higher bands auto EQ is normally not so heavily applied (only factor 0.3). That's also missing with the A16 and no mentioning what factors are implemented.
What's also not clear to me are the positions of the speakers that will be recorded, e.g. where exactly do they have to be placed? In the appendix there are lists with azimuth and elevation angles but sometimes those angles seem very odd to me. Does this mean that the speakers have to be placed at those angles? You can manually specify angles when setting up PRIR sound rooms (Chapter 5.1) but I don't think that the Atmos decoding algorithm remaps the objects to the speaker layout you entered manually. This would be the functionality of a Trinnov that is x times more expensive.
So I think the Atmos decoder assumes specific angles for the speakers and I clearly want to know them.
Next point: How do I get my A8 PRIR into the A16? And is it even possible with this FW at the moment?
If I use the A16 like my A8 with decoded sound coming from my AVR to the multichannel analog ins of the A16, how does this work?
Do I have to set up a PCM sound room then?