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I have now played quite a bit with my installation and I‘m very happy with Moode on the RPi4.
After having moved my loudspeakers quite a lot in the room and measuring the sound with and without the subwoofer (with REW) I have found a final place for them. I let REW calculate filters for CamillaDSP as a first step, and then „interactively“ worked towards a better overall response (including the room peculiarities).
Regarding this I have two questions:
- What is the maximum number of filters that CamillaDSP can process on an RPi4? I currently have 10 filters for each of the two channels, and this produces a load of 2% CPU and 0.1% MEM for CamillaDSP, which is negligible in my opinion. Are there other restrictions that I do not see, that limit it?
- If I switch to the CamillaDSP view it shows on the devices tab a selection input named „samplerate“ with a tool tip „sample rate for processing and output“. This is set to 44100 (I didn’t change anything here). If I use source files with other rates, then the output rate as shown by the DAC is the same as the input material. Does this mean that the samplerate chosen here is ignored (this would be my preferred variant)? Or does it mean that CamillaDSP resamples the source to 44100, then does its transformation, and the result is again resampled to the same rate the original material had?
I‘m also thinking about having an additional DAC for the subwoofer. CamillaDSP could mix into 2 additional channels that could be processed differently from the main speakers then. But could this be done with Moode or with this have to be done behind the back of Moode?
And again, thanks a lot for Moode, it has made life a lot more pleasant.
Cheers, Joachim
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1. - What is the maximum number of filters that CamillaDSP can process on an RPi4?
Best to ask this question in the main CamillaDSP thread at diyAudio
https://www.diyaudio.com/community/threa...tc.349818/
2. Does CamillaDSP resample the source to the rate displayed in the Pipeline editor.
Not in the moOde implementation which uses the alsa-cdsp format switcher plugin for CamillaDSP. This plugin automatically updates the working CamillaDSP config with the source format (rate, bits, channels).
3. I‘m also thinking about having an additional DAC for the subwoofer.
a) What audio device is connected to moOde?
b) What make/model subwoofer?
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Ok, your answer to 2. makes a lot of sense and makes me happy because no unnecessary rate adaptions are happening.
My current audio setup is as follows:
- An RPi4 with Moode connected to a Topping E70
- The E70 is connected to a Concentra II amplifier that drives two Magneplaner MG1.6
- In addition the Pre-Amp out of the E70 is connected to an SVS SB 1000 Pro
The SB1000Pro contains an in-built DSP that I currently use to adapt the subwoofer to the MG1.6.
In addition I use CamillaDSP to adapt this joint loudspeaker system (The MG1.6 and the SB 1000 Pro) to the room.
This already works pretty well (some of the room‘s problems cannot be corrected).
But it might be even better to have the sub instead connected to a second DAC (e.g. a Topping E30) that in turn is also connected to the RPi4 and to do the correction I currently do in the subwoofer in CamillaDSP instead. Why? The sub receives the audio analog, digitizes it, does its transformation and converts back to analog.
I‘m not sure that this approach would yield an overall better sound, but it would be nice to know whether it could work in principle.
Cheers, Joachim
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I'd stick with the super nice built-in DSP in the Sub and it's companion Control app :-)
Another reason to stay with Sub DSP is that moOde only supports a single USB audio output. You would not be able to have 2 USB DAC's connected to the Pi-4 each simultaneously receiving the same audio signal.
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(10-08-2023, 09:48 PM)Tim Curtis Wrote: I'd stick with the super nice built-in DSP in the Sub and it's companion Control app :-)
Another reason to stay with Sub DSP is that moOde only supports a single USB audio output. You would not be able to have 2 USB DAC's connected to the Pi-4 each simultaneously receiving the same audio signal.
Isn't the DSP in the SUB introducing latency with its ADC => DSP => DAC ?
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(10-08-2023, 09:56 PM)Nutul Wrote: (10-08-2023, 09:48 PM)Tim Curtis Wrote: I'd stick with the super nice built-in DSP in the Sub and it's companion Control app :-)
Another reason to stay with Sub DSP is that moOde only supports a single USB audio output. You would not be able to have 2 USB DAC's connected to the Pi-4 each simultaneously receiving the same audio signal.
Isn't the DSP in the SUB introducing latency with its ADC => DSP => DAC ?
Why do you think there would be a latency issue?
Signal propagation and processing latencies exist throughout digital and analog playback pipelines but are almost always way below any threshold where they would produce audible artifacts.
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(10-08-2023, 11:57 PM)Tim Curtis Wrote: Why do you think there would be a latency issue?
Signal propagation and processing latencies exist throughout digital and analog playback pipelines but are almost always way below any threshold where they would produce audible artifacts.
TTYTT I don't know how big the delay would be, but what I focus on is the different length (in the time domain) of the 2 signal paths, the SUB one baing "longer"... then, whether such delay could be audible or not, I cannot say.
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(10-08-2023, 09:48 PM)Tim Curtis Wrote: I'd stick with the super nice built-in DSP in the Sub and it's companion Control app :-)
Another reason to stay with Sub DSP is that moOde only supports a single USB audio output. You would not be able to have 2 USB DAC's connected to the Pi-4 each simultaneously receiving the same audio signal.
Do you plan to extend Moode to support multiple audio outputs at some time? I don‘t really need it, and as you said, the companion app is very nice. But it would still be a very interesting addition and open up many new application cases.
Cheers, Joachim
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(10-09-2023, 12:09 AM)Nutul Wrote: (10-08-2023, 11:57 PM)Tim Curtis Wrote: Why do you think there would be a latency issue?
Signal propagation and processing latencies exist throughout digital and analog playback pipelines but are almost always way below any threshold where they would produce audible artifacts.
TTYTT I don't know how big the delay would be, but what I focus on is the different length (in the time domain) of the 2 signal paths, the SUB one baing "longer"... then, whether such delay could be audible or not, I cannot say.
First of all, I cannot hear any delay, and the subwoofer is very „dry“ i.e., the sound from it is cut off very sharply (it is the sealed variant, which is faster than the ported version). This was important to me when I selected it, so it plays along with the loudspeakers nicely (which are very fast).
I think the delay might be audible earliest if it is more than maybe 25ms, introducing some booming, because this would add around one wave length of the signal (at 40Hz). I remember that if we look at video/audio sync, then everything lower than 50ms is no problem.
SVS advertises to connect the subwoofer to the loudspeaker signal directly (which I haven‘t done), before it is sent to the loudspeakers. If we take this as a worst-case scenario, then we have the delay that is introduced through the loudspeaker cable vs. the delay introduced by the DSP and the class-D amplifier in the subwoofer.
As far as I understand it, class-D amplifiers are pretty fast and don‘t introduce much delay, and without calculating it, I would say the signal propagation through the loudspeaker cable and through the amp might be in the same ballpark. So the interesting remaining factor is the delay introduced by the DSP.
The DSP has 3 filters, a low pass, phase correction and room gain compensation. This means 6 transformations in 25ms, an initial FFT and a final one, leaving 3ms for each operation. With a modern processor, this is absolutely no problem, and I assume they chose the CPU based on this. An M4-Cortex for instance could do an FFT in under one ms. And if they are using a specially designed DSP chip then the delay is even lower.
TL;DR: I don‘t think the signal-processing delay introduces any audible latency.
Cheers, Joachim
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(10-09-2023, 06:32 AM)jbaumann Wrote: I think the delay might be audible earliest if it is more than maybe 25ms, introducing some booming, because this would add around one wave length of the signal (at 40Hz). I remember that if we look at video/audio sync, then everything lower than 50ms is no problem.
Yes, delay/latency begin to become audible around 25/30ms, but the most important thing is that if you don't perceive it, everything is fine.
I keep forgetting that nowadays audio processing, in terms of speed, is not an issue, especially in dedicated scenarios... ;-)
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