Hi,
I don't quite understand your configuration and why you would set ALSA volume to 85%. ???
Are u using an external volume control (on your amp or preamp)?
-Tim
Really thanks
I did above 2 days ago but when boot pi again I cannot connect to moode .
So I think I make mistake .
just now image write again and edit the config.txt ( I see only config file icon) boot and connected
so play music tonight
really thanks again.
Holzohren,
All digital volume controls affect the bits - whether it is an issue is debatable.
You would have to set the level to 50% to lose 1 bit, 85% is not a serious deficit.
Ian
All digital volume controls affect the bits - whether it is an issue is debatable.
You would have to set the level to 50% to lose 1 bit, 85% is not a serious deficit.
Ian
Hi Tim,
I want to adjust the volume for a direct listening comparison between LP and HiFiBerry. I'm not sure if the change of the volume (ALSA) synonymous the quality of the HiRes file changes.
Hi,
Some improvements to Source Config coming in moOde 3.8 ncluding quick Re-mount and Reset library cache.
The Re-mount function is useful in cases where NAS sources are offline when Pi is rebooted. Resetting the Library cache resolves the issue where the Library was opened before MPD update completed thus leaving the Library incomplete.
-Tim
Some improvements to Source Config coming in moOde 3.8 ncluding quick Re-mount and Reset library cache.
The Re-mount function is useful in cases where NAS sources are offline when Pi is rebooted. Resetting the Library cache resolves the issue where the Library was opened before MPD update completed thus leaving the Library incomplete.
-Tim
Attachments
Missing metadata / tracks
Moode sounds amazing and remains the best piece of freeware I've paid for, but the library view appears to be dropping artists / albums that MPAD "sees" so I guess the tagging is Ok, I have >15000 tracks can that be an issue?
I'm quite happy using MPAD except I think the next IOS update will kill it off.
Moode sounds amazing and remains the best piece of freeware I've paid for, but the library view appears to be dropping artists / albums that MPAD "sees" so I guess the tagging is Ok, I have >15000 tracks can that be an issue?
I'm quite happy using MPAD except I think the next IOS update will kill it off.
Attachments
I am considering upgrading from RuneAudio to Moode on my Pi 2B. Could anyone help me with a few questions?
1. Is it compatible with the Behringer LCA202?
2. Could I mix the line level input of the LCA202 into the output from Moode?
I use the Pi as a phone controlled audio player in my boat. I would like to mix in audio from the VHF radio via the line in of the LCA202. I am hoping this is just a matter of making changes to mixer settings.
Thanks
1. Is it compatible with the Behringer LCA202?
2. Could I mix the line level input of the LCA202 into the output from Moode?
I use the Pi as a phone controlled audio player in my boat. I would like to mix in audio from the VHF radio via the line in of the LCA202. I am hoping this is just a matter of making changes to mixer settings.
Thanks
Can you do it with Rune?
If yes, then you should be able to do the same with Moode.
What you can mix with your unit should probably be addressed to the Behringer forums. To Moode, it will present as a 2-channel audio device, any other functions like mixing USB out with analog in would be outside of Moode.
The UCA202 worked fine with Rune and seems to work just as well with Moode.
Is there a Behringer forum here that deals with Alsa mixing questions?
The UCA202 has a monitor function but I believe that is limited to the headphone port. I would like to mix the line level input with the Pi's audio then send it out through the optical output. I experience a ground loop issue whenever I use the line level output from the UCA202.
You misunderstood - can you do now, with Rune, what you are asking? 'Working fine' wasn't your inquiry - you are asking for a specific use case. Can you do that now with Rune? Essentially Rune and Moode use the same subsystems, so if you can do it with Rune, you should be able to do it with Moode.
Dunno. The 'diy' part of the forum name has always served me well when looking for support.
ALSA has nothing to do with it - you are outputting music you have decoded from a digital source, be that radio, NAS, USB, from the Pi to the UCA202 - the subsequent mixing of multiple UCA202 sources to an output, if any, is done outside of the Pi, and ALSA has no way of controlling that.
As I said before, the Behringer unit presents as an output device to Linux. It doesn't offer input channels to mix.
Last edited:
Tim, as always, I have been enjoying Moode greatly! Thanks for your continuous work.
I have a question that may go beyond the scope of Moode itself, but you may be able to point me in the right direction.
My DAC, in theory, is able to play at 16, 24 and 32 bits. As I reported earlier, it actually plays DSD through DoP perfectly! But inside Moode, ALSA only detects 16 and 32 bit options. 24 bits is not available to the OS and therefore MPD reports that it is outputting 32bits every time I play a 24 bit file, even the DSD files through DoP, which is weird as I assumed that DoP was done through 24 bits. Still, the DAC correctly understands the DSD.
So, is there a way to 'add' a 24 bit mode even if not detected, or this is Linux / Kernel issue? Perhaps it's a DAC issue?
Also, what is the inner procedure going on when a bit rate is not supported? Audio info inside Moode reads that the original file is 24 bits and the output is 32, but over sampling says 'none'. So in that scenario, who is doing the over sampling and how is it handling? Is it going through MPDs configuration of SoX or is it relaying on some other algorithms from ALSA directly in such cases?
Thanks for any insight! Best regards,
Rafa.
I have a question that may go beyond the scope of Moode itself, but you may be able to point me in the right direction.
My DAC, in theory, is able to play at 16, 24 and 32 bits. As I reported earlier, it actually plays DSD through DoP perfectly! But inside Moode, ALSA only detects 16 and 32 bit options. 24 bits is not available to the OS and therefore MPD reports that it is outputting 32bits every time I play a 24 bit file, even the DSD files through DoP, which is weird as I assumed that DoP was done through 24 bits. Still, the DAC correctly understands the DSD.
So, is there a way to 'add' a 24 bit mode even if not detected, or this is Linux / Kernel issue? Perhaps it's a DAC issue?
Also, what is the inner procedure going on when a bit rate is not supported? Audio info inside Moode reads that the original file is 24 bits and the output is 32, but over sampling says 'none'. So in that scenario, who is doing the over sampling and how is it handling? Is it going through MPDs configuration of SoX or is it relaying on some other algorithms from ALSA directly in such cases?
Thanks for any insight! Best regards,
Rafa.
Hi Rafa,
I tested a DSD64 file with MPD set to DoP=yes and ALSA output was the expected 24/176.4K Note that the DAC I'm using is an I2S DAC that doesn't support DoP. Also I'm using a moOde 3.8 build that has MPD 0.20.9 compiled with only FFMPEG codecs.
You could try setting MPD logging to verbose and then look at the log to see if it shows in detail how the particular DSF file is processed.
-Tim
I tested a DSD64 file with MPD set to DoP=yes and ALSA output was the expected 24/176.4K Note that the DAC I'm using is an I2S DAC that doesn't support DoP. Also I'm using a moOde 3.8 build that has MPD 0.20.9 compiled with only FFMPEG codecs.
You could try setting MPD logging to verbose and then look at the log to see if it shows in detail how the particular DSF file is processed.
-Tim
Attachments
Thanks! I was afraid my post would be useless, as I explained myself very poorly.
Let me rephrase:
The OS returns from: `cat /proc/asound/card1/stream0` only S32_LE and S16_LE.
But there are no traces of the S24_LE or S24_3LE that would allow outputing native 24bits.
So for EVERY FILE (not only DSDs) that has a bit rate higher than 16, MPD is playing at 32bits (obviously!).
My question was: I'm sure that my DAC actually supports 24 bits. So, is there a way to 'force' the OS to detect and / or output 24 bit audio and not 'upsample' to 32 bits?
If not, who is doing the 'upsampling' when playing a 24 bit file (again, not DSD)? Because in audio info the resampling reads 'no', yet the origin file is 24 bits but the output stream is 32 bits.
Hope I made myself clearer this time around, sorry for the confussion.
Best regards,
Rafa.
Let me rephrase:
The OS returns from: `cat /proc/asound/card1/stream0` only S32_LE and S16_LE.
But there are no traces of the S24_LE or S24_3LE that would allow outputing native 24bits.
So for EVERY FILE (not only DSDs) that has a bit rate higher than 16, MPD is playing at 32bits (obviously!).
My question was: I'm sure that my DAC actually supports 24 bits. So, is there a way to 'force' the OS to detect and / or output 24 bit audio and not 'upsample' to 32 bits?
If not, who is doing the 'upsampling' when playing a 24 bit file (again, not DSD)? Because in audio info the resampling reads 'no', yet the origin file is 24 bits but the output stream is 32 bits.
Hope I made myself clearer this time around, sorry for the confussion.
Best regards,
Rafa.
Attachments
Last edited:
Hi Rafa,
Possibly ALSA thinks your DAC's USB receiver wants only 16 or 32 bit word length so 24 bit is zero-padded to 32-bit. This bit-padding does not change the original quantized value in the 24 bit word and the sample rate remains unchanged so no "resampling" is happening.
I'm not sure which component does the bit-padding but MPD verbose log might show something.
You can try forcing 24 bit word length by selecting "24 bit / * kHz" item in SoX resampling.
-Tim
Possibly ALSA thinks your DAC's USB receiver wants only 16 or 32 bit word length so 24 bit is zero-padded to 32-bit. This bit-padding does not change the original quantized value in the 24 bit word and the sample rate remains unchanged so no "resampling" is happening.
I'm not sure which component does the bit-padding but MPD verbose log might show something.
You can try forcing 24 bit word length by selecting "24 bit / * kHz" item in SoX resampling.
-Tim
Just muckin about... I changed my sox resampling to 24/*, but when I checked the audio status it said it was decoding to 32-bit, something my IQ Audio digi-Amp+ doesn't handle - sounded fine, but I wonder if there's some calculation amiss?
Attachments
Just done the same with my Hifiberry Digi+ with my 24/96 dac at the end of it.
Any 16/44 output displays 16bit format. A 24/(44/48/96/192) formats to 32/source frequency, the same with sox resampling with the output frequency set to sox output.
Nothing sounds amiss.
Sent from my P01M using Tapatalk
Any 16/44 output displays 16bit format. A 24/(44/48/96/192) formats to 32/source frequency, the same with sox resampling with the output frequency set to sox output.
Nothing sounds amiss.
Sent from my P01M using Tapatalk
Yes, it appears the S24_LE and S24_3LE formats are rare within the DACs. I did find some mention of the zero-padding that Tim mentions (although not widely spread).
If that is the case, all is fine! DACs probably understand the zero padded words and everything is OK. I just wanted to make sure that all the effort to do bit-perfect was not being wasted on upsamling within ALSA and bypassing SoX. Otherwise,I would prefer to force everything to 32/*. But if it gets zero-padded, I prefer bit perfect and let the DAC do its thing!
Thanks!
Rafa.
If that is the case, all is fine! DACs probably understand the zero padded words and everything is OK. I just wanted to make sure that all the effort to do bit-perfect was not being wasted on upsamling within ALSA and bypassing SoX. Otherwise,I would prefer to force everything to 32/*. But if it gets zero-padded, I prefer bit perfect and let the DAC do its thing!
Thanks!
Rafa.