Hi there .I am using Moode audio for many years but I never tried to tweak it .Recently I read about SoxResampling and I decided to give it a go .What are the recomended bitrate and sample rate for my device ? I use an usb dac (musical fidelity v90) that supports 24/96 from usb port (which I am using),32/192 from coaxial feeded from a rpi3b+ .I set Sox to 24/96 but I noticed that setting it 32/768 the Dac still output music so I get confused and need your help to set up this right ...Also another question which I beleive I should made it first ...Are there any benefits from using resampling?
Generally its best to avoid resampling and just go with the native rate of the source file but there are many exceptions to this depending on the audio device and the format of the music files in the collection.
Menu, Audio info will show the actual output format being sent to the audio device.
In the ALSA section of Audio Config read the help for "Output mode". The mode "Default" will allow ALSA to perform format conversions if needed to make sure the actual output format can be accepted by the audio device. This assumes that the audio device correctly reports its capabilities to ALSA.
Turning off all the resampling options and revert to the settings I use to set up moode, at the audio info page shows that the bit rate output is different from the source file regarding bit depth (file played is 16/44.1 and the output shows 24/44.1) so an upsampling occurs is this normal ?
Probably you device only supports 24 bit playback, which require adjusting if from 16b to 24b. Pretty normal. This isn't considered to be upsampling, but a bit width adjustment without any harm.
03-24-2024, 10:36 AM (This post was last modified: 03-24-2024, 10:37 AM by Tim Curtis.
Edit Reason: spelling
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Your device reports that it only supports S24_3LE format (24 bit)
If ALSA output mode is set to "Default (plughw)", ALSA will zero-pad the bit depth to the width required by the audio device. This process preserves the original bits and thus maintains "bit-perfectness" of the samples.
In your case though you have ALSA output mode set to "Direct (hw)" which means that ALSA does not perform any format conversions whatsoever and thus the bit depth padding is happening upstream by MPD.
As I understand it MPD and ALSA actually negotiate the output format according to what the audio device requires and so it may be that if MPD sees that ALSA hw is specified (no format conversion) then MPD does the bit depth padding. The other possibility is that the bit depth padding is due to using MPD volume control (Software or Fixed) which IIRC is a 24 bit volume algorithm.
If your audio device supports Hardware volume i.e., has on-chip volume controller then try setting Volume type to "Hardware". This bypasses MPD volume routine. MPD volume is also bypassed when using CamillaDSP volume.
