Linux music player recommendation

[supereb]

I would like to set up a laptop as a music player for headphone listening, but have some unique requirements, which none of the players I've found seem to meet.

I need to be able to playback through a couple of plugins - a parametic EQ as well as either an IR loader or a .sofa file plugin (Sparta Binauralizer). I'll be using a .sofa file created by Genelec's Aural ID service.

I would like to avoid resampling - my collection is half 44/16 and half 96/24. I would also like to avoid having to manually change the system sample rate whenever I select an album with a different rate.

I know I could use something like Carla to load plugins, but what would be the easiest way to handle sample rate changes for the whole system?

If there isn't a convenient way, is there a media player that would let me use the Sox resampler?

Maybe the easiest would be to upsample my 44/16 collection?

I'm trying to make this as simple to use as possible. This is for listening sessions in a dark room with a glass of whiskey!

Would love to hear everyone's thoughts and suggestions.

[thetotalchaos]

This is a good place to start: http://aqualung.jeremyevans.net/

If you have an audiophile demands i recommend that one: https://www.musicpd.org/
It is the default player in Audiophile Linux. You can find good recommended configuration, somewhere on their website https://www.ap-linux.com/

But if I, am going to sit at my sofa, and listen to my favorite album, my choice of music player is MPD+Cantata
Optionally i enrich my listening experience with appropriate visualization, using this: https://biniou.net/index.php

Best regards,
Biser, aka Totalchaos

[supereb]

Thank you, I'll check those out.

[merlyn]

I would think that for most users resampling is a desirable feature. Otherwise music files would play back at the wrong pitch and tempo. The idea of a music player includes resampling so it can take any file in any format and play it back so it sounds at the right pitch and tempo.

The only way to avoid resampling would be to set the hardware (the sample clock) to the same frequency as the music file. Even then ... it's mostly pro-audio software that plays back 'raw' files.

You are describing a pretty unique setup.

If you want plugins in the signal path then you may want to use JACK and DeaDBeeF can use JACK. You have to complie a module, but it's straightforward : https://github.com/DeaDBeeF-Player/jack.

You could change the JACK settings to match each file, but I wouldn't think that is 'simple' and it would lead to gaps between tracks. One approach would be to set the hardware to 24/96 and use on-the-fly resampling for 16/44.

[AlexTheBassist]

Why do you “need” all that stuff? This isn't making the sound any better, despite you might think otherwise. Binaural recordings are already binaural, so you don't need any “binauralizers” to get this effect; again, mixed and mastered material is already processed in the way it sounds the best. Instead of trying to make a gold plate out of piece of shit, you better get some hardware that doesn't make your ears bleed. Or, even better, go to a professional recording studio to see how equalizers and “binauralizers” are used in real world.

Also, resampling on the fly isn't bad, but upsampling in before is. You won't get “better quality” if you resample all your library to more bits per sample or higher sample rate. All you will get this way is a freaking huge waste of hard drive space.

[supereb]

I would think that for most users resampling is a desirable feature...

Indeed it is. I have used resamplers that are transparent and some that have audible artifacts. Since switching the system sample rate every time I switch albums is not practical, I'll have to use a reseampler, I just want to make sure that the media player includes high quality resampling. Looks like VLC's Secret Rabbit Code SRC has some very high quality options.

[supereb]

Why do you “need” all that stuff?...

You made some assumptions my level of experience and about what I am doing that are incorrect. The "binauralizer" I am using is not something to "enhance" the sound. It's purpose is to load .sofa files (https://www.sofaconventions.org/mediawi ... Acoustics)), allowing one to position sound sources in the horizontal and vertical planes in headphones. For music lovers and audio engineers, this means two speakers at +30°and -30°horizontally at 0°elevation. Since we are listening through headphones, these are virtual speakers.

You may have heard Genelec. They offer a service called Aural ID https://auralid.genelec.com/, that creates a personalized .sofa file containing HRIR's for each ear at various angles, which gets loaded into the binauralizer plugin. The plugin is where you position your virtual speakers as well as you angular position relative to them. The audio then passes through the correct HRIR on it's way to the DAC/headphones.

I'm well versed in sampling theory. The option of upsampling part of my collection was to avoid an inferior upsampling algorithm in the media player and to avoid having to change the system sampling rate whenever I listened to an album of a different rate. I've heard good SRC and bad SRC. My question was about finding an existing media player that would be a good balance of minimizing loss of quality and as well as convenience.

[AlexTheBassist]

You may have heard Genelec

Sure, I mixed on their monitors at my friend's studio in Finland. Great stuff, but too expensive for my budget.

The "binauralizer" I am using is not something to "enhance" the sound. It's purpose is to load .sofa files <a link was here>

If you were reading anything that you linked here to explain me what you want, you'd certainly find at least one solution that works with Linux native players. Yeah, it's right there.

The option of upsampling part of my collection was to avoid an inferior upsampling algorithm in the media player and to avoid having to change the system sampling rate whenever I listened to an album of a different rate.

You don't need to use any conversion provided by the audio player. Pulseaudio automatically resamples with SoX if you configure it to do so.

[supereb]

Pulseaudio automatically resamples with SoX if you configure it to do so.

I did not know that Pulseaudio could be configured to use Sox.

After starting this thread I found that VLC might be an option that could meet my criteria, otherwise I'm using Jack and piecing things together.

[AlexTheBassist]

otherwise I'm using Jack and piecing things together.

JACK and Pulseaudio can work together. However, if you don't need any pro audio capabilities, using JACK is a huge overkill.

[merlyn]

After thinking about it the simplest option is to convert your 16/44 files to 24/96. Then you'll factor out resampling. Hard drives are cheap these days, and I'm compelled to disagree with Alex on principle

[Linuxmusician01]

How does one even obtain 96 kHz 24 bit music? I have a lot of CD's: they are 44,1 kHz 16 bit.

[wjl]

Since switching the system sample rate every time I switch albums is not practical,

Hi supereb,

I'm wondering why you have to do that - my system automatically switches between everything I feed it. My audio interface is set to 48k, but 44.1 works well without any reconfiguration - and so does everything up to 192k.

Using Rhythmbox for CDs, Audacious and/or Audacity and Ardour for everything else, together with Jack of course (as a musician you simply need it). If I get 44.1kHz material to play on, Ardour converts it to my preferred sample rate of 48k during import.

But I guess your use case is totally different?

Cheers,
Wolfgang

[supereb]

After thinking about it the simplest option is to convert your 16/44 files to 24/96. Then you'll factor out resampling. Hard drives are cheap these days...

I was thinking I would need to go that way, but I didn't realize how prevalent media players with high quality SRC were. VLC has Sox as an option that can be set as the default SRC. I've A-B-X'ed source vs multiple up/down SRC with Sox VHQ and could never tell with certainty which was which, so I'm good with that.

How does one even obtain 96 kHz 24 bit music? I have a lot of CD's: they are 44,1 kHz 16 bit.

I get my high res music from HDtracks.com, but there are a few other services. I get hi-res not so much for the high resolution - we can't hear that high, but because often (not always) either the newer mastering is better and/or because better converters are available now than when the music was first put to CD.

I'm wondering why you have to do that - my system automatically switches between everything I feed it...
...But I guess your use case is totally different?

I'm using Jack to route audio into Ardour which is being used as a plugin host to apply an EQ curve as well as process the audio through a .sofa file for spatial audio on headphones. I would have to start and stop Jack as well as load a different Ardour session every time I played music with a different sample rate.

Listened to quite a bit of music with VLC last night, using the Sox resampler and leaving the system set to 48kHz. As I get time I will try to get the sofalizer plugin installed and do away with Jack/Ardour. Gapless playback seems to be an issue though - how can I listen to prog rock albums with gaps between tracks!?!?

Thanks to everyone who has participated in this thread.

[merlyn]

I've found DeaDBeeF plays back without gaps. Don't let the name put you off It's not written by committed carnivores or militant vegans -- it's a number : 0xdeadbeef.

In my head VLC is for video.