Linux Audio the way to go!?

[1audio]

For hardware for audio playback I think there are several roads ahead, all valid.

First and my current goal, is a simple package that will play the original audio stream untouched, no level adjustments, no sample rate conversions noting but the raw data stream at its original sample rate. For the most sophisticated DACs (like the Berkeley Audio or the Pacific Microsonics) anything upstream may and probably will reduce the performance of the DAC. Getting a low jitter stream at the native sample rate is paramount. And with HRx and other high resolution audio formats becoming available a means for playing them is in demand.

Second would be a system that supports eq/crossovers and any other similar types of audio processing. This gets harder fast and needs more horsepower. I would like a system that would not change sample rates but rather switch co-efficients to match the sample rate of the source.

There are a raft of other issues around user interface and distributed audio and network/internet sources as well. And no single solution will handle all of this well.

[ewildgoose]

Originally posted by 1audio
For hardware for audio playback I think there are several roads ahead, all valid.

First and my current goal, is a simple package that will play the original audio stream untouched, no level adjustments, no sample rate conversions noting but the raw data stream at its original sample rate. For the most sophisticated DACs (like the Berkeley Audio or the Pacific Microsonics) anything upstream may and probably will reduce the performance of the DAC. Getting a low jitter stream at the native sample rate is paramount.

Since you stuck your head up, can I ask you to try a test.

Please try foobar with the ABX plugin. Use Sox (or some other resampler which scores well in the link above) to resample your most revealing audio track, then use headphones for best transparency (sure use your hifi system if you prefer).

Foobar should be able to give you sample perfect output at a variety of output rates

I would be very interested if you can detect a difference...

If you can't hear a difference of course then you potentially free up a bunch of parameters making it simpler to pursue your goal... Let us know - honest it's quite an interesting experience.

Try it with some other things like converting to MP3 at various bitrates - it's amazingly how many "night and day differences" melt away when you blind test them. Everyone should try it...

[1audio]

I'll be happy to explore per your suggestions but it will take a few days. However a little clarity here would help. Are you suggesting using SOX to upsample from 44.1 or downsample from a higher sample rate? I would first explore downsampling from 176.4 material to see if it degrades audibly (it always has in the past). I would use the SRC offline for this. I don't have any MP3 material myself but I suppose I could encode and decode for what its worth.

I use MPD and it is configured to automatically switch sample rates through to the output as I described with full 24 bit resolution throughout. Only in the current git for now.

I'll check out the ABX feature of foobar but I have always found the abx paradigm to make the whole process unpleasant and harder to tell differences. Perhaps its better as a software plugin than some clunky hardware.

[ewildgoose]

Originally posted by 1audio
I'll be happy to explore per your suggestions but it will take a few days. However a little clarity here would help. Are you suggesting using SOX to upsample from 44.1 or downsample from a higher sample rate? I would first explore downsampling from 176.4 material to see if it degrades audibly (it always has in the past). I would use the SRC offline for this. I don't have any MP3 material myself but I suppose I could encode and decode for what its worth.

Well, to be honest, I would actually not expect you to be able to tell the difference between stuff sampled up or down, as long as it's done well.

I had in mind that you probably had 44Khz audio and you could set it to some multiple upwards, however, either way would be instructive (where did you find some 176Khz material?)

As for the MP3, obviously you need to generate it yourself or else you won't have the high quality original to compare with!

Use a decent encoder and see at what point you can reliably tell the difference. It's quite likely to be quite a low bitrate actually unless you pick some type of music that the encoder doesn't handle well. I found this quite an interesting test actually...

Originally posted by 1audio
I'll check out the ABX feature of foobar but I have always found the abx paradigm to make the whole process unpleasant and harder to tell differences. Perhaps its better as a software plugin than some clunky hardware. [/B]

Well, you struck the nail on the head really. Obviously you don't want the process to be a pain, hence try ABX and foobar - I found it fairly simple and convenient. However, you are right, once you try to prove the difference between two samples I personally find that many of those "obvious differences" can shrink away and be quite hard to prove.

Obviously if you find that your resampler is inaudible in practice then you potentially opened a bunch of avenues for yourself

[UnixMan]

Originally posted by LinuksGuru
You can find how to convert APE files on Linux on my web page dedicated to Linux tips and tricks. In short - use K3B.

well, thanks. But IMHO it's just so easy to simply use "mac" (the monkeys-audio codec) to convert .ape back to wav or "shntool" to convert ape to just about anything else directly from the cmd line...

Otherwise, if you can't live without a GUI there are plenty of other applications which are more specific to sound file conversion than k3b. For KDE a very versatile one is SoundKonverter.

It supports:

Decoding: ogg, mp3, mp2, m4a/mp4, aac, 3gp, mpc/mp+, flac, ape, wma, asf/asx, ra, rv, rm, avi, mpeg, wmv, qt/mov, flv, ac3, au/snd, shn, tta, bonk, ofr, ofs, wv, la, pac, spx, mid, mod/s3m/stm/ult/uni/xm/m15/mtm/669/it, wav

Encoding: ogg, mp3, mp2, m4a, aac, mpc, flac, ape, ra, ac3, au, shn, tta, bonk, ofr, ofs, wv, la, pac, spx, wav

Is it enough?

P.S.: sorry for the late answer...

[UnixMan]

Originally posted by ewildgoose
Use a decent encoder and see at what point you can reliably tell the difference. It's quite likely to be quite a low bitrate actually unless you pick some type of music that the encoder doesn't handle well. I found this quite an interesting test actually...

done that. ( Though not using foobar... I don't do windows, ever! ).

I can tell mp3 apart from lossless even for LAME "extreme" VBR and 320Kbps CBR settings.

For lower quality mp3s, on my system I can promptly detect 'em even without comparing.

If you carelessly listen to a (high quality) mp3 alone it may seems good. But - given you have a good enough audio system (that's NOT just some random computer sound card with some random hearphones), if you can compare it with the lossless original the difference is obvious.

The lossy codecs simply "wash out" the music. At a first glance sometimes the good mp3s may even seem to sound "better" then the original: they usually sounds somewhat "cleaner". Then you realize that they seems cleaner because they lack sooo many little details, ambience, etc.

Problem is, so many audio system (likely all of the "consumer" types, but also way too many supposedly "hi-end" ones) sounds like mp3 themselves.

I guess that's why so many people can't tell the difference between mp3 and "real" music.

Actually, sometimes I use the "mp3 test" to check the quality of an audio system: if I can't clearly hear the difference between a "good mp3" and the real thing (lossless file), than the system isn't any good... no matter how expensive it may be. Using mp3s of different quality, you may even sort of "quantify" the quality of the reproduction system!

[soundcheck]

Hi folks.

Good news for folks running the .wav format and MPD/Minion.

I just tested the newest version of Minion (Firefox-MPD add-on) : 1.99.23

Chris - the guy in charge for Minion introduced .wav tagging from filenames (tag-guessing). I am not aware of any other application under Linux doing this.
( You tell me )

On my file-name-format: "Number-Artist-Album-TrackTitle" it works quite well.

For the time being you need to configure a special key in Firefox to enable it.

1. Type "about:config" in the address field .
2. Right click in the main window-field- select: "new""boolean"
3. add "extensions.mpm.guess_tags"
4. set to true
5. Restart FF

That's about it.

I am still discussing with him to get rid of underscores in tags ( Standard feature supplied by EAC). I got all blanks with underscores replaced (for better Unix handling) on my filenames.

Cheers

[soundcheck]

@ewildgoose

Did you try MPD+Minion+brutefir as I describe it in the Wiki?

http://picasaweb.google.com/lh/photo...eat=directlink

[ewildgoose]

Nope, but it looks nice

I use Mythtv on my main media machine and fairly happy with it

I do need a good solution for listening to all this music from a laptop somewhere else mind, however, for the time being just use various media players on each platform.

Ed

[soundcheck]

Originally posted by ewildgoose
Nope, but it looks nice

I use Mythtv on my main media machine and fairly happy with it

I do need a good solution for listening to all this music from a laptop somewhere else mind, however, for the time being just use various media players on each platform.

Ed

The great thing with MPD and "pipe-out":

Every system can be differently configured, with a great set of options. You can control any audio-client from any place in the network. It can be controlled from any place in the network, even your mobile phones. Pretty much like the Sonos system but much more flexible, because you can use any HW.

If you ask me. I don't need a different application for everyday-use.