Clementine 1.3: no more alsa output card options?

Hello,

with the new Clementine v1.3 , I sadly discovered that we are not able to configure ALSA output card parameters anymore.
I fortunately own an audiophile DAC and I used to output streams to "hw:2,0" to avoid resampling.
Now this option is gone, while two new options are added ("Sample Rate" and "Minimal Buffer fill").
To me this is a regression. Before it worked well, now high resolution files are downsampled in any case, sounding worse than 44.1KHz ones.
This is a pity, since Clementine is by far the best Linux audio player.

Kind Regards
Alex

[hatstand]

I don't think we care about "audiophile" things.

[Ferroin]

I don't think he's going to be back as a user...

I'd give you 10 to 1 odds that what he means by 'audiophile DAC' is 'high-end sound card that can natively do 24-bit at 196KHz audio'

I assume the argument for removing this functionality was so that people didn't get confused by it, which is funny because the current functionality is almost as confusing. In the 'Output device' selection on my laptop, i get two options for Pulseaudio sinks (I need to just get rid of the simultaneous output device...), and 5 that appear to be intended as GStreamer internal device aliases (one each for ALSA, OSS, JACK, Pulse, and generic IPC).

[hatstand]

I'd be happy if we just removed all of it tbh and just used pulseaudio with no options.

[Ferroin]

We'd lose a lot more users if we went that way. To be entirely honest, the only reason I use Pulse audio at all is because there's software that I actually need on occasion for audio work that only supports Pulse (and because Chrome appears to refuse to work with anything else), if it weren't for that, I'd have ripped it out of my system long ago (or more likely never installed it) and switched to using JACK.

Clementine is a music player, which means that to a certain extent, we have to care about people who care about audio quality, and PulseAudio just falls flat on it's face in some cases there (and some of the design choices that they're discussing upstream are positively terrifying, I've heard rumors that they're thinking about routing sound over DBus, which is going to be horrible for latency, even if kdbus ever makes into mainline Linux).

[Ferroin]

I should also qualify my previous comment with the statement that I don't use 196k audio, I just stick to 48k because I can't discern any difference beyond that (and this is coming from someone who can often discern some codec parameters for lossy audio compression just by listening to the audio itself), but I care enough that I avoid Pulse for multimedia work if at all possible, going either straight to hardware if possible, and through JACK if not.

[hatstand]

http://www.aes.org/e-lib/browse.cfm?elib=5539

Examination of published blind listening tests reveal that listeners are strongly disposed to report differences in sound quality when given two alternatives that are identical. The tendency was found in every published test over the past 15 years where analysis was possible and ignorance of the effect caused some tests to be improperly designed or analyzed. The author conducted an extensive single blind listening response bias and it's relationship to order of presentation, listener expectations and the presence of loudness differentials. The findings lead directly to suggestions for consumer purchase behavior and scientific listening test design and analysis.

Not inclined to care :-)

[Ferroin]

And that is exactly why you use real double blind tests. Assuming that people really can't tell the difference, they'll guess wrong on average 50% of the time.

Now, with respect to pulse, I'm not talking about stuff like dithering and quantization issues as much as latency. If your audio pipeline stalls whenever you're doing computationally intensive DSP work on it while you're trying to do real-time work, it's a quality issue in the audio pipeline. I see this type of thing exponentially more often with Pulseaudio than with JACK or direct through ALSA. Hell, I even sometimes get things like this happening when I'm just listening to music with Clementine over Pulseaudio and am building some big software package at the same time.

[hatstand]

Buffer under-runs are an actual reason to change things.

it's not very cool to buy high resolution tracks and have them downsampled by whatever resampler (for ex. crappy default pulseaudio speex-floating-1), lowering quality, adding latency, overloading cpu, while having a 300€ DAC supporting files up to 192KHz.
Before it worked like a charm while there will be more and more people buying "HD" tracks.

(surely this problem should be resolved by pulseaudio upstream too, agreed)

[hatstand]

http://www.aes.org/e-lib/browse.cfm?elib=13185

To study the difference of hearing impression among several high sampling digital recording formats, we conducted subjective evaluation tests of perceptual discrimination among following digital recording formats: 48kHz 24bit PCM, 192kHz 24bit PCM and DSD. Sound stimuli for the evaluation were originally recorded to maintain the exact same quality of analog input feeds to the three A/D conversion systems. The sound reproduction system for the subjective evaluation tests was also carefully designed to reproduce the original quality of each digital recording format. Listening panels were selected from students and faculty stuffs of university of music, recording engineers, and musicians. A pair test method was applied to the subjective evaluation. Through the several evaluation tests, no significant difference was observed.

-> among following digital recording formats: 48kHz/24bit PCM, 192kHz 24bit PCM and DSD <--

these are HQ streams. Nothing to do with 16bit/44.1KHz against 24bit/88.2KHz where qualitative difference is easely audible. Sorry.

[Ferroin]

24 versus 16 bit audio doesn't make as much difference as you think either...
And easily audible for you doesn't mean it is for everyone either. For example, if you give me the same audio encoded in a 128kbps MP3 file and a FLAC file, same sample rate, same bit-depth, I can roughly 80% of the time identify which one is the MP3 just by listening to them both, yet I know numerous people who claim they can tell the difference between 44.1k and 192k sample rates who can't do the same better than about 50% of the time (which pretty much means they're guessing).

It's like the difference between using 24-bit color and 48-bit color in image work, there's minute differences, but better than 95% of humans can't tell the difference.

Even people who work for Xiph.org have written articles on this:
http://xiph.org/~xiphmont/demo/neil-young.html

Now, on the other hand, avoiding the dithering and quantization issues is a practical reason to want to avoid Pulse (especially because a lot of high-end audio cards do up-sample to their native sample rate and format before doing any internal processing).

24bit is a great improvement for music with great dynamics and, above all, it highers the sound/noise ratio by almost 50dB.

I agree that a sample rate beyond 88.2 is pure commercial stuff, and that a listener need a high freq tweeter (as I do) and a newborn earing sensibility to enjoy it.
That said, higher resolution helps reduce aliasing and other kind of distortion, that means, purer audio playback.

It is true that the listener should have a trained ear. Today's radio/streaming compressed dynamics and lossy formats music don't help, as crappy pc speaker and similar paraphernalia.
You just have to get used, trained to enjoy the improvements.

But the loss of hw: alsa output means high res files that sounds worse than CD quality ones and higher resources used.