PulseAudio 2.0 Is Set To Be Released Very Soon

[schmidtbag]

I know because I have an ASUS Xonar D2X, which is capable of 192KHz. If I tried to push more than this through the card, I suspect ALSA would kick up a fuss. I'm no expert either but I gather hardware mixing is the key and contrary to some comments here, this feature is missing from more than just the cheapest sound cards. Creative have spoiled us with cards that have all the bells and whistles while other high-end cards focus on playing just one sound well. My card doesn't have hardware mixing and yet the otherwise crappy ALi integrated card from my old 2004 Sony Vaio does. Note the HWMIX tag present on the ALi page but absent on the ASUS page. If I'd realised the ASUS card didn't have hardware mixing before I bought it, I might have gone for something else but PulseAudio sidesteps this issue as well as providing better sound quality and reliable handling of surround sound than ALSA with dmix does. Trust me, I was a PulseAudio refusenik in the beginning and I tried really hard to make dmix work but no amount of fiddling with asoundrc allowed me to mix surround sound with other sources. I also tried JACK, which was good except that it had no way of automatically upmixing stereo to surround. PulseAudio, on the other hand, worked great out of the box. It's also not the resource hog that everyone makes it out to be. Right now, I see that it is using a whopping 4MB. Wow. As for CPU usage, that is down to the fact that it resamples using a much better quality algorithm than dmix does by default. Ubuntu changed the PulseAudio default resampler for this very reason. dmix can also be configured to use libsamplerate instead, for a fairer comparison.

Well, how much of an audiophile are you? First of all, going beyond 96KHz only works if you're using SPDIF, and many sound cards don't let you go up to 96KHz unless you're using SPDIF. If you aren't, then you're wasting your time trying to achieve something higher. I'm almost positive that analog speakers aren't capable of handling sample rates that high. But whether they can or can't, 96KHz is really the absolute maximum anyone really needs to go, because most audio tracks are not recorded that high and even if they are, most speakers won't let you hear a difference.

Just as an FYI, people can waste a lot of money on speakers/headphones with ridiculous frequency ranges, such as 10Hz-40KHz. The average human can't hear beyond 22KHz, so you could pay more for a speaker with the exact same quality as another, but the only difference is the more expensive speaker has a frequency range that goes ultrasonic.

[droidhacker]

Well, how much of an audiophile are you? First of all, going beyond 96KHz only works if you're using SPDIF, and many sound cards don't let you go up to 96KHz unless you're using SPDIF. If you aren't, then you're wasting your time trying to achieve something higher. I'm almost positive that analog speakers aren't capable of handling sample rates that high. But whether they can or can't, 96KHz is really the absolute maximum anyone really needs to go, because most audio tracks are not recorded that high and even if they are, most speakers won't let you hear a difference.

Just as an FYI, people can waste a lot of money on speakers/headphones with ridiculous frequency ranges, such as 10Hz-40KHz. The average human can't hear beyond 22KHz, so you could pay more for a speaker with the exact same quality as another, but the only difference is the more expensive speaker has a frequency range that goes ultrasonic.

But what if you want to play music your DOG will appreciate??? For this we need speakers up to 60 kHz.
Further, one should note that humans can detect very low frequencies, i.e. 4 Hz. This is through sense of TOUCH rather than hearing, but our brains blend them.

[schmidtbag]

But what if you want to play music your DOG will appreciate??? For this we need speakers up to 60 kHz.
Further, one should note that humans can detect very low frequencies, i.e. 4 Hz. This is through sense of TOUCH rather than hearing, but our brains blend them.

...i hope your whole post is a joke.

dogs can hear that high but it doesn't mean they like the sound. when humans hear a noise of 19khz, its a terrible sound we want to avoid. either way, dogs don't give a crap 20khz-60khz music, or even 30hz-20khz music for that matter.

that whole 4hz finger thing is so irrelevant that i'm not even going to comment about it.

[vertexSymphony]

Now if PA come to fix all this, plus give a nice retro compatibility layout that really works for apps using alsa and such. Then it definitively will replace alsa.

Fundamental mistake : PA won't replace alsa ... won't happen, because alsa provides the drivers and the abstraction of hardware. PA is another abstraction of top of that

{ insert here : WTF!?!? It's a DE, how can it depend on such library...}

Well, that argument is valid for whatever technology you choose, I see the use of PA as something good for desktop, it increases portability and adds fundamental features.
Of course, we have different needs and because of that we appreciate software differently.

Every essential part of a Linux distro should at least support 90% of the modern hardware (cpu,gpu,memory,acpi,soundcard) if it doesn't, then it's not ready for public.

PA supports whatever alsa supports, because PA sits on top of an audio API which provides his own drivers, in this case, ALSA. the problem was PA exposing ugly alsa bugs (similar to what happened with KWin exposing lying drivers or stuff not implemented), but PA was cheap to blame and throw the ball at.

As mentioned earlier in this thread, PA is also very resource hungry

Never saw that happening nor even in Linux, FreeBSD or IlumOS/OpenSolaris
of course, I have to say I never played +32 streams at once because on day-to-day usage never had the need for that. but probably I'll do a stress test out of curiosity

plus give a nice retro compatibility layout that really works for apps using alsa and such

Depends on how the app. uses alsa, but most of them work with the traditional routing of alsa>PA>alsa

Of course that PA needs more work, but as I said, is pretty decent for DESKTOP usage, never had a problem with it ... just waited a little bit until it matured and didn't jump on the hype wagon.
But of course, people will always complain and you can't keep everyone happy, same goes for wayland with choices on compositing and ability to remote, same for kwin dropping opengl 1 (possibly), for gallium 3d usage and such ... but at the end of the day those are pretty good advancements.

Regards

P.S → Sorry if my grammar/orthography blows, I'm not a native english speaker.

[FireBurn]

Wine works fine for me starting with 1.3.31 (1.3.30 introduced "DirectSound reimplemented on top of MMDevAPI." and 1.3.31 some seem-to-be-important fixes). Maybe you should try some recent Wine release?

I'm running Wine 1.4 (which is pretty recent) - it works fine on my laptop but not on my media box where I'm using HDMI

It says there's no audio devices to use

[Luke_Wolf]

I think there are no traces of PulseAudio in my Slackware distro so all programs are using Alsa directly. What critical features I am missing?

Well if you're planning on using Wine or Skype then you'll need to compile 32 bit versions as -compat packages alongside your 64 bit versions. As far as packages required go, everything you need should be laid out at slackbuilds.org, and then follow this for post-installation setup http://www.freedesktop.org/wiki/Soft...r/PerfectSetup

[smorovic]

PA doesn't use hw mixing by design, because many sound cards don't support it anyway and sw support allows full control over details like per-application volume, switching outputs, and now even echo cancellation. It's also useful as abstraction layer which makes sure that applications don't talk directly to hardware and thus break on some configurations. PA takes care of breaking and fixing bugs on those configurations
Not sure how good the cancellation is, but it is possible to achieve good results in software (Skype does that properly).

[allquixotic]

I know because I have an ASUS Xonar D2X, which is capable of 192KHz. If I tried to push more than this through the card, I suspect ALSA would kick up a fuss. I'm no expert either but I gather hardware mixing is the key and contrary to some comments here, this feature is missing from more than just the cheapest sound cards. Creative have spoiled us with cards that have all the bells and whistles while other high-end cards focus on playing just one sound well. My card doesn't have hardware mixing and yet the otherwise crappy ALi integrated card from my old 2004 Sony Vaio does. Note the HWMIX tag present on the ALi page but absent on the ASUS page. If I'd realised the ASUS card didn't have hardware mixing before I bought it, I might have gone for something else but PulseAudio sidesteps this issue as well as providing better sound quality and reliable handling of surround sound than ALSA with dmix does. Trust me, I was a PulseAudio refusenik in the beginning and I tried really hard to make dmix work but no amount of fiddling with asoundrc allowed me to mix surround sound with other sources. I also tried JACK, which was good except that it had no way of automatically upmixing stereo to surround. PulseAudio, on the other hand, worked great out of the box. It's also not the resource hog that everyone makes it out to be. Right now, I see that it is using a whopping 4MB. Wow. As for CPU usage, that is down to the fact that it resamples using a much better quality algorithm than dmix does by default. Ubuntu changed the PulseAudio default resampler for this very reason. dmix can also be configured to use libsamplerate instead, for a fairer comparison.

You could theoretically push "any" sample rate that could be resampled in software. The alsa plug layer or pulseaudio itself would catch the out-of-hardware-bounds sample rate and resample it down to the nearest matching sample rate on the CPU.

For example, gstreamer's audioresample plugin claims to support sample rates from 1 Hz up through 2147483647 Hz (about 2.1 GHz). 2.1 GHz is almost the frequency of an 802.11b signal (2.4 GHz) so that's definitely not perceivable to the human ear (unless you can "hear" WiFi signals..........).

But please be aware that taking Cd-quality 44.1 kHz audio and resampling it "up" to a higher sample rate is NOT going to make the audio sound "better". You can't get more definition than already exists in the audio. Well, you can have a program that tries to synthetically add definition (SRS Labs releases a software package for Windows that tries to do this in real time), but if you are just using ordinary PCM processing like resampling, the quality can't possibly increase by upsampling.

It's the same idea as: if you have a heavily compressed GIF picture and it looks like crap, will saving it as a 32-bit BMP make it look better? No, why? Because the pixels that are being displayed in the rendered gif are going to be the same ones in the BMP, just taking up more space on disk.

The only way to truly use all that extra kHz on your audio card would be to work with source audio that was originally recorded using a capture device at that sample rate. If you have a microphone that can accurately capture details at 192 kHz and you plug it into the capture port of your sound card, then yes, you could hear anything captured by that mic at 192 kHz of fidelity. But the fidelity of CD Audio or Blu-Ray audio or whatever is not going to be higher than the source material, which is usually 44.1 KHz or 48 KHz.

[Chewi]

But please be aware that taking Cd-quality 44.1 kHz audio and resampling it "up" to a higher sample rate is NOT going to make the audio sound "better". You can't get more definition than already exists in the audio.

I never said that was the case. I use 88.2KHz to get better quality from 96KHz DVDs than I would at 44.1 or 48. I chose 88.2KHz because resampling CD audio to exactly double 44.1KHz is less work for the CPU and CD audio is what I play most of the time. I'll write more tomorrow.

[boast]

I haven't had any issues with PA for the longest time. But then again I also don't work at a sound studio or do anything fancy. But the linux community always has issues with the "well, it works for me so you're just a special case" in the graphics realm...

It's the same idea as: if you have a heavily compressed GIF picture and it looks like crap, will saving it as a 32-bit BMP make it look better? No, why? Because the pixels that are being displayed in the rendered gif are going to be the same ones in the BMP, just taking up more space on disk.

I'm not sure thats a good analogy for audio. Because with good digital signal processing techniques, you, for example, add an extra bit to the sampled data, and with a digital filter calculate/estimate more precisely the original source before it hits the digital-to-analog converter.

For an image, you can add an extra bit for each pixel, but you wouldn't be able to tell what the original source color was, so its useless. But for audio, if your data shows it goes from 0.5v to 0.7v, you can assume that it goes to 0.6 before reaching 0.7, and can create a smoother transition in your analog conversion, thereby making it sound better.

(note, signal processing is not my forte)

Oh, and I'm not sure why some people are confusing the sampling rate with an analog output frequency one would hear...