Which distribution? Some are more easy than others to graft new parts onto old systems.

From the documentation included with the source of ALSA plugins 1.0.11:

I agree that modifying the defaults might be a good idea. The problem is finding a default that both those with old hardware and those with strict quality requirements will find acceptable. I think SRC_SINC_FASTEST is about right - but there are a lot of factors to consider.

Hi cabbagerat,
Thanks for taking the time to help. It's a Mandriva 2005, which I have promised myself to upgrade some day to Debian Etch or Ubuntu, but right now it has to stay that way.
What do you recommend?

Unfortunately I have no experience with Mandriva whatsoever - I haven't used an RPM distro since RedHat 5, so I don't know what to advise. The short answer is that you need to update to ALSA lib 1.0.11 or later - but this might require more complex upgrades. You might successfully build ALSA-lib and ALSA-plugins from source and "make install" them, but I'm not sure.

It would probably be easier (if this is a home PC and not a server or something) to upgrade to Ubuntu 6.10 or the latest Fedora.

Has anyone gotten this to work on Fedora Core 6? I have alsa version 1.0.14rc (i think, i'm not at home right now) but i do not have the samplerate files in my /usr/lib/alsa-lib. Either Fedora puts them somewhere else, or they don't include them at all. Any ideas?

Scratch that, i got it working in Ubunty 7.04 ... now if i could only find a way to kill this insane popping/crackling on trackchange...

Yes. That's very anoying issue. I hope it gets resolved soon.

Seems to be alsa related too, since it happends no matter what player i'm using. Tried changing buffers and some other settings but to no avail...

yes. this is quite a good thread, I'm glad I found it!

for those that care

then download alsa-plugins-1.0.14rc1a.tar.bz2
go to downloaded directory and do

I'm resurrecting an old thread (again) but I wanted to add something important. Amarok by default does not seem to obey .asoundrc or resamples by itself or something..... Either way it makes the udial sample sound terrible whereas mplayer (using alsa) and aplay sound fine. I have spent hours trying to fix this and I have finally succeeded. I am using arch linux. Here are my steps:

1. Install libsamplerate and alsa-plugins
2. Put "defaults.pcm.rate_converter "samplerate_best"" in .asoundrc (without the first and last "s)
3. Go to the amarok configuration, Engine tab, select alsa as the output plugin.
4. (This is the one that took me hours to figure out) Under 'ALSA Device Configuration' in the 'Mono' and 'Stereo' boxes put "hw:0" (again without the "s). This made the udial sample sound the same as with mplayer and I'm sure my music sounds more crisp

On most ALSA configurations, using the "hw:0" devices bypasses dmix and resampling. I don't know what ArchLinux does, but I think the reason you don't hear the problem is that your card is playing udial at the native 44100Hz. What sound card do you have?

For the record, resampling to 48000 using samplerate_medium and samplerate_best with the newest libsamplerate sound absolutely perfect on udial.

Interesting. I think you might be right about it bypassing dmix. Any ideas what I could change to get it to resample properly and go through my alsa settings? My CPU usage is lower when I use amarok than with mplayer which suggests it isn't using the samplerate_best setting I set. Why on earth do the default settings for alsa in amarok not use my .asoundrc?

I am using the onboard sound (AD1988 Chip) on my Asus P5B. I have no idea if it can play at 44100 but I would have thought that it doesn't.

edit: So...using "aplay -D plug:pcm.48khq *.wav" I get:


I was using this .asoundrc:


However if I use "defaults.pcm.rate_converter "samplerate_best"" in .asoundrc I get higher cpu usage than with no .asoundrc so the rate_converter is working. This is great (when using mplayer or aplay) but when I try to play flash videos in firefox, flash crashes with something like:


Forgetting my flash woes for a moment. If I set amarok alsa output to front (rather than hw:0) it also works with no distortion. Is this also bypassing dmix or something?

While ALSA device is your mplayer using?

It seems that by default the front device has "softvol" enabled but not dmix. Only the "default" device seems to have dmix enabled out of the box. If you use this device for everything, everything should get resampled and mixed with the default rate converter. If you set that to samplerate_whatever it should use that.

You might need to ask these questions on one of the ALSA mailing lists to get a sensible answer.

Thanks. I have no idea what device mplayer is using...it's just using -ao alsa. I shall post on the alsa list.

I had this whole thing figured out a while back when I was trying to get my old Chaintech AV-710 working with no resampling in software. I don't have time this minute, but I'll go back into my notes and settings to see if I can help.

I'm on Arch too btw.

I'm running Ubuntu Hardy 8.04 on a 32-bit Athlon and an A7N8XE mobo. My sound software is ALSA and the sound card ID follows:
00:05.0 Multimedia audio controller: nVidia Corporation nForce Audio Processing Unit (rev a2)
00:06.0 Multimedia audio controller: nVidia Corporation nForce2 AC97 Audio Controler (MCP) (rev a1)
Subsystem: ASUSTeK Computer Inc. nForce2 AC97 Audio Controler (MCP)

Can anyone help me with the following Internet Archive show: Brahms - Tragic Overture in D Minor, Op. 81

The original FLAC file has a sample rate of 48 KHz, which file I will temporarily keep in the show till I resolve the problem: the playback speed is too fast.

As you can see, the solution I have applied is to resample the performance at 44.1 KHz.

First of all, is my diagnosis of too-fast playback correct, or is there something wrong with my hardware and/or software?

Secondly, if resampling is not the solution, what is?