pulsar08
New Head-Fier
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Almost every digital source these days uses a dac chip which will employ hardware resampling to convert a 44.1khz signal to native 48/96/192khz. This interpolation cause signal degradation to some degree. Foobar2000 has a built in resampler dsp (PPHS) which is optimized for low cpu usage. There are also a few others which supposedly run on older versions of foobar (SOX, SSRC,and an old version or SRC). Of these 3 I only ever managed to get SOX to work. The SRC (Secret Rabbit Code) resampler aka libsamplerate is another option widely considered to have the best quality. If you have basic linux skills enough to install ubuntu from a live cd and quickly switch a couple of values in a config file you can get this working. Here goes:
Edit the pulseaudio config by typing the following into a terminal window: Quote:
Enter your root password and edit these lines in the config file: Quote:
Quote:
s16le = 16bit, s24le = 24bit Quote:
You will want to pick whatever value your card outputs natively. Entering higher values than native will cause hardware resampling to continue (I put in 384000 just to see if it would work and sound still played although an error message did show in the terminal when I went to switch it back to 96000) Click save and reboot your computer.
I believe most of the music players in linux use the pulseaudio sound daemon. Rhythmbox, songbird, audacious, xmms2 are a few good ones.
I have used this on an old pentium 4 and it still doesn't consume much cpu. Rhythmbox stayed about 5-6% and pulseaudio 1-3%. Under heavy multitasking it did skip a few times. I haven't tried it on my overclocked i7 to check for skipping yet cuz the bootloader got corrupted and I need to restore it.
I might be crazy but this change really sounds good, very clear/clean/analog sounding with loads of detail. In the least you have peace of mind knowing you are degrading the signal as little as possible. I'm pretty sure I'll never use foobar again after trying this. Enjoy
Edit the pulseaudio config by typing the following into a terminal window: Quote:
gksudo gedit /etc/pulse/daemon.conf |
Enter your root password and edit these lines in the config file: Quote:
resample-method = src-sinc-best-quality |
Quote:
default-sample-format = s24le |
s16le = 16bit, s24le = 24bit Quote:
default-sample-rate = 96000 |
You will want to pick whatever value your card outputs natively. Entering higher values than native will cause hardware resampling to continue (I put in 384000 just to see if it would work and sound still played although an error message did show in the terminal when I went to switch it back to 96000) Click save and reboot your computer.
I believe most of the music players in linux use the pulseaudio sound daemon. Rhythmbox, songbird, audacious, xmms2 are a few good ones.
I have used this on an old pentium 4 and it still doesn't consume much cpu. Rhythmbox stayed about 5-6% and pulseaudio 1-3%. Under heavy multitasking it did skip a few times. I haven't tried it on my overclocked i7 to check for skipping yet cuz the bootloader got corrupted and I need to restore it.
I might be crazy but this change really sounds good, very clear/clean/analog sounding with loads of detail. In the least you have peace of mind knowing you are degrading the signal as little as possible. I'm pretty sure I'll never use foobar again after trying this. Enjoy