Fabithierry
New Head-Fier
- Joined
- Jan 17, 2015
- Posts
- 44
- Likes
- 16
I usually use J river and Foobar, but i tried Winamp with this plugin and i think that this plugin is much better than the Wasapi plugin of Foobar. Great Job!
I usually use J river and Foobar, but i tried Winamp with this plugin and i think that this plugin is much better than the Wasapi plugin of Foobar. Great Job!
It happened to me too on my Windows 7 x64. It's a buffer issue ; you can fix it in the plugin settings (client buffer size in milliseconds / client invalidate & reload / genuine latency formula).
Try to raise the buffer to let's say 400 or 500 ms (for a start) and click client invalidate & reload. Then try lower values or even 0 (auto buffer). Click client invalidate & reload each time you enter a new value.
BTW my soundcard default settings (Windows) are 24bit / 44KHz.
CD ripping:
When ripping cd's I suggest you use EAC, but if you are gonna rip ALOT of cd's, this method may take alot of time.
Winamp have a really easy ripping feature, and it gets all the info like band/album/song from gracenote automatically .
If you are gonna rip a large amount of cd's in FLAC quality, this may save you some time.
I suggest to rip in FLAC with the best compression, you do this by:
Options ---> Preferences ---> CD Ripping ---> choose FLAC - best compression
-BleaK
As a new member, let me ask this: what is the advantage of ripping on EAC over Winamp? Is there any improvement in sound quality or fidelity towards the original CD?
I see now. Well, I guess I'll use EAC for mi favorite albums and keep using Winamp for the rest of my library. Thanks for the info.
I've still no clue on how to set the sampling rate. Most of my files are 16/44.1.
It's all in the first post.
Preferences -> output -> Maiko Wasapi -> configure.
Then in the top you choose exclusive:
Then you tick all the boxes like the picture, and if you want to have 16-bit and 44,1kHz you must put "slave to input sample rate": 44100 , and "slave to input encoding when enabled" to 16.
err so i change first box from 0 to 16 and
second box stays 44100?
No you change it from 0 to 44100 or other, down the page you see "24+padded", change that to 16.
May I ask why you are doing this?
Most of my files are 16bit,44.1khz
I do not want any resampling to occur which decrease audio quality.