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PC Audio foobar2k Q&A, Guides, help & comments are welcome...

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Angel of Deth
post Apr 28 2009, 10:46 PM

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QUOTE(Najmods @ Apr 26 2009, 11:49 PM)
The reason why some resample to 48kHz because their cards (usually old AC97 based codec and Audigy series cards, which resamples everything to 48kHz) have bad resampling algorithms that create intermodulation distortion that even software can do it better, so they resamples the source first before the DSP itself do it.

About resampling in foobar, there is resampler in DSP Manager, but I don't know how it done in Vista as I never ever use Vista
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So if i dont change anything in Foobar, it will play my FLAC audio as 44.1 hz right?? And Replay Gain doesn't change anything and still produce the original source?
TSgac
post May 12 2009, 02:19 AM

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QUOTE(Angel of Deth @ Apr 28 2009, 10:46 PM)
So if i dont change anything in Foobar, it will play my FLAC audio as 44.1 hz right?? And Replay Gain doesn't change anything and still produce the original source?
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generally yes to all. smile.gif
Najmods
post May 12 2009, 11:35 AM

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Replay Gain acts like a normalizer, it normalize all the selected tracks so you don't have to adjust the volume when changing track. Its up to you whether you want to use it or not, try to do some testing, if you like it use it.

Be wary though that it might cause distortion, as it makes the quieter track louder if its mixed with louder track. I don't know how effective this Replaygain is as I never use it.

Read about it here
TSgac
post May 13 2009, 01:02 PM

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i use replay-gain all the time, for my quality CDs collections after converting to FLAC, working very fine. the track gain averagely +/- 1.5dB unless some with more than that kinda range.
fariz
post May 13 2009, 04:59 PM

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QUOTE(Angel of Deth @ Apr 28 2009, 10:46 PM)
And Replay Gain doesn't change anything and still produce the original source?
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ReplayGain is similar to turning your volume knob down because you find one track's louder than the next. It's just an automated way of doing that. Since the value is stored in the tag, your source will remain the same.

more detail info: http://www.head-fi.org/forums/5492266-post2.html
Angel of Deth
post Jun 28 2009, 12:05 AM

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ASIO buffer - the higher the better or the least (low latency) is better?? I know latency is not an issue when you're not recording, but i really want to know.
andrew9292
post Jun 28 2009, 12:16 AM

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Does Replay Gain apply gain in waveform(audio frequency signal) or is it like controling the program's master volume.

Also what is Dither for?


Added on June 28, 2009, 12:19 am
QUOTE(Angel of Deth @ Jun 28 2009, 12:05 AM)
ASIO buffer - the higher the better or the least (low latency) is better?? I know latency is not an issue when you're not recording, but i really want to know.
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If u're not recording, higher latency better. It does not use up alot of CPU when u set the latency higher. Try setting lowest latency on your sistem, see if it lags or not, on my pentium 4 2.4Ghz it eats up my CPU time like crazy...even opening webpage will lag. There is also no difrerent in sound when higher latency...

If u want me to be more techincal, lower latency will use up more CPU which will jam up the system bus etc and cause the board to produce more EMI interference in the PC, which might affect sound, but heh, i think only dolphines can hear the difference.... biggrin.gif

This post has been edited by andrew9292: Jun 28 2009, 12:22 AM
Angel of Deth
post Jun 28 2009, 12:24 AM

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QUOTE(andrew9292 @ Jun 28 2009, 12:16 AM)
Does Replay Gain apply gain in waveform(audio frequency signal) or is it like controling the program's master volume.

Also what is Dither for?


Added on June 28, 2009, 12:19 am

If u're not recording, higher latency better. It does not use up alot of CPU when u set the latency higher. Try setting lowest latency, on my pentium 4 2.4Ghz it eats up my CPU time like crazy...even opening webpage will lag. There is also no difrerent in sound when higher latency...
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But if more buffer it will eat your RAM?
andrew9292
post Jun 28 2009, 12:30 AM

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QUOTE(Angel of Deth @ Jun 28 2009, 12:24 AM)
But if more buffer it will eat your RAM?
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If TOO MUCH buffer yes, it will cause lagginess.

If TOO little also it will cause lagginess, because the program is struggling to keep up with such little buffer length by increasing program priority so that the music doesnt lag or stop and continue when u multitask with processer and RAM heavy software (eg. photoshop) So like when you photoshop, the music is still smooth with small buffer, but your photoshop program will suffer.

basically if u use a new generation processor like core2duo, duo core etc...u dont need to worry so much about this. those using pentium 3, celerons or early pentium 4...those without HT will feel the performance drop.

This post has been edited by andrew9292: Jun 28 2009, 12:34 AM
Angel of Deth
post Jun 28 2009, 12:46 AM

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QUOTE(andrew9292 @ Jun 28 2009, 12:30 AM)
If TOO MUCH buffer yes, it will cause lagginess.

If TOO little also it will cause lagginess, because the program is struggling to keep up with such little buffer length by increasing program priority so that the music doesnt lag or stop and continue when u multitask with processer and RAM heavy software (eg. photoshop) So like when you photoshop, the music is still smooth with small buffer, but your photoshop program will suffer.

basically if u use a new generation processor like core2duo, duo core etc...u dont need to worry so much about this. those using pentium 3, celerons or early pentium 4...those without HT will feel the performance drop.
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Thanks for your advice, now i feel i need to forget about bit-perfect. Now i know bit-perfect is subjective because there is no absolutely perfect in technical aspect, even if you use ASIO + blablabla cable + RM100k speaker + Rm233k cd player can't gurantee 100% bit-perfect. At least i love the sound coming from my speaker, that is enough for me.
fariz
post Jun 28 2009, 07:24 AM

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QUOTE(andrew9292 @ Jun 28 2009, 12:16 AM)
Does Replay Gain apply gain in waveform(audio frequency signal) or is it like controling the program's master volume.
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Volume is adjusted on playback. Since it's metadata based (stored in tag), the audio data is not modified. The desired playback effect can be switched back and forth in the playback setting.
TSgac
post Jul 11 2009, 04:30 AM

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how to configure the album art, why cant i find the setting like before after installing the latest version sweat.gif
fariz
post Jul 14 2009, 02:07 PM

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QUOTE(gac @ Jul 11 2009, 04:30 AM)
how to configure the album art, why cant i find the setting like before after installing the latest version  sweat.gif
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Default UI and Columns UI can read album arts.

Default UI through UI Element "Album Art Viewer": no configuration needed
Columns UI through NG playlist or Album Art Display: should be shown automatically as far as I know. See Preferences > Columns UI > Tab "Artwork".
fariz
post Jul 14 2009, 02:08 PM

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DVD-Audio Decoder
DVD-Audio Decoder input plugin for foobar2000. Decoder is capable of playing back DVD-Audio discs and MLP files in full resolution.

http://dvdadecoder.sourceforge.net/
TSgac
post Jul 14 2009, 06:47 PM

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Thanks, used Columns UI everything superbly fine now smile.gif


QUOTE(fariz @ Jul 14 2009, 02:07 PM)
Default UI and Columns UI can read album arts.

Default UI through UI Element "Album Art Viewer": no configuration needed
Columns UI through NG playlist or Album Art Display: should be shown automatically as far as I know. See Preferences > Columns UI > Tab "Artwork".
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Only DVD-Audio??? can the normal DVD RW read DVD-Audio? i hv been looking for DVD-A for long time, cant a single dics sweat.gif

QUOTE(fariz @ Jul 14 2009, 02:08 PM)
DVD-Audio Decoder
DVD-Audio Decoder input plugin for foobar2000. Decoder is capable of playing back DVD-Audio discs and MLP files in full resolution.

http://dvdadecoder.sourceforge.net/
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This post has been edited by gac: Jul 17 2009, 05:45 PM
TSgac
post Jul 17 2009, 05:45 PM

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how to get the downloaded lyrics deleted? shakehead.gif
gilbertarenas
post Jul 22 2009, 09:40 PM

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foobar is good, better than the china ttplayer, but the lyrics function can be better
andrew9292
post Jul 22 2009, 09:53 PM

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QUOTE(gilbertarenas @ Jul 22 2009, 09:40 PM)
foobar is good, better than the china ttplayer, but the lyrics function can be better
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But foobar's SSRC resampler causes average +1dB in the signal when resampling, which is distortion...u can see the in the visualisation "Peak Meter", when the meter crosses over the max (0 dB) to become +X dB.... TTPlayer's resampler does not...
TSgac
post Jul 23 2009, 12:31 AM

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QUOTE(andrew9292 @ Jul 22 2009, 09:53 PM)
But foobar's SSRC resampler causes average +1dB in the signal when resampling, which is distortion...u can see the in the visualisation "Peak Meter", when the meter crosses over the max (0 dB) to become +X dB.... TTPlayer's resampler does not...
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dont use resampler lah.... originality is always the best! wink.gif
andrew9292
post Jul 23 2009, 12:24 PM

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QUOTE(gac @ Jul 23 2009, 12:31 AM)
dont use resampler lah.... originality is always the best!  wink.gif
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I need the resampler because my Audigy 1 soundcard have to up-sample to 48khz for the DSP to process, and the resample algorithm/chip on that card does it badly...bass becomes soft...weak vocal...harsh and dead trable

If SSRC resample for me to 48khz, the Audigy doesnt need to do that lousy resampling anymore...it just bypasses that lousy on-card resampler...

Last time i use foobar to resample, but got distortion, so now using TTplayer...sound much smoother rclxms.gif

And yeap, only audigy and sound blaster live based cards needs to SOFWARE resample due to the lousy on-card resampler... others like X-fi already fix that issue...

This post has been edited by andrew9292: Jul 23 2009, 12:26 PM

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