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1
B&W's 'Society of Sound' is now only providing AIFF files & not FLAC as previous.
Having just back to them, it is possibly relevant that I copy/paste the latest to them.
"It is only today that I actually tried to play them as a complete series from a particular album.
Previously I could select all the tracks & 'dump' them into Foobar, ie, "Play in Foobar 2000" or "Enque in Foobar 2000".
With the IAFF files, it does not give this 'option, (can only select "Play") & if a 'group' of tracks selected will do strange things such a briefly playing a track & then going to the next & continuing in similar fashion.
It WILL play the whole track if only ONE track selected, (but this is obviously quite useless, particularly if wishing to listen to the 'Verdi Requiem').
I do not expect you to be the expert on AIFF & Foobar 2000, but perhaps you can check with 'others' as to the problems I relate, as to not be able to 'effectively' use the 'Society of Sound' music would be disappointing."

I think this shows the actual problem, but to mention that when I enquired initially, they indicated that I should have no trouble with the 'new' AIFF files.
So, I guess basically to ask if you agree that AIFF should be no trouble using Foobar 2000, ( I just updated just in case it made a difference). And particularly as to my findings as provided to B&W as above, can you assist me.
Regards & Thanks,
John Haines

2
Hey @NEMO7538, here's what I sometimes see:
I forced foobar2000 to save its config (available when holding down shift when opening File menu) and first time I got the error again, second time seemed OK. So my Playlist Organiser window is still intact. Anyway, just FYI:



3
Opus / Re: Opusenc's built-in resampler
Last post by jmvalin -
Heh, It worked!
If you edit the sample rate of a WAV file with a hex editor and change 44 AC with 80 BB, encode the file to opus and the force opus to spit the samples out at 44100Hz you essentially did it. I don't hear anything wrong with the output file. Is there a sample file that might introduce audible issues because in theory it shouldn't.
You realize that in doing so, you're not only causing the max coded frequency to be around 18 kHz rather than 20 kHz, but you're making all the psychoacoustics wrong by moving all the critical bands? You could also just flip the stereo bit while you're at it, no? Despite any class you might have had in college, I don't think you actually understand much of lossy codecs. If you *actually* want to learn something about psychoacoustics, I would suggest starting here.
4
Opus / Re: Opusenc's built-in resampler
Last post by saratoga -
Is there a sample file that might introduce audible issues because in theory it shouldn't.

Depends on the bitrate.  You're basically making all the encoders assumptions about frequency be off, so nothing will be encoded quite right.  I bet it sounds ok though if you throw enough bitrate at it.  The difference is frequency is close enough that it'll just make everything a little less accurate, but I doubt it breaks anything completely. 
5
Wiki says it just writes more tracks in parallel, so it'd have more data per length.
Now that's what I don't get. While it has 9 tracks, 8 are for audio (4 for each side and one for auxiliary/data) which resulted in a datarate of ~384Kbps.
Analog cassete tapes have 4 tracks, 2 for each side. If we treat the cassette as a one side only (like how old-school 4 track cassete recorders worked (see Tascam MF-P01) we are hundreds of Kbps away from achieving such data rate. Now even with 2 tracks for each side with half the datarate (192Kbps) even then we are really far from achieving such datarate. What is the reason behind such issue? What was the magic into these DCC tapes that made them possible to achieve such datarate?

http://audiophilereview.com/cd-dac-digital/in-praise-of-the-sony-pcm-f1.html

I remember reading about this after stumbling accross DVHS:
https://en.wikipedia.org/wiki/D-VHS
Fancy stuff!
6
3rd Party Plugins - (fb2k) / Re: foo_dsp_effect
Last post by mudlord -
use the seperate component.
7
Quote
Wikipedia says 17 KB/s for modern software, although I bet a good head is required to do that.
In the late '80s one could record digital CD quality on VCR tape.

http://audiophilereview.com/cd-dac-digital/in-praise-of-the-sony-pcm-f1.html
8
Opus / Re: Opusenc's built-in resampler
Last post by saratoga -
Is the implication that you believe this has something to do with resampling?  If so, you should explain carefully how you came to that conclusion. 
Yep. It sounded exactly like when you apply nearest neighbor algo on samples. Harmonics everywhere! No filtering, nothing!

If you really did base that determination on nothing, then this thread will have been a great example of why we have TOS#8. 

I bet the ones that were fed to it were either slightly above or slightly below 32KHz or 44,1KHz. I repeat again this is a wild guess, I can't know what the streamer did behind his desk.

I would have blamed clipping, but no, its not due to the sampling rate being off.  All that does is introduce a pitch error. 
9
Opus / Re: Opusenc's built-in resampler
Last post by Klimis -
And BTW, there *are* ways to use 44.1 kHz directly with Opus (no, I'm not going to tell you how).
You actually gave me an idea, I doubt it's the same thing but let me try it.

EDIT:
Heh, It worked!
If you edit the sample rate of a WAV file with a hex editor and change 44 AC with 80 BB, encode the file to opus and the force opus to spit the samples out at 44100Hz you essentially did it. I don't hear anything wrong with the output file. Is there a sample file that might introduce audible issues because in theory it shouldn't.
10
What kind of technical advantages did they apply to DCC format (https://en.wikipedia.org/wiki/Digital_Compact_Cassette) that made it possible to carry digital audio into a tape that a regular tape as a physical format (TYPE II or III let's say) can't do?

Wiki says it just writes more tracks in parallel, so it'd have more data per length. 

I mean if you would use the same type of modulation and recording speed wouldn't you be able to store digital audio on a regular tape? I'm really trying to find a way to store digital audio on a tape as a project but the modulation methods available and the software available for storing digital data on tapes is not enough to achieve the robustness and the bitrate required for such project. Also I wasn't able to find enough documentation on the format which makes such project even harder)

Yes you can.  This was super common back in the 1980s.  It is less common now though due to the relatively low datarate.  In the old days capacity was super low (less than 1 KB/s).  I bet with modern processing power you can do better.  Wikipedia says 17 KB/s for modern software, although I bet a good head is required to do that.

2. How come there was never a competitor to redbook CDs that actually uses somekind of lossless compression (let's say like DVD-A can use Meridial Lossless).

By the time the hardware was available to do something like that cost-effectively, CDs were so completely entrenched that there was no room for an alternative format.