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Opus / Re: Opusenc's built-in resampler
Last post by bennetng -
What really matters are not the resampling and spectrogram. It's the sound. For example the file here when encoded with 35kbps, opus has more audible distortion than fhgaac.;topic=113187.0;attach=10692

Again the encoded files are attached for further inspection. I only used foobar's encoder pack and front end to encode the files, without any custom tweaking. Pre-SoX resampled opus file also attached to show the distortion is not caused by bad resampling.
1. What kind of technical advantages did they apply to DCC format ( 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? 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)
First of all, the DCC uses 9 tracks per side, as opposed to 2 tracks on analog tape (1 of 9 tracks is an auxiliary track, so the audio actually uses 8 tracks). To compensate for this, you would have to run the tape at 4 times the speed.

The crucial parameter for the achievable bitrate is the bandwidth of the medium. If you would record the bits as is, you would need a bandwidth from zero up to half the bit rate. If you use some kind of modulation, it is the properties of the modulation that determines the bandwidth. A tape doesn't have a bandwidth down to zero, so using a modulation is unavoidable. DCC uses an 8/10 modulation that makes 10 bits out of 8 bits. I don't know the exact code used, but it is certainly chosen to minimize bandwidth requirements, by consisting of runs of more than 1 consecutive 0-bit or 1-bit. For example, the code would avoid single 0-bits surrounded by 1-bits, or vice versa.

A total data rate of 384 kbits/s divides down to 48000 bits/s per head. In addition to 8/10 modulation, there's further data added for error correction. So we end up with 72000 bits/s per head. If the encoding ensures run lengths of at least 2, the raw bandwidth demand would be around 72000 bits/s divided by 4, or 18000 bits/s, which is achievable with good tape, given that the head gaps are going to be very small. In practice the necessary bandwidth will be somewhat wider, but I'm concerned with ballpark figures here. So this rough estimate would indicate that it can work.

If you want to duplicate this with an ordinary analog tape recorder, speeding up the tape 4-fold would be unlikely to suffice. Even if you used the encoding in DCC in an analogous form, you'd still find that you would need tape heads with finer gaps, and align them more accurately.

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). I do get that it would mean that they wouldn't be backwards/redbook compatible but it's not like other competing formats were. There would be so many advantages, especially compared to rival formats, like:
*The actual hardware part of the reader is the cheapest one out of all the competitors and it's going to be present either way on the device that reads the actual disc.
*The medium is also the cheapest one compared to SACD discs and DVDs
*Compressing the audio data would open the gate to compete directly with SACDs by offering higher sampling rates, higher bit depths or even longer playtimes.
*Longer playtimes would eliminate the extra costs for more pressings with multiple discs that might not be needed.
*Ability to include more metadata.
*Ability to introduce somekind of strong(er than redbook technology) copy protection on the disc.
Copy protection helped to kill the other formats, too. I don't think your proposed variant would have escaped this fate.

I don't think your list of features contains anything compelling enough to overcome people's hesitation to embrace yet another format. Multiple competing formats is bad for the consumer market. We have seen this over and over again. The long-time success of the CD shows that its feature set was pretty well defined and adequate even after decades. Kudos to the foresight of the inventors.

What's the third question?
General - (fb2k) / Re: Default UI Gallery
Last post by nelive -
Just saw this topic and wanted to share my layout. Simple design with panels to display artist photo and album arts (front, back and disc covers) and other information like album notes, translation of non-English titles or version of the release.

Default components + Lyric Show Panel 3 (top right) + Playlist Organizer (bottom right) + Text Display (bottom bar), on macOS :D
Support - (fb2k) / mactype incompatibility
Last post by スラッシュ -
Hi!  :)) This is very likely not your fault at all, but I thought I would report it either way--the Win10 Creators Update has broken compatibility between MacType and Foobar, specifically it crashes when you try to open any file's properties or the Preferences window (and possibly other scenarios).

I've included a link to the failure dmp and txt files that were generated upon crash: DMP and TXT!

Like I said, this never happened before the Win10 Creators Update, so it's very likely to be a problem on that end (or with MacType), but I guess I just wanted to mention it just in case it can be fixed or is a symptom of a bigger problem.
3rd Party Plugins - (fb2k) / Re: foo_musicbrainz
Last post by Kraeved -
Thank you for continuing to work on the plugin.
I have also tried Build C at 128 kbps but Opus gets into transparency region very quickly at this bitrate.

Only the most critical samples were tested at 128 kbps. (Linchpin and Castanets).

A graph of previous 64, 96 kbps results and two samples at 128 kbps.

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

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:

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.
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.