Scientific Discussion / Re: A few ideas and questions about audio mediums that have been bugging me.Last post by saratoga -
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.
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! I don't know why the resampler failed though. I'm not sure but I have a feeling it -might- have to do with the encoder being picky when fed non-standard sampling rate files, 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.
Last post by Klimis -
I'm sorry that I'm making a topic about 3 separate questions but I don't think they deserve separate topics for each.
So I've been having some ideas that come together with some questions.
1. 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? 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)
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.
You're not understanding me. The Celt layer in Opus only operates at 48 KHz. 24 KHz can be implemented by decimation. This is not a matter of my opinion. It is a simple fact that you can verify by checking the spec.
xHE-AAC is not an extension to AAC. It is the branding for USAC, which is a new audio format.
If you are saying that you hope Xiph will one day create a different audio format than Opus that supports 44.1k, then you are in luck. They have one. It is called Vorbis and you can use it right now if you like.
This is wrong thinking. People say they want things all the time. To win the lottery, to be young again, etc. I can assure you that this just isn't going to happen.
So have I. I remember one person asking me why if he had a fast enough processor, why our gameboy emulator would not handle his N64 ROMs. I told him that he was wrong, that different devices required different emulators, not just faster processors. Do you think he was right and that I will be proved wrong in my belief that the N64 and Gameboy are different hardware devices?
My intention is not the typical "This sounds better than that /endofpost" but more like "It can't be just me that thinks that ..." way. I'd hope there was a way that the TOS was a bit more "loose" in a way that I don't have to struggle in order to not offend any person or violate the TOS when it's not my intention and even worse my whole point gets lost in the filtering eyes of a mod.
The intention is to prevent people from wasting everyone's time making uninformed claims about audio, and instead encourage them to go out and test their assumptions. You aren't offending people, but you are doing the exact thing everyone wants you not to do.
My biggest issue with resampling is that every now and then (very rarely though) something might go horribly wrong,
Resampling is a deterministic process. There is no now and then. It either works or it does not. We know how to make resamplers that work. Simple as that.
like there was that one time I was listening to some chiptune/PSG generated music that failed horribly (loud harmonic aliasing ).
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.
I'm not such an amateur, I know how lossy codecs work, even worse, we had to study a couple of them back in college (I thought I'd like this course when I was giving my application but I ended up hating it). My biggest issue with resampling is that every now and then (very rarely though) something might go horribly wrong, like there was that one time I was listening to some chiptune/PSG generated music that failed horribly (loud harmonic aliasing ). There were 3 available streams: AAC-HE, MP3 and OPUS. The OPUS one was the only one with the issue. And it was a pitty because it was the only high bitrate stream available.
Also btw, I just wanted to get it out of me:
I absolutely, LOATHE vinyl. There I said it.
I do not miss it and I find this retro fashion thing that goes on with vinyl at best cringeworthy.
I do get that people may have a nostalgic attachment to their vinyls and the experiences they had with them back in the day but as a format to be used today instead of any kind of proper digital format gives me this face:
PS: I hope everbody laughed as much as I did! XD
○ Soul SL300
X Soul SL300 WB
What I see is that this whole topic would have been avoided if OPUS simply supported 44.100Hz audio without the need of resampling.Why do you care about the codec having an internal resampler? It's not like Opus is the only one doing that. HE-AAC does it for SBR, MP3 does it at low bitrate to avoid encoding the high frequencies. Opus handles 44.1 kHz just fine and if you didn't "look under the hood", you wouldn't be able to tell how it does it.
I feel like it's a matter of time until it eventually gets supported (officially or not). It's possibly the last thing that holds me back making the switch from AAC to OPUS.
How is it preventing you from using Opus. What *observable* difference does it make? And BTW, there *are* ways to use 44.1 kHz directly with Opus (no, I'm not going to tell you how). They have been shown (through listening test) to produce worse results than resampling. Is the worse quality worth getting rid of the resampler?
The less layers of change applied to the source the better outcome to the final product (audible or not), with less doubts and need for research.
If something like a resampler scares you, I hope you've never looked at what a lossy audio codec does! Also, I hate to break it to you, but when you play your favorite 44.1 kHz music, it very often gets resampled to 48 kHz by your audio driver before being fed to your soundcard. At that point, the soundcard resamples it again to a few MHz because it's DAC is a 1-bit sigma-delta conversion. I would suggest you go back to good old vinyl, but even those include RIAA compensation filters. Oh and tapes have these low-pass filters to remove the ultrasonic bias. Watch out, there's filters everywhere, and they're out to get you!
Yes, I do know that. What I meant with my original post was that there may be a day that OPUS inherits a newer generation layer that may allow for such change. It's not a very accurate example but it's the best one I can think of right now is xHE-AAC (that's an extension I know). I hope you get what I mean and where I'm going.No no no.
We understood perfectly what You wanted to say.
xHE-AAC isn't a simply nice and shiny "extension". It's a new format which is only backward compatible with AAC and HE-AAC.
It means if You want to add 44.1 kHz sample rate your new "extension" won't be compatible with original Opus format. The only backward compatible sampling rate will be still 48 kHz. And I'm sure developers don't want to f*** up Opus this way.
The bitstream of Opus is frozen. https://tools.ietf.org/html/rfc6716
You can't add sampling rate without breaking compatibility.
The mobile version of fb2k seem to recognize custom ID3 tags e.g. style, chorus, instrument, soloists etc.
Unfortunately, neither Mp3tag nor foobar2000 (windows version) could tag mp3 files, that fb2k mobile recognizes the tags😢
Is there any reference, how to name the tag fields correctly?
or are the custom tags just prepared at the app interface, but not yet implememted?