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91
Lossless / Other Codecs / Re: HALAC (High Availability Lossless Audio Compression)
Last post by Hakan Abbas -
HALAC version 0.3.6 is both faster and has a better compression ratio. And the ‘lossyWAV’ results are also now more impressive.
https://github.com/Hakan-Abbas/HALAC-High-Availability-Lossless-Audio-Compression/releases/tag/0.3.6

Basically the entropy encoder stage has completely changed. This version uses Rice coding. It was a bit of a pain, but I finally finished my new Rice Coder. Of course, the results can be further improved both in terms of speed and compression ratio (we can see a similar effect for HALIC). That's why I'm delaying the 24/32 bit generalisation. No manual SIMD, GPU or ASM was used. Compiled as Encoder AVX, Decoder SSE2.
The results below show the single core performance of version 0.2.9 with version 0.3.6. I'll leave the API and Player update for later, I'm a bit tired.

https://www.youtube.com/watch?v=Pxq-WXja78Y
Code: [Select]
AMD RYZEN 3700X, 16 gb RAM, 512 gb fast SSD

WAV RESULTS (Encode Time, Decode Time, Compressed Size)
Busta Rhymes - 829.962.880 bytes
HALAC 0.2.9 Normal 2.985  4.563  574,192,159
HALAC 0.3.0 Normal 2.578  4.547  562,057,837
HALAC 0.2.9 Fast   2.010  4.375  594,237,502
HALAC 0.3.0 Fast   1.922  3.766  582,314,407

Sean Paul - 525.065.800 bytes
HALAC 0.2.9 Normal 1.875  2.938  382,270,791
HALAC 0.3.0 Normal 1.657  2.969  376,787,400
HALAC 0.2.9 Fast   1.266  2.813  393,541,675
HALAC 0.3.0 Fast   1.234  2.438  390,994,355

Sibel Can - 504.822.048 bytes
HALAC 0.2.9 Normal 1.735  2.766  363,330,525  
HALAC 0.3.0 Normal 1.578  2.828  359,572,087
HALAC 0.2.9 Fast   1.172  2.672  376,323,138
HALAC 0.3.0 Fast   1.188  2.360  375,079,841

Gubbology - 671.670.372 bytes
HALAC 0.2.9 Normal 2.485  3.860  384,270,613
HALAC 0.3.0 Normal 1.969  3.703  375,515,316
HALAC 0.2.9 Fast   1.594  3.547  410,038,434
HALAC 0.3.0 Fast   1.453  3.063  395,058,374
----------------
lossyWAV RESULTS
Busta Rhymes - 829.962.880 bytes
HALAC 0.2.9 Normal 3.063  2.688  350,671,533  
HALAC 0.3.0 Normal 2.891  4.453  285,344,736
HALAC 0.3.0 Fast   1.985  2.094  305,126,996

Sean Paul - 525.065.800 bytes
HALAC 0.2.9 Normal 1.969  1.766  215,403,561
HALAC 0.3.0 Normal 1.860  2.876  171,258,352
HALAC 0.3.0 Fast   1.266  1.375  184,799,107
92
FLAC / Re: Is Bandcamp truly lossless? 21 kHz cut-off
Last post by Porcus -
You might of course ask the band at https://www.kalandra.no/contact and hope it is read by someone nerdy enough.

Anyway, there are two free compilations at the label's bandcamp, and there are two Kalandra tracks on the most recent one. You see that they don't go all the way to 22, compared to say the top spectrogram which is off a track from label founder Einar Selvik (of Wardruna fame):
https://imgur.com/a/oVce6mG

And the first compilation:
https://imgur.com/a/Xg79O2E

Now playing: Kalandra's take on "Helvegen".
93
3rd Party Plugins - (fb2k) / Re: foo_vis_vumeter
Last post by 2tec -
Quote from: Air KEN on
@2tec / Thank you for creating such an amazing page (Skin *.bin).
You're welcome.  I add new meter skins to the list as I find them. There's also a list of Foobar2000 themes and I'm working on a theme gallery,

It's the least I could do given how much Peter and the other developers have done.

Credit should really go to all the meter skin developers who've done such an impressive job for free.
94
AAC - Tech / Re: AAC frame lengths
Last post by Klymins -
Quote from: saratoga on
The link above is for an HEv1/v2 compatible decoder.  No idea about xHE (USAC).

Normal transform codecs support two transform lengths at a time, so that is what AAC provides.  The WMA codecs were the only ones I am aware of that has supported more than 2 lengths in the same file, and 25 years later that idea seems to have been forgotten along with WMA. 



I couldn't understand what the link gives. But, do you have an idea about why normal transform codecs support two transform lengths at a time?
95
MP3 - Tech / Re: Current status of MP3 encoders
Last post by saratoga -
Quote from: fooball on
Quote from: darkalex on
The point being, it's been 20 years since that encoder came, and more like 15 years since it has had no updates... and I find it very hard to believe that it can't be improved further with the tech we have today... it's just not reasonable yk?
I don't understand where you're coming from.

MP3, AAC, whatever, have defined methods for pre-processing and encoding, then decoding and post-processing.  Modern tech makes it faster to perform those stages in software (or implement them directly in hardware), but it can't change the basic architecture of the processing stages otherwise it wouldn't be MP3/AAC.

Minor tweaks to the psycho-acoustic model won't yield the step improvement in "quality" you're looking for, and might result in a split opinion whether the tweak is actually better.  Major tweaks would be the equivalent of ripping it up and starting again – ie not MP3/AAC at all, a new codec.

Codecs are limited by the era they were created in.  In particular, the decoder had to be able to decode in real time on the hardware available in that era, which put a ceiling on the complexity of architecture which could be accommodated.  You can't now turn around and say "we have much faster hardware now, so let's redefine the architecture of MP3".

AAC is much more recent than MP3, therefore has the capacity to be much better than MP3, but I suggest that both have reached their ceilings (or very close to them).

The ironic thing is that MP3 decode complexity is actually relatively high by modern standards since you have to do both the MDCT and filterbank.  AAC-LC is quite efficient in that you just do a single transform.  Aside from the low bitrate extensions like SBR, newer codecs have not much changed the overall decoder complexity much since the 1990s, they've just used that complexity more cleverly. 
96
FLAC / Re: Is Bandcamp truly lossless? 21 kHz cut-off
Last post by saratoga -
Quote from: Argonil on
Quote from: saratoga on
Quote from: Argonil on
Hmm, you are right. I just assumed since none of the albums I've purchased were higher than 44.1 kHz.

I'm still curious about what the cut-off actually means though. What happened? Is it lossless?

It means the audio is 44.1 KHz.  If you want higher cut off, look for a 48k or 96k album.

Other 44.1 kHz FLAC files go all the way up to 22 kHz, filling it out entirely.

Those are probably downsampled from some higher sampling rate or have added dithering or other effects added.

Quote from: Argonil on
Are you saying it's a bad FLAC? It's the only one I can buy...

It sounds better than YouTube at least. Guess I did get something for my money, even if it's not everything it should be.

Is there a better way to analyze the quality of a sound file?

It isn't good or bad, that is just what audio looks like on a spectrum analyzer.  For a better approach, try a coin toss. 
97
AAC - Tech / Re: AAC frame lengths
Last post by saratoga -
The link above is for an HEv1/v2 compatible decoder.  No idea about xHE (USAC).

Normal transform codecs support two transform lengths at a time, so that is what AAC provides.  The WMA codecs were the only ones I am aware of that has supported more than 2 lengths in the same file, and 25 years later that idea seems to have been forgotten along with WMA. 

98
FLAC / Re: Is Bandcamp truly lossless? 21 kHz cut-off
Last post by Argonil -
Quote from: Wombat on
And there are lousy filters that allow tons of aliasing but they will show content up to 22.05kHz with these funny spectral pics.
The first pic looks sanity bandwidth limited.
Are you saying it's a bad FLAC? It's the only one I can buy...

It sounds better than YouTube at least. Guess I did get something for my money, even if it's not everything it should be.

Is there a better way to analyze the quality of a sound file?
100
FLAC / Re: Is Bandcamp truly lossless? 21 kHz cut-off
Last post by Argonil -
Quote from: ktf on
This is a result of how sampling works. See https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem

In order for digital audio to work, a low-pass is required. Some do this with a low-pass filter with a very, very steep cut-off, which means frequencies almost all the way to 22kHz are preserved. Others prefer a less steep cut-off, because humans don't actually hear much beyond 20kHz anyway.

In fact, when 44.1kHz was chosen as the sampling rate for CD audio, one of the considerations was that this leaves some room between 20kHz hearing limit and 22.05kHz theoretical sampling limit for the high-pass filter

So, this has nothing to do with being lossless or not. This is simply a choice made either conciously during mixing or mastering, or a result of whatever analog-to-digital device was used when recording.

Does that means that the reason you can tell in Spek that, for an example, a file was originally a 128 kbps MP3, it's because the 128 kbps compression uses a much more aggressive low-pass filter which cuts it off at a much lower frequency (e.g. 16 kHz or whatever it is)?
So the frequency cut-off and the actual quality aren't entirely correlated? A FLAC that cuts off at 16 kHz could have higher quality than a 128 kbps MP3 even though they cut off in the same place?