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Recent Posts
11
3rd Party Plugins - (fb2k) / Re: 64-bit recompiles
Last post by Squeller -
Really great initiative of you. Still happy.
CASE, on top of your web page, can you also link Peters official component page? I felt the itch, others may feel the same, to look and compare, thx.
12
Other Lossy Codecs / Re: LossyFlac vs Opus 256
Last post by Squeller -
I'd recommend you to find Gurus listening tests and go for a robust, very compatible (maybe you want to use your converted files elsewhere) and space saving format: Apple LC AAC. Become more opportunistic and don't waste too much energy into theoretical problems. Example: In one car, I can only use 16 GB sticks, so I encode to Q64 or Q73 (say, q64, depending on the audio, ranges from ~130 down to 80 kbps, pads, lowpassed, piano music, in general few transients seem to qualify for low bitrates). Totally perfect. In another car, I use LAME mp3 @V2 - just because that one does not display embedded album art from the AAC files.
Good headphone, I personally would probably go for Apple AAC @ q82 or q91
And don't hunt for problems, it's the beginning of the end. It distracts. Enjoy the music. Don't forget, your cat walking by has a bigger impact on your audible experience than the almost academic problems we tend to find.

I'm saying that, because in the past I perfectly heard if magnetic tape heads were not correctly adjusted. Hence I couldn't 100% enjoy the music, everyone else could.

Learn to make things right but then free yourself from over-critical listening ("i feel like i'm able to hear small differences in background drums"). It makes few sense. In a concert, there is huge differences at every moment because of the audience.
13
FLAC / Re: FLAC v1.4.x Performance Tests
Last post by ktf -
Thanks. It turns out that some changes I made when working out the 32-bit encoder/decoder did affect the 24-bit part more than I thought. I was under the impression the code paths meant for 32-bit audio were only seldomly used for 24-bit audio, but it turns out certain kinds of 24-bit audio (especially those with a completely empty upper half of the spectrum) do use these code paths a lot, and they are much slower.

So, these changes make the choice between these code paths more strict: that choice was rather made rather roughly (on the safe side of course), but now the encoder goes through a little bit more math to only choose the slow code path when absolutely necessary.

The speed-up is highly dependent on source material. Audio with a high samplerate in which the upper frequencies are fully 'utilised' do not see any change at all, most audio I've tested sees quite some improvement at preset 8, and those where really no audio exists above 20kHz see most improvement at preset 5, but are still slow at preset 8.
14
Other Lossy Codecs / Re: LossyFlac vs Opus 256
Last post by Nick.C -
Quote from: guruboolez on
-extraportable is the lowest LossyFLAC preset. Quality seems really high but I don't think it's supposed to be fully transparent.
lossyWAV Beta 1.4.3c includes, for testing at this time, four more quality options below extraportable (-q -5): "unadvised" (-q -10), "dubious" (-q -8.75), "questionable" (-q -7.5) and "aggressive" (-q -6.25). These will yield lower resultant lossyFLAC bitrates, with the consequent increased risk of noticeable added noise as the quality numeral reduces.
16
3rd Party Plugins - (fb2k) / Re: foo_truepeak True Peak Scanner
Last post by Case -
Quote from: Defender on
Quote from: Case on
If you are scanning just one track the scanning speed will be limited by the source file decoder. Scanning is multithreaded but threads are used only for handling multiple tracks at once.
Not sure what you mean. Can truepeakscanner (and/or foobar RG scanner) use multithreaded scanning on multiple tracks/subsongs within a single container (iso/mkv)?
Absolutely. As long as there are separate tracks the work can be spread across multiple threads. You can look at task manager and you should see possibly even 100% CPU usage for foobar2000, as long as you aren't bottlenecked by storage.
17
Polls / Re: Sound colors of codecs
Last post by sld -
I just prefer that the codecs and bitrates I use don't mess with the original input file according to my ears.
My ears are pretty ok, but I guess the important part here is "don't mess". The only time things get messed with are:
1) Out of my control - the budget hardware I have
2) Within my control - EQing
18
3rd Party Plugins - (fb2k) / Re: foo_vis_spectrum_analyzer
Last post by TF3RDL -
Quote from: TF3RDL on
Finally, we got the IIR filter bank spectrum analyzer for foobar2000, so if I see this actually working, I don't need Voxengo AnSpec or foo_dsp_vst3 anymore just for visualization
I've tested both of them but SWIFT and analog-style analyzer mode as in my own CodePen project aren't implemented properly, it currently look like this (music):
X
and this (sine wave sweep):
X
for both cases for screenshots of analog-style (IIR filter bank) mode

Also, both SWIFT and analog-style analyzer's output doesn't get reset upon seek (also applies to peak decay and time smoothing) unlike foo_enhanced_spectrum_analyzer
20
Scientific Discussion / Re: Are complex-input FFTs really useful for audio analysis?
Last post by TF3RDL -
Quote from: mycroft on
Its just smart optimization, to make faster processing.
Using 2 FFT transforms for real-only inputs is stupid, if you insist on using separate transforms use RDFT and not FFT.
That's why treating the stereo input as complex numbers (useful for SDR-related I/Q signals, but idk for audio, assuming you skipped the process of turning into a stereo FFT) in the aforementioned CodePen project is faster than without, and that could be helpful optimization for @Crossover's foo_enhanced_spectrum_analyzer when comes to a feature request to display multiple channels but I'd imagine it only works on stereo pairs (especially on surround sound formats where you have center and LFE channels in-addition to usual channel pairs)