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    How to obtain same volume while listening to the converted opus file in android phone with headphones.

    I want to get same volume throughout the file I mean I don't want ups and downs while listening to opus file. Is there any DSP to achieve this.

    Suppose while listening to original mp3 source file at point A I got 90 % of maximum volume of my headphones and at point B I got 50% percent of maximum volume of my headphones and at point C I got 95 % of maximum volume. I want to convert to convert this mp3 file to opus file and if I listen this opus file in android phone with headphones I must get say 80 % of maximum volume of my headphones I mean there must be no ups and downs say 90 % at point A, 50 % of maximum volume at point B, 95 % of maximum volume at point C.
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General Audio / Re: AES 143rd NY 2017
Last post by ajinfla -
(too bad if you'r not on FB).
Why? I'm not on FB and I can see the comments. Or did you mean something else?
Wasn't sure if non FB members can see all the comments below his, so was trying to stave of the inevitable bitching about Facecrook (much of it justified).
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AAC - General / Re: qaac under EAC quations
Last post by Case -
Refalac is short for reference ALAC. It's an ALAC encoder based on Apple's reference source code. It can not encode AAC but it's a standalone program not requiring any additional dlls or installed programs.

Qaac is a command line frontend for iTunes' libraries. It can encode both AAC and ALAC but it requires having iTunes installed (or its libraries extracted to a directory it can find them from).

The produced ALAC files will not be identical but ALAC is a lossless codec and both encoders will produce files that decode perfectly to the original source file.

If you have the drive space I'd recommend not encoding to AAC with EAC. CD ripping is slow and most people don't want to do it several times. If you use a lossless codec such as ALAC you'll have a perfect copy that can be later reconverted quickly to a different lossless format. And you can quickly make AAC versions for special uses as required.
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General Audio / Re: AES 143rd NY 2017
Last post by danadam -
(too bad if you'r not on FB).
Why? I'm not on FB and I can see the comments. Or did you mean something else?
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General Audio / AES 143rd NY 2017
Last post by ajinfla -
Anyone attend?
Seems there were some rather interesting exchanges at a certain "Hi Re$" session, I think this one
Bruno Putzeys take https://www.facebook.com/permalink.php?story_fbid=2044350049128525&id=100006606498666
Quote
This isn't a prelude to suddenly becoming active on FB but I felt I had to share this.
Yesterday there was an AES session on mastering for high resolution (whatever that is) whose highlight was a talk about the state of the loudness war, why we're still fighting it and what the final arrival of on-by-default loudness normalisation on streaming services means for mastering. It also contained a two-pronged campaign piece for MQA. During it, every classical misconception and canard about digital audio was trotted out in an amazingly short time. Interaural timing resolution, check. Pictures showing staircase waveforms, check. That old chestnut about the ear beating the Fourier uncertainty (the acoustical equivalent of saying that human observers are able to beat Heisenberg's uncertainty principle), right there.
At the end of the talk I got up to ask a scathing question and spectacularly fumbled my attack*. So for those who were wondering what I was on about, here goes. A filtering operation is a convolution of two waveforms. One is the impulse response of the filter (aka the "kernel"), the other is the signal.
A word that high res proponents of any stripe love is "blurring". The convolution point of view shows that as the "kernel" blurs the signal, so the signal blurs the kernel. As Stuart's spectral plots showed, an audio signal is a much smoother waveform than the kernel so in reality guess who's really blurring whom. And if there's no spectral energy left above the noise floor at the frequency where the filter has ring tails, the ring tails are below the noise floor too.
A second question, which I didn't even get to ask, was about the impulse response of MQA's decimation and upsampling chain as it is shown in the slide presentation. MQA's take on those filters famously allows for aliasing, so how does one even define "the" impulse response of that signal chain when its actual shape depends on when exactly it happens relative to the sampling clock (it's not time invariant). I mentioned this to my friend Bob Katz who countered "but what if there isn't any aliasing" (meaning what if no signal is present in the region that folds down). Well yes, that's the saving grace. The signal filters the kernel rather than vice versa and the shape of the transition band doesn't matter if it is in a region where there is no signal.
These folk are trying to have their cake and eat it. Either aliasing doesn't matter because there is no signal in the transition band and then the precise shape of the transition band doesn't matter either (ie the ring tails have no conceivable manifestation) or the absence of ring tails is critical because there is signal in that region and then the aliasing will result in audible components that fly in the face of MQA's transparency claims.
Doesn't that just sound like the arguments DSD folks used to make? The requirement for 100kHz bandwidth was made based on the assumption that content above 20k had an audible impact whereas the supersonic noise was excused on the grounds that it wasn't audible. What gives?
Meanwhile I'm happy to do speakers. You wouldn't believe how much impact speakers have on replay fidelity.
________
* Oh hang on, actually I started by asking if besides speculations about neuroscience and physics they had actual controlled listening trials to back their story up. Bob Stuart replied that all listening tests so far were working experiences with engineers in their studios but that no scientific listening tests have been done so far. That doesn't surprise any of us cynics but it is an astonishing admission from the man himself. Mhm, I can just see the headlines. "No Scientific Tests Were Done, Says MQA Founder".
Very interesting comments by Paul Frindle as well (too bad if you'r not on FB).
Finally a bit of pushback on MQA and other assorted nonsense?

cheers,

AJ
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AAC - General / Re: qaac under EAC quations
Last post by AlexQc -
So refalac can be used to encode AAC and not just ALAC??

What's the difference between  qaac.exe and refalac.exe? Don't the resulting files are identical?

Thanks!
7
Also, if I may add, and any foobar developers are reading this - as a programmer myself, I can't help but notice that the way Foobar handled this crash was pretty bad. The situation was that the computer is left with no RAM, causing foobar to crash. My question is why does Foobar attempt any writing with no available memory? I know it's usually saving all the data upon the closing, but in this case it resulted in overwriting empty bits over the actual data...Simply no writing whatsoever in a special circumstance like this would be much better.
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Validated News / Re: Opus 1.2 is out!
Last post by Ajaja -
Could anybody compile recent revision, I would like to test this fix
https://git.xiph.org/?p=opus.git;a=commit;h=b30f45b9a8bfc7b97afb75042bf2ab16a2150972
Thanks
opus-tools x64 v0.1.10-9-gbd65450 (using libopus v1.2.1-25-g43dfdc08)
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3rd Party Plugins - (fb2k) / Re: foobar2000 DeskBand Controls
Last post by fuffi -
When I'm skipping a song (from within the default f2k control button: next) the stub image is shown for about 500ms.
So it takes over 500ms to skip to the next song on your setup?
I've set the delay to 200ms... I guess I just make this delay configurable to be able to handle slower setups.
You are right. Its shorter than 500ms. About 200ms as you said. Sometimes its longer, sometimes even shorter.
I recorded a video.
23102017_1219_56.avi

The first skips are from the f2k controlbutton 'next' button (with activated component foo_preview).
Then from the f2k controlbutton 'next' button with deactivated component foo_preview.
Then I used the buttons from DBC.

You see the stub image very good 8-)
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When I'm skipping a song (from within the default f2k control button: next) the stub image is shown for about 500ms.
So it takes over 500ms to skip to the next song on your setup?
I've set the delay to 200ms... I guess I just make this delay configurable to be able to handle slower setups.