I hope to obtain good results with all filters OFF and with the psy fine tuning (interchannel masking ratio and sticking around with the psy masking values). The default settings output something crappy and the differences are obvious for any real Hi-FI audio system.
Default 320kbps audio, of almost any format, is generally transparent. If you don't already realize this, you should stop what you are doing and try it out.
Now I have to prove there are improvements based on the psy fine tuning and concludent ABX tests.
Maybe this is a language barrier, but logically to improve something you must first find a problem. If you don't find any problems, then how are you expecting to improve anything? You are not making a lot of sense.
Last post by Sandrine -
While musicbrainz is helpful as a tagger in many situations it also introduces its very own set of challenges. The biggest one for me are their cosmetic rules. Musicbrainz insists on using special quotes, hyphens and other diacritics. This leads to incompatibilities for me when using other software, especially in conjunction with renaming via file operations when the non-ASCII character is passed onto the file name.
I would like to request a way inside of foobar2000 to normalize diacritics in tags, that is, replace non-ascii characters with an ASCII character. Obviously that is not desirable in all situations so may this could be done via a user-defined table of cases. I think someone like that is already happening "under the hood", to liken several characters to each other.
At the very least, I would hope for a user-option that would turn on or off using non-ASCII characters in file names. Thanks in advance!
Last post by John Carter -
I'm helping a bunch who have a couple of Ye Olde Analogue radio repeaters scattered about the mountains to help hikers and hunters and search and rescue.
They have a "keep an archive and listen either live or to archived recordings via the web" service. Very useful if a hiker in distress said something nearly inaudible and you want to go back and confirm you heard the right thing.
I think they are using the wrong audio codec for this and I'm looking around for something better....
* The audio quality is Bad Bad Bad by modern standards, mostly due to range, low power hand held transmitters, intervening dirty great mountains, weather, and RF interferers (eg. some farmers rusty electric motor). * The dynamic range is low. * There is substantial "crackle" interferers. * There is substantial "white noise" from RF and various components.
They have broadband connections from the repeater stations to the cloud servers. So bandwidth isn't the limiting issue, audio quality is, and latency is. (Ever tried to have a conversation where the audio lags terribly?)
Part of me says "Go lossless", but the utility might be increased by a codec that is good at emphasizing speech and losing crackle and white noise.
Which open source / patent unencumbered codec would you recommend?
At the bitrates you are using it won't be possible to have an improvement generally, so that won't work. Instead, you start with a sample where you can reliably detect a difference and then make adjustments until you cannot. Then you compare the encode you make to the original and try to show that it is now transparent.
I hope to obtain good results with all filters OFF and with the psy fine tuning (interchannel masking ratio and sticking around with the psy masking values). The default settings output something crappy and the differences are obvious for any real Hi-FI audio system. The increment of the psy masking values is +/-0.25 dB, added/subtracted to the default psy values. The sfb21 is treble dependent and, along with the low pass filter, cuts the high frequency sounds brutally. Working in progress.
At the bitrates you are using it is very unlikely you'll find many files with a difference that is detectable. Do you actually have one?
With the settings I used in lame, I saw in spectrograms some kind of denoise, but the music sounds apparently identical, from the subwoofer bumps to cymbals. I tested >30 samples. Now I have to prove there are improvements based on the psy fine tuning and concludent ABX tests.
You want to look at the celt/celt_encoder.c file. The transient_analysis() function returns 1 when the current frame is a transient and 0 otherwise. It also computes an estimate how how "strong" the transient is, which it returns in tf_estimate. That value gets used in compute_vbr() to boost the bitrate. To change the behaviour, you'd have to change the value of that estimate, and then update tf_calibration so that the average bitrate doesn't change.
I'm not a programmer so I'm a little stuck on where to make the change to the value in question (lets say I want to double it, for simplicity). I have celt_encoder.c open and there are 22 mentions of "tf_estimate" in the code (including in the comments, so I know to ignore those).
If it's not too much trouble, could you point me to the lines and the numbers I need to change? I'm looking at line 400 as potentially the one I need to edit for "tf_estimate" (maybe line 2,090 also), and line 1345 for "tf_calibration". But I'm just guessing.
Anyone else who is familiar with the OPUS source code is free to chip in.
2) Second suggestion--check your ears. This is a common insult among the audiophool crowd, but the reality is that anyone my age (mid-50s) that thinks their ears are still golden is either delusional or exceptional. Exceptionally delusional, that is. Between military service and too much Van Halen, my ears are a bit ragged and roll off over 8kHz. At 10kHz and up I can't hear anything at any volume below painful. However, while that isn't a really a problem, certain midrange frequencies will sound like broken glass in a garbage disposal if they go over a certain volume. Your statement that you avoid loud volumes because your ears are sensitive leads me to think you may have a similar issue. So you can see what it is that your ears are sensitive to by running the tone sweep tests at the links I provided. Run them at a volume somewhat higher than your normal listening level to find your upper hearing limit and see if any points sound exceptionally harsh. If you find a point that irritates your ears, recruit a test subject to listen and see if they hear the same harshness.
Thank you so much for your input and a special thank you for those links. I actually never conducted tests like these so it was all very useful. I learned that I pass the mosquito test (17.4 kHz) at my regular listening volume level and can still hear 18 kHz but it's a lot quieter. And I cannot hear 19 kHz and above. 17 kHz was somewhat hurtful to my ears whereas 18 kHz did not. I guess that's where my sensitivity really rolls off - at 18 kHz...
P.S. I will try implementing your other suggestions a little later.
Last post by realsmart987 -
I'm sorry if this was already posted. I can't find it on Google.
I ripped a CD and it shows as finished in EAC, but if I go to the Music folder through the Windows Start Menu it isn't there. I tried creating the folder manually but Windows stopped me because it said there was already a folder with the same name. Some other post said it might be hidden so I enabled "show hidden folders" in File Explorer. Still nothing.
The EAC version I used is "V1.3 From 2. September 2016" My Windows 10 build number (if it matters) is build 17134.
Maybe it's because the album is from 1998 and something changed since then. If there isn't a reply by the time I get home from work I'll try ripping a newer CD and post the results here. I know the problem is at least not DRM-related.