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falabaac-v2.1.0.226 release, better than last old version, new aac encoder, totally free software, support linux and windows
blueheart73
post Mar 3 2013, 15:08
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URL for download source code and product:
http://code.google.com/p/falab/downloads/list


Fix many problems and bugs compare to v2.0.0.178 such as:
1. block switch problem
2. short block grouping problem
3. pre-echo control
4. quantize protected thrshold problem
...

in addition, add vbr mode and quality control, now default is -q 0.7(bandwidth is 17kHz), using -q 1 will extend to 22kHz
default setting is using vbr mode(strongly recommend, can compress according to frame feature, large bits for complicate frame and little bits for simple frame)

maybe still have some problems for different audio samples, and need your testing feedback, and I will fix it

I will continue optimize the encoder such as pre-echo control and attack precise detect problem

if you have any good encoding paper link can supply to me I will appreciate , such as aac pre-echo control (this is the first level problem I want to optimize)
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o-l-a-v
post Mar 3 2013, 21:47
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From: Norway
Member No.: 101545



Tried windows version. It has GREAT artifacts at -q 1.0, ~ 180-190kbps. Source was a lossless hardstyle tune.
Foobar2000 can't read time values, and it's therefore not possible to scroll. In vlc its possible to scroll, but the total time value changes constantly while playing the song.
Your encoder has a long way to go dude.

And whats up with compressions of the files? Every folder and subfolders is compressed individually.

Here is 15 first seconds of the song. Flac is original, aac is converted one. Used mp3directcut, so no loss in quality.
http://www.datafilehost.com/download-aaa434bf.html

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blueheart73
post Mar 4 2013, 14:44
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QUOTE (o-l-a-v @ Mar 4 2013, 04:47) *
Tried windows version. It has GREAT artifacts at -q 1.0, ~ 180-190kbps. Source was a lossless hardstyle tune.
Foobar2000 can't read time values, and it's therefore not possible to scroll. In vlc its possible to scroll, but the total time value changes constantly while playing the song.
Your encoder has a long way to go dude.

And whats up with compressions of the files? Every folder and subfolders is compressed individually.

Here is 15 first seconds of the song. Flac is original, aac is converted one. Used mp3directcut, so no loss in quality.
http://www.datafilehost.com/download-aaa434bf.html


thanks for your testing, yes, the audible artifacts is caused by the pre-echo and my switch-window(LONG_STOP_WINDOW and LONG_START_WINDOW) problem, I fix the block-switch and detect the attack, but after attack, the psychmodel masking threshold is not right for switch.

I also have some thoughts to fix this , and I will release a fix version in my project websites as soon as possible.

if anyone find some hardstyle audio sample has problem, pls send me the file link and I will testing and fix it , thank you all
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