IPB

Welcome Guest ( Log In | Register )

 
Reply to this topicStart new topic
New AAC encoder(falabaac), Best quality, 128kbps support 22kHz, 96kbps also support 20kHz bandwidh for 44.1kHz sample rate
blueheart73
post Jan 1 2013, 07:36
Post #1





Group: Members
Posts: 9
Joined: 22-December 12
Member No.: 105337



URL for download source code and product:
http://code.google.com/p/falab/downloads/list


This is the new aac encoder , different from FAAC, I think it is better than FAAC.
And I compare with the qtaac and neroAAC, I also think falabaac is better than both of them when bitrate >= 96kbps(now falabaac only support LC, HEv2 is developing)

quality:
128kbps hear almost lossless(support 22kHz bandwidth), and 96kbps is very good(support 20kHz bandwidth)

speed:
6 speed for your choice, default is -l 2, is fast enough, and use -l 4 (faster than current all aac encoder)

other:
open source code, GNU license

download them from:
http://code.google.com/p/falab/downloads/list

please download the latest version: falabaac-v2.0.0.178.tar.gz

Need all of you test, and if have any questions and bugs please email to me : luolongzhi@gmail.com
Go to the top of the page
+Quote Post
alter4
post Jan 1 2013, 14:14
Post #2





Group: Members
Posts: 105
Joined: 14-September 04
From: Belarus, Vitebsk
Member No.: 16992



QUOTE (blueheart73 @ Jan 1 2013, 09:36) *
...I also think falabaac is better than both of them when bitrate >= 96kbps(now falabaac only support LC, HEv2 is developing)


Daring statement! I'll try to ABX some samples at 128kbps.
Go to the top of the page
+Quote Post
greynol
post Jan 1 2013, 17:41
Post #3





Group: Super Moderator
Posts: 10000
Joined: 1-April 04
From: San Francisco
Member No.: 13167



...a statement that should not have been made without DBT data.

That the OP is specifying frequency response causes me to wonder if he knows how to assess quality properly.

This post has been edited by greynol: Jan 1 2013, 17:45


--------------------
Your eyes cannot hear.
Go to the top of the page
+Quote Post
lvqcl
post Jan 1 2013, 18:45
Post #4





Group: Developer
Posts: 3214
Joined: 2-December 07
Member No.: 49183



I compiled the encoder with MSVC2010, but with suspicious warnings:

CODE
falabaac\libfalabaac\fa_aacquant.c(1234): warning C4700: uninitialized local variable 'gr' used
falabaac\libfalabaac\fa_tns.c(215): warning C4700: uninitialized local variable 'k' used
falabaac\libfalabaac\fa_mdctquant.c(907): warning C4700: uninitialized local variable 'enrgs' used
falabaac\libfalabaac\fa_mdctquant.c(908): warning C4700: uninitialized local variable 'maxs' used
falabaac\libfalabaac\fa_mdctquant.c(910): warning C4700: uninitialized local variable 'enrgd' used
falabaac\libfalabaac\fa_mdctquant.c(911): warning C4700: uninitialized local variable 'maxd' used
falabaac\libfalabaac\fa_mdctquant.c(913): warning C4700: uninitialized local variable 'enrgl' used
falabaac\libfalabaac\fa_mdctquant.c(914): warning C4700: uninitialized local variable 'enrgr' used
falabaac\libfalabaac\fa_mdctquant.c(916): warning C4700: uninitialized local variable 'maxl' used
falabaac\libfalabaac\fa_mdctquant.c(917): warning C4700: uninitialized local variable 'maxr' used
Go to the top of the page
+Quote Post
db1989
post Jan 1 2013, 19:19
Post #5





Group: Super Moderator
Posts: 5159
Joined: 23-June 06
Member No.: 32180



QUOTE (greynol @ Jan 1 2013, 16:41) *
...a statement that should not have been made without DBT data.

That the OP is specifying frequency response causes me to wonder if he knows how to assess quality properly.
Seriously this. Has lossy encoding reached a point where it’s not absurd to (attempt to) encode 22.05 kHz at a mere 128 kbps? I doubt it.

Edit to clarify, just in case: I was referring to a sampling rate of 44.1 kHz with no lowpass, i.e. 22.05 kHz as the maximal frequency (well, just over it, technically), not the sampling rate.

This post has been edited by db1989: Jan 2 2013, 00:27
Go to the top of the page
+Quote Post
sluggy
post Jan 1 2013, 20:12
Post #6





Group: Members
Posts: 54
Joined: 22-June 12
Member No.: 100900



Has anyone managed to test this out yet?
Go to the top of the page
+Quote Post
IgorC
post Jan 1 2013, 21:03
Post #7





Group: Members
Posts: 1506
Joined: 3-January 05
From: Argentina, Bs As
Member No.: 18803



To begin with, as we all know that the human upper limit of hearing is 20 kHz.
One could argue that some 7 years old kids can hear 21 kHz pure tone but they rather won't appreciate or even notice that in real case wink.gif. Anyway it's entirely another discussion.


An excellent and very mature LC-AAC encoders as Apple and FhG can preserve approx. 15.7-17kHz at 96-128 kbps respectively. Given that AAC (LC profile) format is mature and actually old it's very unlikely (impossible) to get any better numbers.
That's all staying on topic.

Now it's a whole different story if a format can use bandwidth extension. For example HE-AAC can achieve 20 kHz at 64 kbps. Of course, SBR wasn't design to be transparent but since a new bandwidth extensions have appeared (if somebody is interested google for a literature).

Opus has a high grade of transparency at 128 kbps (VBR) and use a bandwidth extension which preserves 20 kHz bandwidth. Both Opus and LC-AAC are on par at 128 kbps (VBR) and it would be difficult to see which approach is better (AAC's lowpass at 17 kHz or Opus high quality bandwidth extension up to 20 kHz).

Speaking of my experience, I can ABX 16 kHz tone and still can do it with 17 kHz but I don't hear the last one rather feel it. 18kHz isn't perceived by me in any way.

So when I hear a lossy file with enough high bitrate and 16 kHz bandwidth I can't spot any artifacts but feel that something is different. It's still ABXable but scores are very high like 4.8-4.9 of max 5.0.
It's ABXable because of absence of energy from high frequencies as I don't actually enable to hear a details in that range.
That's where bandwidth extension makes a good job. So my humble guess is that a good bandwidth extension can be superior to a classic lowpass at transparent rates as it can preserve an energy at extremely low bitrate cost.

This post has been edited by IgorC: Jan 1 2013, 21:11
Go to the top of the page
+Quote Post
AiZ
post Jan 1 2013, 23:50
Post #8





Group: Members
Posts: 37
Joined: 4-February 02
Member No.: 1251



Hello everybody and happy new year,

QUOTE (sluggy @ Jan 1 2013, 21:12) *
Has anyone managed to test this out yet?


Have just tried the Windows binary provided in falabaac-v2.0.0.178.tar.gz, I can say it is not that transparent at 128kbit/s :

CODE
foo_abx 1.3.4 report
foobar2000 v1.1.18
2013/01/01 23:39:46

File A: C:\Program Files (x86)\qaac\cmm.flac
File B: D:\temp\falabaac\cmm_128.aac

23:39:46 : Test started.
23:41:23 : 01/01  50.0%
23:41:31 : 02/02  25.0%
23:41:40 : 03/03  12.5%
23:41:47 : 04/04  6.3%
23:41:55 : 05/05  3.1%
23:42:04 : 06/06  1.6%
23:42:11 : 07/07  0.8%
23:42:19 : 08/08  0.4%
23:42:25 : 09/09  0.2%
23:42:31 : 10/10  0.1%
23:42:35 : Test finished.

----------
Total: 10/10 (0.1%)


And I've not taken the most difficult song to encode (Carly Rae Jepsen - Call me maybe)...

Bye,


AiZ


--------------------
AiZ stupid homepage - http://aiz.free.fr
Go to the top of the page
+Quote Post
blueheart73
post Jan 3 2013, 12:15
Post #9





Group: Members
Posts: 9
Joined: 22-December 12
Member No.: 105337



Thanks for all of your advice and attentions!!
The purpose of this encoder is to supply a totally free software which has good encoding quality of AAC, maybe the quality test is not enough now because of the short time development.
In my experiments, I use the sqam which supply from the EBU test audio files and some music samples I have. I optimize the encoder serverly times(espically the quantize method, the key factor which influence the sound quality, I use 3 method and found that the last one is good and I satisfy with it, but maybe is also not good enough).
So , I need you guys to help me to test the encoder problems and bugs, and tell me what the problems is ( eg: frequency response problem, or for same special sample performance is worse), I hope you can send me the problems and the problem audio sample is better!

If my opinions is not right , pardon me and help me, I think this encoder will be bettern if I find what the problem with it!!
And I really want to know the difference compare with the qtaac and neroaac, I hope you can help me to do the test experiment and give me the results(I want to know where is the faults or problems, and I can fix it and optimize it )

Thanks for your attentions and happy new year!!
Go to the top of the page
+Quote Post
LithosZA
post Jan 3 2013, 13:08
Post #10





Group: Members
Posts: 181
Joined: 26-February 11
Member No.: 88525



QUOTE
I can say it is not that transparent at 128kbit/s...


FAAC also isn't transparent at 128Kbit/s. How does falabaac compare to FAAC?
Go to the top of the page
+Quote Post
greynol
post Jan 3 2013, 16:46
Post #11





Group: Super Moderator
Posts: 10000
Joined: 1-April 04
From: San Francisco
Member No.: 13167



http://www.hydrogenaudio.org/forums/index....showtopic=16295


--------------------
Your eyes cannot hear.
Go to the top of the page
+Quote Post
lvqcl
post Jan 3 2013, 19:36
Post #12





Group: Developer
Posts: 3214
Joined: 2-December 07
Member No.: 49183



castanets2 sample encoded with falabaac and faac @ 128 kbps (encoding options: only -b 128)

[attachment=7277:castanet...falabaac.aac][attachment=7276:castanets2_faac.aac]

This post has been edited by lvqcl: Jan 3 2013, 19:41
Go to the top of the page
+Quote Post
LithosZA
post Jan 3 2013, 19:51
Post #13





Group: Members
Posts: 181
Joined: 26-February 11
Member No.: 88525



QUOTE
castanets2 sample encoded with falabaac and faac:

Attached File castanets2_falabaac.aac ( 111.53K ) Number of downloads: 0
Attached File castanets2_faac.aac ( 111.28K ) Number of downloads: 0


Ok, falabaac lost with that sample.. I wonder if decreasing the bandwidth to the same bandwidth that FAAC uses would help?
Edit: Where can I find the castanets2 sample?

This post has been edited by LithosZA: Jan 3 2013, 19:52
Go to the top of the page
+Quote Post
lvqcl
post Jan 3 2013, 19:59
Post #14





Group: Developer
Posts: 3214
Joined: 2-December 07
Member No.: 49183



Currently it can be downloaded here:

http://web.archive.org/web/20030618115117/...s/test_samples/
Go to the top of the page
+Quote Post
LithosZA
post Jan 3 2013, 20:09
Post #15





Group: Members
Posts: 181
Joined: 26-February 11
Member No.: 88525



QUOTE


Thanks, I quickly tried it out with 16Khz bandwidth and it didn't make a difference.
Go to the top of the page
+Quote Post
blueheart73
post Jan 4 2013, 03:22
Post #16





Group: Members
Posts: 9
Joined: 22-December 12
Member No.: 105337



QUOTE (LithosZA @ Jan 4 2013, 02:51) *
QUOTE
castanets2 sample encoded with falabaac and faac:

Attached File castanets2_falabaac.aac ( 111.53K ) Number of downloads: 0
Attached File castanets2_faac.aac ( 111.28K ) Number of downloads: 0


Ok, falabaac lost with that sample.. I wonder if decreasing the bandwidth to the same bandwidth that FAAC uses would help?
Edit: Where can I find the castanets2 sample?



yes, I tried this sample before and found that there will be pre-echo affect because using LONG_WINDOW. if you use option "-w 16" or use high bitrate to encode, the high frequency will be filterd, and maybe sound better.
And , to solve this quilkly change audio, you can use option "-t 1" (means time resolution first, I will use ONLY_SHORT_WINDOW to process this sample), and sound good!

But the seriously problem is that I didn't found the good method to switch the LONG_WINDOW and SHORT_WINDOW!!! the method in the ISO I tried , but it does;t work well!!
Can somebody teach me a good method(If can send a paper detail about it I will apprecaite for you)
Go to the top of the page
+Quote Post
blueheart73
post Jan 4 2013, 03:29
Post #17





Group: Members
Posts: 9
Joined: 22-December 12
Member No.: 105337



One more I want to say is that, use "-t 1" option maybe have some problem or bugs in the low bitrate, the quantizer is not good, so this option is not recommed except for the sample only quickly change
And I tried FAAC in 96kbps, the FAAC limited the bandwidth to 10kHz. Besides, I analyzed the FAAC code and found that it always use SHORT_WINDOW to process the audio(no LONG_WINDOW, it canceled in some code),
So I am thinking if it faced the same problem with me ? I am not sure about that.

Can someone familiar with the AAC tell me how to resolve this problem and how to do quickly and stably window-switching method?
Go to the top of the page
+Quote Post
jrmuizel
post Jan 4 2013, 05:54
Post #18





Group: Members
Posts: 1
Joined: 4-January 13
Member No.: 105573



QUOTE (blueheart73 @ Jan 4 2013, 03:29) *
Can someone familiar with the AAC tell me how to resolve this problem and how to do quickly and stably window-switching method?

Maybe look here:
https://github.com/mstorsjo/fdk-aac/blob/ma...lock_switch.cpp

or here:
Robust Block Switching Decision for Transform-based Audio Coder
http://citeseerx.ist.psu.edu/viewdoc/downl...p1&type=pdf
Go to the top of the page
+Quote Post
blueheart73
post Jan 4 2013, 06:00
Post #19





Group: Members
Posts: 9
Joined: 22-December 12
Member No.: 105337



QUOTE (jrmuizel @ Jan 4 2013, 12:54) *
QUOTE (blueheart73 @ Jan 4 2013, 03:29) *
Can someone familiar with the AAC tell me how to resolve this problem and how to do quickly and stably window-switching method?

Maybe look here:
https://github.com/mstorsjo/fdk-aac/blob/ma...lock_switch.cpp

or here:
Robust Block Switching Decision for Transform-based Audio Coder
http://citeseerx.ist.psu.edu/viewdoc/downl...p1&type=pdf


thank you, I will try it in the next version and see the results!
Go to the top of the page
+Quote Post

Reply to this topicStart new topic
1 User(s) are reading this topic (1 Guests and 0 Anonymous Users)
0 Members:

 



RSS Lo-Fi Version Time is now: 21st April 2014 - 14:20