Aften 0.0.8 is out

[raquete]

oh, i don't knew about aften -longhelp option.
EncWAVtoAC3 have one kind of "online help" when you let the mouse over the option but dissapear fast and you have to take the mouse there again...between this blinkings moments when the help appear and dissapear, me, with my magnific english forgot what i read and need to restart the lecture thousands times...lol
better is print the -longhelp!
i will do one short list(really short) with specifics questions.
thanks tebasuna.

[wisodev]

wisodev
when and if you have time, can you do a "how to" explaining each parameter of advanced mode/options and advanced options in EncWAVtoAC3?
(for boreds users like me :p)
thanks.

Tip: Each advanced option has a tool tip if you mouse over it in program window, even each item in adv. options list has tool tip based on Aften -longhelp text.

Edit: Ups I was to slow :)

[OAKside]

OAKside, what i upload for you was EncWAVtoAC3-4.1 with Aften R832 and works perfectly, right? (or not? )
now is ok as wisodev reupload "everything" but i'm curious about what i send for you. Thanks raquete. EncWAVtoAC3 does not include the various Aften EXEs (only DLLs), which I wanted to test in a few other apps.
(AVStoDVD, MeGUI, etc.) I didn't want to bug you about it again. :) I appreciate the help and response, guys.

I noticed Aften R843, happily tried out the EXE (MeGUI) and that's when I noticed the weird/slow behavior compared to R762.
So I was looking for a slightly older build. (It's worth mentioning that I haven't reproduced weird R843 behavior again, though. :confused:)

[raquete]

all that i have is here, if you're interested in any or in all, i upload for you:
dates here appear as day/month/year

aften_20060709(09/07/2006), aften_parallel(22/11/2006), aften-0.0.8.tar(19/07/2008), aften-0.01(11/07/2006), aften-0.02.tar(11/07/2006), aften-0.02(15/07/2006), aften-0.03(20/07/2006), aften-0.03-dev(23/04/2006), aften-0.03-dev new(30/07/2006), aften-0.03-dev(05/08/2006), aften-0.04(05/08/2006), Aften-0.04_rev13(06/08/2006), Aften-0.04_rev27(09/08/2006), aften-0.04kurtniose(06/08/2006), aften-0.04-win32-i586(05/08/2006), aften-0.04-win32-i686(05/08/2006, aften-0.05(09/01/2007), Aften-0.05_rev185(23/10/2006), Aften-0.05_rev211(28/10/2006), aften-0.05_rev214(30/10/2006), aften-0.05-win32, aften-current.tar(20/07/2006), aften-daily-080606.tar(06/08/2006), AftenGui 1.0,1.1,1.2,1.3,1.4.
another folder(maybe can repeat some versions): aften_rev382(18/02/2007), aften_rev467(18/03/2007), aften-0.03(09/01/2007), aften-0.04(09/01/2007), aften-0.04-win32-i686(Justin)(10/01/2007), aften-0.05_rev216(09/01/2007), aften-0.05-win32(Justin)(10/01/2007), aften-svn-r216-win32-bin(Wisodev)(10/01/2007), aften-svn-r224-bin(Wisodev)(10/01/2007), aften-svn-r224-bin(09/01/2007), aften-svn-r433-win32-bin(27/02/2007).

EncWAVtoAC3: seems that have all versions.(16 versions)

[Paulcat]

Due to a strange failure in my BeSweet/BeLight combo, the wav file I was trying to encode to an ac3 file ended up with no sound. I needed a quick fix and went to FFMpegGUI (0.3) and to WavToAC3Enc (4.1). I tested both with some small wav files (Windows Startup Sound!) and only the FFMpeg version of the ac3 file would play (Using AC3 Filter 1.65, the latest one, and Media Player Classic 6.4.9.1).

Since I was pressed for time, I had to use FFMpeg to do my main file and build my dvd, but it is quite old (adding newer versions of FFMpeg.exe to the directory caused the same error in playing the WavToAC3Enc converted files, namely the "Cannot render the file" error from MPC)

BeSweet/BeLight always worked for me before, I don't know if there was some strange error in the wav file or whether the files on my HD got corrupted, but I still need a way to do the conversion.

I'm assuming that WavToAC3Enc is producing genuine ac3 files, what can I use to play them on my pc? I don't want to go through the trouble to create a physical dvd and find out after that it won't play correctly in my set top!

Paul

[raquete]

play your file with winamp(using AC3 Filter, can be found in winamp or AC3 filter homepages)

[Paulcat]

So this is just an issue with MPC? I've never had an issue before...

[tebasuna51]

... I needed a quick fix and went to FFMpegGUI (0.3) and to WavToAC3Enc (4.1). I tested both with some small wav files (Windows Startup Sound!) and only the FFMpeg version of the ac3 file would play ...
Please upload your converted ac3 files.

[Vincent Vega]

hi everyone, i just used megui .1051 (aften r832 sse, nicaudio 2009-06-25, all the other updates too) to convert 5.1 dts to 5.1 ac3 (default aften settings, except bitrate 448).

as i understand the theoretical math for my audio stream should be:
dts: 763872 frames x 512 samples/frame = 391102464 (2:15:47)
to
ac3: 254624 frames x 1536 samples/frame = 391102464 (2:15:47)

now valex's ac3file filter property page shows one extra frame in the encoded ac3 stream (254625) and duration 2:15:48

is this normal or not? heres ac3file screenshots in attach:

[tebasuna51]

Each time you encode to ac3 (commercial encoders also) a initial silence of 256 samples is padded before the source.
For a 48 KHz samplerate audio means a delay of 5.333 ms and the output is 5.333 ms long needing a frame more.

You can avoid this delay usin the Aften parameter -pad 0

[Vincent Vega]

thank you very much tebasuna51 for the explanation. then i have one more question: now to do the best to sync this ac3back to video i'd need a negative audio delay of 6ms and mkvmerge does allow delays of multiples of 2ms. at the same time from what i remember audio delays can only work if they are multiples of frame size (32ms in this case), so telling mmg to use -6ms delay will only result in some unpredictable behavior right?

[tebasuna51]

Forget the delay or use BeHappy instead MeGUI to put the -pad 0 parameter

[Terrachild]

I'm new to Aften, I'm using "WAV to AC3 Encoder 4.1" as the GUI, and boy are there a lot of settings.

I'm making 5.1 surround files.

If I want to get the best sound possible, what should I be changing. I don't care how slow the encoder is, just best quality.

Thanks.

[tebasuna51]

Use 640 Kb/s, -readtoeof 1 (for big files) and defaults.

[Vincent Vega]

hi tebasuna51, -pad 0 and behappy did the trick, thanx a lot.

perhaps you could kindly help me figure out one more thing: i converted this 5.1 dts to 5.1 ac3, i used Norm. to 100% and encoder defaults. now im playing these files using ac3filter and downmixed to stereo (old ac97 alc101 on this machine ).

with exactly the same ac3filter settings and same mixer matrix, dts sounds significantly louder, why this loudness difference? actually i've had the same issue with all my previous dts->ac3 megui encodes.

[Abradoks]

Objective comparison of AC3 encoders

Source: DTS 1510 kbit/s 48kHz 6ch (Pirates of the Caribbean: At World's End)
Decoding into WAV (PCM 24 bit) — eac3to (libav).
SNR (http://en.wikipedia.org/wiki/Signal-to-noise_ratio) measurement (bigger is better) and padding detection — CompAudio (http://www-mmsp.ece.mcgill.ca/Documents/Software/Packages/AFsp/CompAudio.html)
PEAQ (http://en.wikipedia.org/wiki/PEAQ) measurement (bigger is better) — PQevalAudio (http://www-mmsp.ece.mcgill.ca/Documents/Software/Packages/AFsp/PQevalAudio.html)
Padding — delaycut
WAV splitting — BeSplit
Spectrograms — Sound Forge

aften r832 (from MeGUI)
Dolby Digital Encoder 6.2.2 (Sonic Foundry Soft Encode 1.0.19): output.DD6448.ac3 (448 kbps), output.DD6224.ac3 (224 kbps)
Dolby Digital Encoder 7.0 (Sony Vegas Pro 9.0.563): output.DD7448.ac3 (448 kbps), output.DD7224.ac3 (224 kbps)

Difference between aften_wide and aften_spec:
- aften (wide) encodes full frequency range: -w 60
- aften (spec) encodes same frequency range as DD encoders:
448 kbps -> 20.34 kHz -> w=48
224 kbps -> 9.05 kHz -> w=8

Preprocessing in Dolby Encoder was disabled:
http://img41.imageshack.us/img41/3071/sfse.th.png (http://img41.imageshack.us/i/sfse.png/)

Comparison:
Name | SNR | PEAQ

448 kbps, ch L

DD7 | 26.297| -0.226
DD6 | 26.268| -0.222
aften | 20.653| -0.478
aften (spec) | 18.504| -0.288
aften (wide) | 19.368| -1.539

Aften gets less points than proprietary Dolby Digital Encoder (here referred to as "DD") both in SNR and PEAQ.
DD7 has slightly better results than DD6.

Average frequency graphs:
http://img10.imageshack.us/img10/36/freq.th.png (http://img10.imageshack.us/i/freq.png/)
As you can see, aften has bigger frequency cutoff by default than DD.
With "-w 48" (aften_spec) it matches DD and more than that, it saves high frequency average energy closer to original.
But in all three cases aften has higher energy on low frequencies then original track.

DD6 vs DD7:
http://img30.imageshack.us/img30/4406/freqdd6vsdd7.th.png (http://img30.imageshack.us/i/freqdd6vsdd7.png/)

Time-frequency spectrograms
Original: aften_wide: aften_spec: DD7: DD6:
http://img11.imageshack.us/img11/7586/specinput.th.png (http://img11.imageshack.us/i/specinput.png/)http://img15.imageshack.us/img15/9016/specaftenwide.th.png (http://img15.imageshack.us/i/specaftenwide.png/)http://img15.imageshack.us/img15/7751/specaftenspec.th.png (http://img15.imageshack.us/i/specaftenspec.png/)http://img36.imageshack.us/img36/1278/specdd7.th.png (http://img36.imageshack.us/i/specdd7.png/)http://img12.imageshack.us/img12/9971/specdd6.th.png (http://img12.imageshack.us/i/specdd6.png/)


448 kbps, ch C

DD7 | 26.039| -0.211
DD6 | 26.003| -0.208
aften | 20.349| -0.379
aften (spec) | 18.274| -0.263
aften (wide) | 19.108| -0.975

Average frequency graphs:
http://img15.imageshack.us/img15/9238/freqc.th.png (http://img15.imageshack.us/i/freqc.png/)

Time-frequency spectrograms
Original: aften_spec: DD7:
http://img41.imageshack.us/img41/514/specinputc.th.png (http://img41.imageshack.us/i/specinputc.png/)http://img30.imageshack.us/img30/7941/specaftenspecc.th.png (http://img30.imageshack.us/i/specaftenspecc.png/)http://img30.imageshack.us/img30/2899/specdd7c.th.png (http://img30.imageshack.us/i/specdd7c.png/)


224 kbps, ch L

DD7 | 21.187| -2.266
DD6 | 20.845| -2.275
aften | 15.009| -2.584
aften (spec) | 14.830| -2.596
aften (wide) | 6.6604| -3.527

Average frequency graphs:
http://img12.imageshack.us/img12/2221/freq224.th.png (http://img12.imageshack.us/i/freq224.png/)


224 kbps, ch C

DD7 | 19.303| -2.227
DD6 | 19.213| -2.239
aften | 14.542| -2.601
aften (spec) | 14.378| -2.583
aften (wide) | 6.5879| -3.139


448 kbps, all channels (SNR only)

aften L | 20.653 || DD7 L | 26.297
aften R | 20.525 || DD7 R | 26.490
aften C | 20.349 || DD7 C | 26.039
aften SL | 21.137 || DD7 SL | 25.819
aften SR | 21.003 || DD7 SR | 25.541
aften LFE | 22.176 || DD7 LFE | -0.039484 || DD6 LFE | 29.818

When running tests on all channels this strange problem with LFE was discovered.
I have rechecked everything several times, but DD7 was still getting bad results on LFE.
As you see DD6 hasn't this problem. More tests required to check this out.

Average frequency graphs (LFE):
http://img15.imageshack.us/img15/4736/freqlfe.th.png (http://img15.imageshack.us/i/freqlfe.png/)

Time-frequency spectrograms (LFE)
Original: aften_spec: DD7:
http://img11.imageshack.us/img11/3403/speclfeinput.th.png (http://img11.imageshack.us/i/speclfeinput.png/)http://img8.imageshack.us/img8/4563/speclfeaften.th.png (http://img8.imageshack.us/i/speclfeaften.png/)http://img30.imageshack.us/img30/2580/speclfedd7.th.png (http://img30.imageshack.us/i/speclfedd7.png/)


Commandline:
aften.exe -b 448 -pad 0 -w 60 -s on -acmod 7 -lfe 1 -readtoeof 1 input.wav output.aften448_wide.ac3
aften.exe -b 448 -pad 0 -w 48 -s on -acmod 7 -lfe 1 -readtoeof 1 input.wav output.aften448_spec.ac3
aften.exe -b 448 -pad 0 -acmod 7 -lfe 1 -readtoeof 1 input.wav output.aften448.ac3
aften.exe -b 224 -pad 0 -w 60 -s on -acmod 7 -lfe 1 -readtoeof 1 input.wav output.aften224_wide.ac3
aften.exe -b 224 -pad 0 -w 8 -s on -acmod 7 -lfe 1 -readtoeof 1 input.wav output.aften224_spec.ac3
aften.exe -b 224 -pad 0 -acmod 7 -lfe 1 -readtoeof 1 input.wav output.aften224.ac3

C:\comparison\eac3to\eac3to.exe output.aften448_wide.ac3 output.aften448_wide.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.aften448_spec.ac3 output.aften448_spec.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.aften448.ac3 output.aften448.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.aften224_wide.ac3 output.aften224_wide.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.aften224_spec.ac3 output.aften224_spec.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.aften224.ac3 output.aften224.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.DD6448.ac3 output.DD6448.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.DD6224.ac3 output.DD6224.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.DD7448.ac3 output.DD7448.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav
C:\comparison\eac3to\eac3to.exe output.DD7224.ac3 output.DD7224.wavs
del output.*.R.wav output.*.LFE.wav output.*.SL.wav output.*.SR.wav

delaycut.exe -start 5.33 -out input.L.pad256.wav input.L.wav
delaycut.exe -start 5.33 -out input.C.pad256.wav input.C.wav
delaycut.exe -start 13.25 -out input.L.pad636.wav input.L.wav
delaycut.exe -start 13.25 -out input.C.pad636.wav input.C.wav

C:\comparison\AFsp\CompAudio.exe input.L.wav output.aften448_wide.L.wav > output.aften448_wide.L.comp.txt
C:\comparison\AFsp\CompAudio.exe input.C.wav output.aften448_wide.C.wav > output.aften448_wide.C.comp.txt
C:\comparison\AFsp\CompAudio.exe input.L.wav output.aften448_spec.L.wav > output.aften448_spec.L.comp.txt
C:\comparison\AFsp\CompAudio.exe input.C.wav output.aften448_spec.C.wav > output.aften448_spec.C.comp.txt
C:\comparison\AFsp\CompAudio.exe input.L.wav output.aften448.L.wav > output.aften448.L.comp.txt
C:\comparison\AFsp\CompAudio.exe input.C.wav output.aften448.C.wav > output.aften448.C.comp.txt
C:\comparison\AFsp\CompAudio.exe input.L.wav output.aften224_wide.L.wav > output.aften224_wide.L.comp.txt
C:\comparison\AFsp\CompAudio.exe input.C.wav output.aften224_wide.C.wav > output.aften224_wide.C.comp.txt
C:\comparison\AFsp\CompAudio.exe input.L.wav output.aften224_spec.L.wav > output.aften224_spec.L.comp.txt
C:\comparison\AFsp\CompAudio.exe input.C.wav output.aften224_spec.C.wav > output.aften224_spec.C.comp.txt
C:\comparison\AFsp\CompAudio.exe input.L.wav output.aften224.L.wav > output.aften224.L.comp.txt
C:\comparison\AFsp\CompAudio.exe input.C.wav output.aften224.C.wav > output.aften224.C.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 256: input.L.pad256.wav output.DD6448.L.wav > output.DD6448.L.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 256: input.C.pad256.wav output.DD6448.C.wav > output.DD6448.C.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 256: input.L.pad256.wav output.DD6224.L.wav > output.DD6224.L.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 256: input.C.pad256.wav output.DD6224.C.wav > output.DD6224.C.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.L.pad636.wav output.DD7448.L.wav > output.DD7448.L.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.C.pad636.wav output.DD7448.C.wav > output.DD7448.C.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.L.pad636.wav output.DD7224.L.wav > output.DD7224.L.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.C.pad636.wav output.DD7224.C.wav > output.DD7224.C.comp.txt

C:\comparison\AFsp\PQevalAudio.exe input.L.wav output.aften448_wide.L.wav > output.aften448_wide.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.C.wav output.aften448_wide.C.wav > output.aften448_wide.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.L.wav output.aften448_spec.L.wav > output.aften448_spec.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.C.wav output.aften448_spec.C.wav > output.aften448_spec.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.L.wav output.aften448.L.wav > output.aften448.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.C.wav output.aften448.C.wav > output.aften448.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.L.wav output.aften224_wide.L.wav > output.aften224_wide.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.C.wav output.aften224_wide.C.wav > output.aften224_wide.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.L.wav output.aften224_spec.L.wav > output.aften224_spec.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.C.wav output.aften224_spec.C.wav > output.aften224_spec.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.L.wav output.aften224.L.wav > output.aften224.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe input.C.wav output.aften224.C.wav > output.aften224.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe -l 256: input.L.pad256.wav output.DD6448.L.wav > output.DD6448.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe -l 256: input.C.pad256.wav output.DD6448.C.wav > output.DD6448.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe -l 256: input.L.pad256.wav output.DD6224.L.wav > output.DD6224.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe -l 256: input.C.pad256.wav output.DD6224.C.wav > output.DD6224.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe -l 636: input.L.pad636.wav output.DD7448.L.wav > output.DD7448.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe -l 636: input.C.pad636.wav output.DD7448.C.wav > output.DD7448.C.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe -l 636: input.L.pad636.wav output.DD7224.L.wav > output.DD7224.L.PEAQ.txt
C:\comparison\AFsp\PQevalAudio.exe -l 636: input.C.pad636.wav output.DD7224.C.wav > output.DD7224.C.PEAQ.txt

Measurement SNR of different channels:
delaycut.exe -start 13.25 -out input.R.pad636.wav input.R.wav
delaycut.exe -start 13.25 -out input.SL.pad636.wav input.SL.wav
delaycut.exe -start 13.25 -out input.SR.pad636.wav input.SR.wav
delaycut.exe -start 13.25 -out input.LFE.pad636.wav input.LFE.wav
C:\comparison\eac3to\eac3to.exe output.aften448.ac3 output.aften448.all.wavs
C:\comparison\eac3to\eac3to.exe output.DD7448.ac3 output.DD7448.all.wavs
C:\comparison\AFsp\CompAudio.exe input.L.wav output.aften448.all.L.wav > output.aften448.all.L.comp.txt
C:\comparison\AFsp\CompAudio.exe input.R.wav output.aften448.all.R.wav > output.aften448.all.R.comp.txt
C:\comparison\AFsp\CompAudio.exe input.C.wav output.aften448.all.C.wav > output.aften448.all.C.comp.txt
C:\comparison\AFsp\CompAudio.exe input.SL.wav output.aften448.all.SL.wav > output.aften448.all.SL.comp.txt
C:\comparison\AFsp\CompAudio.exe input.SR.wav output.aften448.all.SR.wav > output.aften448.all.SR.comp.txt
C:\comparison\AFsp\CompAudio.exe input.LFE.wav output.aften448.all.LFE.wav > output.aften448.all.LFE.comp.txt

C:\comparison\AFsp\CompAudio.exe -l 636: input.L.pad636.wav output.DD7448.all.L.wav > output.DD7448.all.L.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.R.pad636.wav output.DD7448.all.R.wav > output.DD7448.all.R.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.C.pad636.wav output.DD7448.all.C.wav > output.DD7448.all.C.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.SL.pad636.wav output.DD7448.all.SL.wav > output.DD7448.all.SL.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.SR.pad636.wav output.DD7448.all.SR.wav > output.DD7448.all.SR.comp.txt
C:\comparison\AFsp\CompAudio.exe -l 636: input.LFE.pad636.wav output.DD7448.all.LFE.wav > output.DD7448.all.LFE.comp.txt

C:\comparison\eac3to\eac3to.exe output.DD6448.ac3 output.DD6448.all.wavs
C:\comparison\AFsp\CompAudio.exe -l 256: input.LFE.pad256.wav output.DD6448.all.LFE.wav > output.DD6448.all.LFE.comp.txt

Padding detection:
BeSplit.exe -core( -input input.L.wav -prefix input.L.cut -type wav ) -split( 0 1 )
BeSplit.exe -core( -input output.aften448.L.wav -prefix output.aften448.L.cut -type wav ) -split( 0 1 )
BeSplit.exe -core( -input output.DD6448.L.wav -prefix output.DD6448.L.cut -type wav ) -split( 0 1 )
BeSplit.exe -core( -input output.DD7448.L.wav -prefix output.DD7448.L.cut -type wav ) -split( 0 1 )
C:\comparison\AFsp\CompAudio.exe -d 0:1030 input.L.cut01.wav output.aften448.L.cut01.wav > output.aften448.L.cut01.comp.txt
C:\comparison\AFsp\CompAudio.exe -d 0:1030 input.L.cut01.wav output.DD6448.L.cut01.wav > output.DD6448.L.cut01.comp.txt
C:\comparison\AFsp\CompAudio.exe -d 0:1030 input.L.cut01.wav output.DD7448.L.cut01.wav > output.DD7448.L.cut01.comp.txt


Conclusion:
1) There is a simple method of objective audio comparison that anybody can use.
2) Aften has much worse SNR than DD. Even 224 kbps AC3 encoded with DD beats aften 448 kbps in SNR.
3) When using PEAQ metric (which takes into account some psychoacoustic effects) aften is closer to DD, but still worse.
4) If you want to have frequency cutoff similar to DD encoders, you should use w=48 for 448 kbps and w=8 for 224 kbps.
5) Aften saves high frequency average energy closer to original. But it has higher energy on low frequencies then original track.
6) Aften has better SNR on surround channels than on front. With DD it is inversely.
7) DD7 has very bad SNR on LFE channel. More tests should be done on this issue.

P.S. It was originally written in Russian, so feel free to correct spelling.

[yesgrey]

Objective comparison of AC3 encoders
Why have you used a lossy encoded track (DTS 1510kbps) as the source for the tests? I think it turns the results less meaningfull...

[honai]

Actually, the results are thoroughly invalidated by that.

Also, even though the Blu-ray of At World's End contains a lossless LPCM track, it's 16 bit and therefore not suitable for this test, either.

[jruggle]

Difference between aften_wide and aften_spec:
- aften (wide) encodes full frequency range: -w 60
- aften (spec) encodes same frequency range as DD encoders:
448 kbps -> 20.34 kHz -> w=48
224 kbps -> 9.05 kHz -> w=8

That is actually very good to know. I don't have access to a DD encoder. I would not be opposed to changing the defaults if that's what users would prefer. What are the other cutoffs for different bit rates and channel layouts?

edit: also, could you provide some small snippets of the DD-encoded files so that I can see what encoding features and params they use to get the better PEAQ and SNR values?

[Abradoks]

What are the other cutoffs for different bit rates and channel layouts?

3/2 (L, C, R, l, r)
Bitrate w (48 kHz) w (44.1 kHz) w (32 kHz)
640, 576, 512, 448 48 56 60
384 40 44 60
320 32 36 60
256 20 24 44
224 8 12 28

2/2 (L, R, l, r); 3/1 (L, C, R, l)
Bitrate w (48 kHz) w (44.1 kHz) w (32 kHz)
640, 576, 512, 448, 384, 320 48 56 60
256 32 36 60
224 24 28 48
192 12 16 32

2/1 (L, R, l); 3/0 (L, C, R)
Bitrate w (48 kHz) w (44.1 kHz) w (32 kHz)
640, 576, 512, 448, 384, 320 48 56 60
256 40 44 60
224 32 36 60
192 24 28 48
160 16 20 40
128 0 4 20

2/0 (L, R)
Bitrate w (48 kHz) w (44.1 kHz) w (32 kHz)
640, 576, 512, 448, 384, 320, 256, 224, 192 48 56 60
160 32 36 60
128 24 28 52
112 20 24 44
96 0 4 20

1/0 (C)
Bitrate w (48 kHz) w (44.1 kHz) w (32 kHz)
640, 576, 512, 448, 384, 320, 256, 224, 192, 160, 128, 112, 96 48 56 60
80 32 36 60
64 16 20 40
56 12 16 32


also, could you provide some small snippets of the DD-encoded files so that I can see what encoding features and params they use to get the better PEAQ and SNR values?
Source wavs (http://rghost.ru/594489), 224.DD6.ac3 (http://rghost.ru/594504), 448.DD6.ac3 (http://rghost.ru/594500). It's one minute (00:16:00-00:17:00) from previously mentioned source. Also, I've noticed that encoding with DD takes much more time compared to aften. 224.DD6.ac3 — 15 seconds, 448.DD6.ac3 — 24 seconds. aften -b 448 takes about 3 seconds.
jruggle, you can send me any sample you want to be encoded with DD plugin.


yesgrey3, honai, are you kidding? There is no difference what source do you take if it is complex enough for current encoding. And that 1510 kbps DTS is hard enough to encode in 448 kbps AC3, as you can see from SNR. Also I was trying to make a real life comparison, so I didn't try to encode any synthetic tracks. Nevertheless you can run test by yourself on any other source. It's not hard if you have DD encoder. I don't think that you will get different results.

[nurbs]

I don't know about audio, but if you use a lossy source with video codecs then codecs that are similar to the codec the source has been encoded with have an advantage.

[Abradoks]

I don't know about audio, but if you use a lossy source with video codecs then codecs that are similar to the codec the source has been encoded with have an advantage.
Right. But when somebody makes a comparison between 20 Mbit/s BD source and 3 Mbit/s rips made with different codecs nobody blames him (even if he uses non-DCT codec).

[yesgrey]

Abradoks,
I've sent you a PM with a link where you can get a sample of a lossless 5.1 48kHz 24bit audio file. If you could rerun the test with it it would be great...

Thanks.

[honai]

The test was biased because inherently the "source" DTS has already been processed/filtered in favor of DTS.

Also, since we know that the LPCM track of At World's End is 16 bit, there is a very high probability that the source for the DTS was also 16 bit, regardless of the DTS header information.

[Abradoks]

I've sent you a PM with a link where you can get a sample of a lossless 5.1 48kHz 24bit audio file. If you could rerun the test with it it would be great...

Name | SNR | PEAQ || SNR | PEAQ
448 kbps 224 kbps
DD6, L | 22.618| -0.205 || 16.188| -2.471
aften, L | 18.541| -0.523 || 12.844| -3.115
DD6, C | 23.644| -0.175 || 16.875| -2.203
aften, C | 18.571| -0.377 || 12.790| -3.109


The test was biased because inherently the "source" DTS has already been processed/filtered in favor of DTS.

Also, since we know that the LPCM track of At World's End is 16 bit, there is a very high probability that the source for the DTS was also 16 bit, regardless of the DTS header information.
I know. But as I've already mentioned it doesn't make a difference. Look at results measured on the 24-bit PCM provided by yesgrey3.

[raquete]

I know. But as I've already mentioned it doesn't make a difference. Look at results measured on the 24-bit PCM provided by yesgrey3.
i can't post in mumbers how much tests i did and i trust in your results but in particular, i'm interested to hear as encoding with aften for long time feel better results when comparing with softencode or with the Dolby encoder inside sound forge 9.

you and yesgrey3 could post yours samples for members apreciations if don't break forum's rules.

regards!

[yesgrey]

But as I've already mentioned it doesn't make a difference. Look at results measured on the 24-bit PCM provided by yesgrey3.
Yes, DD still has better numbers, but the difference is smaller.
I would also like to see the numbers with 640kbps...

you and yesgrey3 could post yours samples for members apreciations if don't break forum's rules.
I have considered posting the file here, but decided not to do it because I did not knew if I could be violating any rule...
I've extracted the period between minute 1:00 to 2:10 of "quantum of solace" Blu-ray.

[SquallMX]

That is actually very good to know. I don't have access to a DD encoder. I would not be opposed to changing the defaults if that's what users would prefer. What are the other cutoffs for different bit rates and channel layouts?

edit: also, could you provide some small snippets of the DD-encoded files so that I can see what encoding features and params they use to get the better PEAQ and SNR values?

Aften use by default the average frequency cut of Dolby Encoders, example:

Dolby 5.1 AC3 448 Kbps, Freq 14/20, Aften 17 Khz
Dolby 2.0 AC3 192 Kbps, Freq 10/20, Aften 15 Khz

Probably because the lack of channel/frequency coupling in Aften, if I recall correctly you need 572/256 Kbps for true 20 Khz cut off on 5.1/2.0 audio files encoded by dolby encoders, ironical you can't get more than that in an official encoder, but with aften you can go up to 24 Khz if you want to.

:helpful:

[raquete]

I have considered posting the file here, but decided not to do it because I did not knew if I could be violating any rule...
I've extracted the period between minute 1:00 to 2:10 of "quantum of solace" Blu-ray.
i posted few samples in lots of formats with round 10 seconds in the forum some time ago in differents threads and i don't break rules as the idea is too share and change informations then, i can't see any problem.

[Abradoks]

Yes, DD still has better numbers, but the difference is smaller.
It's not because of lossless source. With 1 minute sample I've previously posted I get results that differ from the whole track results in the same way:

Name | SNR | PEAQ
448 kbps
DD6, L | 23.382| -0.205
aften, L | 19.154| -0.582

And the difference is not "smaller". Look at PEAQ values.
I would also like to see the numbers with 640kbps...

Name | SNR | PEAQ
640 kbps
DD6, L | 27.809| -0.039
aften, L | 22.764| -0.172
DD6, C | 28.911| 0.003
aften, C | 22.982| -0.056

i'm interested to hear as encoding with aften for long time feel better results when comparing with softencode or with the Dolby encoder inside sound forge 9.
Here (http://rghost.ru/600115) is "Quantum of Solace" source wav, output.448.DD6.ac3 and output.224.DD6.ac3. You can get compiled aften from MeGUI server (http://megui.org/auto/aften_r832.zip). Try to compare DD encoding with default aften frequency range and with tweaked through -w option.
But you should note, that all these numbers are mostly about development. If you feel that aften "sounds better" then you should use it.

[shon3i]

Look at PEAQ values.And you start to use ears instead some numbers. AC3 use different acoustic models to cheat human perception, that mean if some number show that codec A is better than B, B still can be fully transparent to A. Same as PSNR and SSIM for Video measuring.

[yesgrey]

And the difference is not "smaller". Look at PEAQ values.
Yes, I forgot to look at those...:)


Name | SNR | PEAQ
640 kbps
DD6, L | 27.809| -0.039
aften, L | 22.764| -0.172
DD6, C | 28.911| 0.003
aften, C | 22.982| -0.056

I know that maybe it could be slightlt off-topic, but could you, by any chance, also show the numbers for DTS 1510kbps?
It would be interesting to compare the results with the AC3 640kbps, since some tests showed that AC3 640kbps is superior to DTS 1510kbps...

[raquete]

Abradoks
first i thank you to send the samples and source. :)

to encode with aften i use WAVtoAC3Enc4.1, all adjusts default(aften R832, newers versions don't work in my old win2000).

224k: sounds equal, frequences don't go ahead 9KHz.

448: WOW, your sample sounds good, is clever and go ahead 20KHz. :cool:
with aften sounds good too but don't go ahead 17KHz.

see?!? hearing we can feel differences!

...now i have requests to use it in WAVtoAC3Enc :p

cheers and thanks again.

[Abradoks]

with aften sounds good too but don't go ahead 17KHz
Please try to compare aften encodings with different frequency cutoffs. Here (http://rghost.ru/604344) are six samples: aften_wide ("-w 60" for 448 kbit/s and 224 kbit/s; full range), aften_spec ("-w 48" and "-w 8"; DD range) and aften ("-w 34" and "-w 6"; default range).

[raquete]

Abradoks, i'm horrible in command lines :mad: and i use WAVtoAC3Enc4.1(Aften) and don't know how to do the rights adjusts in this prog too.
can you write the command lines to use Aften with yours parameters posted please?

thanks.

[arestarh]

Abradoks, i'm horrible in command lines :mad: and i use WAVtoAC3Enc4.1(Aften) and don't know how to do the rights adjusts in this prog too.
can you write the command lines to use Aften with yours parameters posted please?

thanks.

Maybe this option in WAVtoAC3Enc is equel to key -w in command line mode:
http://uaimage.com/t/625155_1a63499e.jpg (http://uaimage.com/image/1a63499e)

[Abradoks]

Abradoks, i'm horrible in command lines :mad: and i use WAVtoAC3Enc4.1(Aften) and don't know how to do the rights adjusts in this prog too.
can you write the command lines to use Aften with yours parameters posted please?
It was just a description of samples I was linking to. You can just download (http://rghost.ru/604344) it, no need to encode by yourself.

[raquete]

It was just a description of samples I was linking to. You can just download (http://rghost.ru/604344) it, no need to encode by yourself.
ok, i got the files to hear and measure.

output.224.aften don't go ahead 9KHz.
output.224.aften_spec don't go ahead 9,5KHz.
output.224.aften_wide go ahead 23KHz.
output.448.aften don't go ahead 17KHz.
output.448.aften_spec go ahead 20,5KHz.
output.448.aften_wide go ahead 23KHz.

"wide" samples are very very good, 224.aften_wide and 448.aften_spec is very nice too!

i could feel better differences if was know musics but the samples are clevers.

thanks and regards!

[jruggle]

Try to compare DD encoding with default aften frequency range and with tweaked through -w option.
But you should note, that all these numbers are mostly about development. If you feel that aften "sounds better" then you should use it.
So far the only significant difference I've found between the Aften output and the DD output is channel coupling. The bandwidth reported for DD is the upper end of the coupling range. Aften does not support channel coupling yet.

The problem I've had implementing it in Aften has been the complexity of the code due to simd and multi-threading. As an experiment, I stripped all of that out and was able to get working channel coupling (although it was much slower obviously). I'm still working on trying to get it ported over to Aften.

Based on the AC-3 features used, the only other place where there could be some differences is in the exponent strategy decision. I have not analyzed this very thoroughly yet. I think Aften's exponent strategy implementation is pretty good though.

edit: oh, and not that it will matter a lot, but for any further comparisons you should turn on block switching in Aften since DD does block switching by default.

[tebasuna51]

...
As an experiment, I stripped all of that out and was able to get working channel coupling (although it was much slower obviously). I'm still working on trying to get it ported over to Aften.
...

Very much appreciated your effort to improve Aften.

[Abradoks]

for any further comparisons you should turn on block switching in Aften since DD does block switching by default.
It was turned on for aften_spec and aften_wide in the first comparison I have posted.
And thank you for your work on aften. I hope we'll see even better results than DD has.

[Blue_MiSfit]

How would you clever AC3 encoding folks suggest maximizing the quality of 384kbps AC3 for 3/2 + LFE encoding?

I'm currently using EncWAVtoAC3 4.2, without any modifications to the advanced parameters.

~MiSfit

[jruggle]

How would you clever AC3 encoding folks suggest maximizing the quality of 384kbps AC3 for 3/2 + LFE encoding?

I'm currently using EncWAVtoAC3 4.2, without any modifications to the advanced parameters.


aften -b 384 -exps 32 -s 1 -bwfilter 1

Also, if you can increase the -w parameter without quality going below q=240 then that would be good. The -w -2 (variable bandwidth) option is too experimental to be used generally, but doing a test run with it can give you a good idea of how much you can increase the bandwidth without affecting quality too badly.

[Blue_MiSfit]

Thanks, jruggle.

I have two different Dolby encoding systems (Media Encoder Server for Mac, and a DP600 encoder server), and will be comparing the quality of both vanilla AC3 and E-AC3. What software was being used earlier in the thread to generate those pretty spectrographs?

~MiSfit

[nurbs]

Spectrographs don't tell you anything about the quality.

[tebasuna51]

aften -b 384 -exps 32 -s 1 -bwfilter 1

Also, if you can increase the -w parameter without quality going below q=240 then that would be good. The -w -2 (variable bandwidth) option is too experimental to be used generally, but doing a test run with it can give you a good idea of how much you can increase the bandwidth without affecting quality too badly.

The same parameters are recommended when use the max bitrate 640 Kb/s?

Maybe the bwfilter is not necesary when use max banwidth.

[wisodev]

New Aften build is available : http://code.google.com/p/wavtoac3encoder/downloads/list

[TFM_TheMask]

New Aften build is available : http://code.google.com/p/wavtoac3encoder/downloads/list

Can you tell us what has changed in this new version?

[Yobbo]

Wisodev, can you just drag & drop these new libaften.dll's over the existing dll's in your Wav to AC3 Encoder? Will you be updating the Encoder soon?

[wisodev]

Wisodev, can you just drag & drop these new libaften.dll's over the existing dll's in your Wav to AC3 Encoder? Will you be updating the Encoder soon?

New version of WAV to AC3 Encoder is available for download , version 4.3 includes libaften.dll version from 20091226 (Git master head snapshot).
http://code.google.com/p/wavtoac3encoder/downloads/list

[tebasuna51]

Thanks wisodev.

There are some channel coupling improvement in last Aften version?

[jruggle]

Thanks wisodev.

There are some channel coupling improvement in last Aften version?
There really have not been many changes in a while. Just some tweaks to the exponent strategy decision. There were a few code structure changes in preparation for channel coupling though. And it looks like Prakash has added Windows x64 support while I was away for Christmas. :)

[TFM_TheMask]

New version of WAV to AC3 Encoder is available for download , version 4.3 includes libaften.dll version from 20091226 (Git master head snapshot).
http://code.google.com/p/wavtoac3encoder/downloads/list

Isn't it possible to only get 2 aften libraries which can detect the PC's SIMD capabilities (SSE, SSE2, SSE3, MMX). So only libaften_x86.dll and libaften_x64.dll and not 6 or more different libraries?

So what I mean is a compiled libaften library with all SIMD instructions in it and it checks which instructions it can use depending on a PC's capabilities.

[wisodev]

Isn't it possible to only get 2 aften libraries which can detect the PC's SIMD capabilities (SSE, SSE2, SSE3, MMX). So only libaften_x86.dll and libaften_x64.dll and not 6 or more different libraries?

So what I mean is a compiled libaften library with all SIMD instructions in it and it checks which instructions it can use depending on a PC's capabilities.

All my libaften.dll builds have enabled ALL Aften SIMD optimizations (if your hardware supports specific SIMD it is enabled on runtime). But additionally the MMX, SSE, SSE2, etc. builds have extra optimization made by Intel C++ Compiler and REQUIRE presence of SIMD support in hardware (else you can expect crushes), so you just can use default builds: libaftendll_x86 and libaftendll_AMD64 most of the time.

[TFM_TheMask]

All my libaften.dll builds have enabled ALL Aften SIMD optimizations (if your hardware supports specific SIMD it is enabled on runtime). But additionally the MMX, SSE, SSE2, etc. builds have extra optimization made by Intel C++ Compiler and REQUIRE presence of SIMD support in hardware (else you can expect crushes), so you just can use default builds: libaftendll_x86 and libaftendll_AMD64 most of the time.

Ok thanks, will do that. One little question, the libaftendll_AMD64 is only for AMD x64 processors or also Intel x64 processors?

[wisodev]

Ok thanks, will do that. One little question, the libaftendll_AMD64 is only for AMD x64 processors or also Intel x64 processors?

It's AMD and Intel CPU compatible (actually it's compiled using Intel C++ Compiler).

[TFM_TheMask]

It's AMD and Intel CPU compatible (actually it's compiled using Intel C++ Compiler).

Better change the name to x64 instead of AMD64.

:thanks:

[raquete]

New version of WAV to AC3 Encoder is available for download , version 4.3 includes libaften.dll version from 20091226 (Git master head snapshot).
http://code.google.com/p/wavtoac3encoder/downloads/list
thank you so much wisodev!
(no more 'old problems' running Win2000, changing to Win7)

veru very Happy New year! :)

[MrVideo]

Better change the name to x64 instead of AMD64.

Ya, had me confused. I downloaded the AMD64 versions, only to discover that it is for 64bit OSs. I have the AMD64 dual core, but run 32bit OS.

[MrVideo]

I've been using an older version of the files found in the aften_x86 directory.

Which is better, to use the files in the aften_x86 directory, or the files in the libaftendll_x86 directory?

If the libaftendll_x86 contents are used, I'm going to assume that they are placed in the same directory as I've previously done with the aften_x86 contents.

There was nothing in the top level readme file about which should be used, and why.

Thanks.

[canuckerfan]

does encwavtoac3 support stereo wav input or do I need to demux the stereo wav into 2 mono wav files?

[Boulder]

No need for that, it supports stereo WAVs just fine.

[canuckerfan]

but when I hit encode it gives me an error saying "Supported are minimum 2 and maximum 6 mono input files!"

[Boulder]

I just tried encoding a 32-bit stereo WAV with the latest version, no problems here.

[canuckerfan]

^maybe its my file... I tried the first file on this webpage ( M1F1-Alaw-AFsp.wav): http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html

and I got the same error. does that one work for you?

ps: the second one also gives me the same error(M1F1-AlawWE-AFsp.wav). somethings definately wrong with my setup. i am running v4.4 from the zipped package

[Boulder]

I think it's a problem with the encoding type in the WAV file.. try some "regular", i.e. PCM encoded WAV. For example this one is such: http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples/AFsp/M1F1-int16-AFsp.wav

[wisodev]

does encwavtoac3 support stereo wav input or do I need to demux the stereo wav into 2 mono wav files?

When you are using "Mmultiple mono input" mode the input files must be mono, if you turn off this mode then you can encode single multichannel input file.

[canuckerfan]

I think it's a problem with the encoding type in the WAV file.. try some "regular", i.e. PCM encoded WAV. For example this one is such: http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples/AFsp/M1F1-int16-AFsp.wav
I get the same error with that file as well.
When you are using "Mmultiple mono input" mode the input files must be mono, if you turn off this mode then you can encode single multichannel input file.
how do I disable mono input mode?

[tebasuna51]

Uncheck the 'Multiple mono input' box.

[canuckerfan]

Uncheck the 'Multiple mono input' box.
wow. I can't believe I missed something like that:eek:

thanks

edit: I have another problem now... my wav source is not sampled to 48k and I have to feed an avs script into encwavtoac3 so that it will convert it. here's my script:

#avisource("G:\temp\1.avi")+avisource("G:\temp\2.avi")
#SoundOut()

WavSource("G:\temp\bgf.wav")
ssrc(48000)

however as soon as I drop the avs into the program it waits about a second and gives an error. "Failed to initialize avisynth". then I have to end task it. I tried playing the avs file in mpc and works just fine... btw i'm running AviSynth 2.5.8

edit: I just found out that it only works with avs 2.5.7 and below as of Aug 2/'09. I guess that's still the case. I'll try it on 2.5.7 and give it a whirl.

[b66pak]

@canuckerfan or you can forget about drag&drop and use the add button...
_

[canuckerfan]

^I can't believe that made a difference haha. thanks, it works now.

[wisodev]

@canuckerfan or you can forget about drag&drop and use the add button...
_

Should be fixed in version 4.5 :)

[Maxiuca]

I keep getting "Encoder Error: Failed to initialize encoder". Funny thing is that when I start the program for the first time after system restart I can get it to encode, but then this error keeps appearing in logs. I'm on 64-bit Vista Ultimate. Any ideas?

[wisodev]

I keep getting "Encoder Error: Failed to initialize encoder". Funny thing is that when I start the program for the first time after system restart I can get it to encode, but then this error keeps appearing in logs. I'm on 64-bit Vista Ultimate. Any ideas?

What is the input file format ? Are you using 64 or 32 bit version ? Is the program configuration preserved after restart ?

[Maxiuca]

Sorry for a false alarm, problem solved, my bad. I was trying to encode a 5.0 ac3 with LFE low-pass filter enabled. It was pure coincidence that the program started encoding after two reboots (different settings).

[Midzuki]

Let's say, I have a 3min 36sec stereo WAV file (extracted from an Audio CD), and compress it with Aften.exe @ 192kbps, and then use spdifer.exe to obtain a DDWAV stream. Instead of getting a 36.3MB .wav, spdifer produces a 34.8MB one (and if I try a different bitrate, e.g., 256kbps, the resulting SPDIFed .wav has only 17.4MB :eek: ). This problem *does not* happen with ac3 files generated by Soft Encode, for example.

[Midzuki]

OK, today I found this post by madshi:

http://forum.doom9.org/showpost.php?p=1231929&postcount=945

which made me give a try to the latest release of EncWAVToAC3. Sadly the "44.1kHz problem" seems to be even worse in the libaften-GUI encoder: the same .WAV source-file mentioned in my post above becomes a 5.77MB .ac3 encoded at 224kbps by EncWAVToAC3, however spdifer.exe obtains a
12-byte .WAV file from that. :eek: Again, spdifer keeps working correctly on files created by a "Dolby-certified" encoder...

[tebasuna51]

OK, today I found this post by madshi:

http://forum.doom9.org/showpost.php?p=1231929&postcount=945

The problem don't exist now.

BTW, I can't understand the process.

You have a stereo wav file 3m:36s (36,3 MB), encoded with ac3 192 Kb/s (4.9 MB) and spdifer convert the file to ddwav (34.7 or 36.3 MB)
What is the benefit to have a lossy compressed audio with the same size than the lossless source?

With my stereo samples, encoded with Aften and Soft Encode, obtain the same wrong output size when use spdifer.

I have some 5.1 ddwav and the size isn't the size of the wav sources. Is the size of the ac3 x 2.2
ac3 file 31.5 M -> ddwav file 69.6 MB

With spdifer I obtain the same ratio with 5.1 ac3 files.
Maybe spdifer don't work properly with stereo files.

[Midzuki]

It depends on the spdifer's version one uses, I think. :confused: FWIW, I'm using the build 0.1a.

Anyway, I am not thinking in terms of "benefits", just in terms of "standard-compliance" --- if someone intends to create a multichannel AC3 Audio CD, it's better that they know they shouldn't be using Aften for doing that. :devil: Given that spdifer does work correctly with the 32kHz && 48kHz AC3 outputs from Aften, it seems to me, *now*, that the 44.1kHz encodes from Aften are not "correct enough". I remember that "many months ago", just for testing/learning purposes,
I created some stereo AC3s @ 44.1kHz with aften.exe, and realized that most of them were played back @ 48kHz when decoded by AC3Filter. Because such files were played back @ 44.1kHz when decoded by ffdshow, I (erroneously) concluded that the actual problem "had to be" in AC3Filter, "not" on the 44.1kHz files generated by Aften.exe.

P.S.: So, ¿¿¿ is there any better alternative to spdifer.exe ???

P.P.S.: ffdshow (r1376) audio processor is not an option,
because its AC3 encoder supports only 48kHz and 32kHz.

[tebasuna51]

...
--- if someone intends to create a multichannel AC3 Audio CD, it's better that they know they shouldn't be using Aften for doing that.
For what?
With spdifer.exe (ac3filter_tools_0_31b) I obtain the same filesize, framesizes and headers when convert to ddwav a 5.1 wav encoded to ac3 with Aften or Soft Encoder.

P.S.: So, ¿¿¿ is there any better alternative to spdifer.exe ???

I don't know.

[Midzuki]

With spdifer.exe (ac3filter_tools_0_31b) I obtain the same filesize, framesizes and headers when convert to ddwav a 5.1 wav encoded to ac3 with Aften or Soft Encoder.

I have some 5.1 ddwav and the size isn't the size of the wav sources. Is the size of the ac3 x 2.2

Something is wrong, then.

Maybe spdifer don't work properly with stereo files.

It DOES work with every stereo .ac3 @ 32 and 48kHz I have tested so far.

What are you trying to say, man? :)

P.S.: Another test --- stereo WAV @ 44.1kHz, encoded with AC3 ACM via VirtualDub; resulting file was "RIFF-stripped", and then given to spdifer.exe;
DDWAV file size = 36.3MB, as it should be. :rolleyes: :devil:
----------------------------------------------
UPDATE:

Just downloaded and tested the latest version of spdifer, and
this time, it did produce a DDWAV with correct-length from an Aften 44.1kHz file :) :) :)

So,

1) the real problem was in the old version of spdifer.exe, OK ;

2) BIG QUESTION: what would be the "magic" differences between the 44.1kHz outputs from Aften and the ones from the other encoders ??? :confused:

[jruggle]

2) BIG QUESTION: what would be the "magic" differences between the 44.1kHz outputs from Aften and the ones from the other encoders ??? :confused:
My first guess is that some other encoders do not vary the frame size. The frame size can be the base frame size or +16 bits at 44.1kHz per the AC-3 spec. Aften uses this option to make sure the average bitrate is exact. Otherwise it would be slightly less than the nominal bitrate. This is not an issue for 48kHz or 32kHz because the math is nice and even. :) Maybe the old version of spdifer did not take this into account?

If you want a reference, see document ATSC A/52B Table 5.18.

[Midzuki]

My first guess is that some other encoders do not vary the frame size. The frame size can be the base frame size or +16 bits at 44.1kHz per the AC-3 spec. Aften uses this option to make sure the average bitrate is exact. Otherwise it would be slightly less than the nominal bitrate. This is not an issue for 48kHz or 32kHz because the math is nice and even. :) Maybe the old version of spdifer did not take this into account?

If you want a reference, see document ATSC A/52B Table 5.18.

Thanks for answering. :)

Yes, I had already looked at the "top-secret" .PDF. :D
But, as you can see, it appears it was NOT ONLY spdifer.exe that didn't consider the «better» possibility. I mean, it seems Aften was/is the first/only "well-known" AC3 encoder to use two different frame sizes for the 44.1kHz sampling rate.

Again, :thanks:

[tebasuna51]

... I mean, it seems Aften was/is the first/only "well-known" AC3 encoder to use two different frame sizes for the 44.1kHz sampling rate.

Nope, the test I make before encoding the same wav 44.1KHz file:

Aften: 3161 frames (2536 "short" + 625 "long"), precise bitrate
Soft Encode: 3161 frames (1585 "short" + 1576 "long"), imprecise bitrate

[Midzuki]

Nope, the test I make before encoding the same wav 44.1KHz file:

Aften: 3161 frames (2536 "short" + 625 "long"), precise bitrate
Soft Encode: 3161 frames (1585 "short" + 1576 "long"), imprecise bitrate

:thanks: for the accurate info.

*THUMBS UP*

[Selur]

I wanted to add support for aften inside Hybrid but ran into the problem that aften doesn't show progress when I pipe into it from ffmpeg.
Here are the command lines I use:
ffmpeg -v -10 -threads 4 -i "input.ac3" -acodec pcm_s16le -ac 6 -ar 48000 -f wav - | aften -b 256 -readtoeof 1 - "output.ac3"
and
ffmpeg -threads 4 -v -10 -y -i "input.ac3" -f u16le -acodec pcm_s16le - | sox --temp "D:\Encoding Temp" -t raw -e signed-integer -2 -c6 -r48000 - -t wav - remix -m 1v0.2646,3v0.1870,4v0.1870,5v0.2291,6v0.1323 2v0.2646,3v0.1870,4v0.1870,5v-0.1323,6v-0.2291 norm | aften -b 128 -readtoeof 1 - "output.ac3"
I know that when feeding aften directly with a file progress works, but since I want to be able to filter with sox and not use intermediate files I really would like too stay with the piping the inptu into aften.

I know that '-v' shows me the current frame processed, but I don't know if I can somehow judge from the frame number to the current position processed.

So if someone knows how to get aften to show progress (e.g. through altering my command line) or knows how to conclude from the current frame to the current position any help is welcome.

Cu Selur

Ps.: Only way to get current position atm I can think of it to get it from ffmpeg output.

[jruggle]

I wanted to add support for aften inside Hybrid but ran into the problem that aften doesn't show progress when I pipe into it from ffmpeg.

Aften needs to know the total number of samples in order to show progress. It can get it 2 ways. For raw audio it is determined from the file size. For wav, aiff, etc, it is determined from the header. When ffmpeg sends output to a pipe, the file size cannot determined. When ffmpeg writes a wav file, the header is not updated with the data size until the end of encoding.

It appears that sox actually writes a valid data size in the header even when streaming to a pipe, if it knows it. Obviously if you're streaming from ffmpeg to sox to aften then sox cannot know the duration either so it cannot pass it to aften.

One workaround might be to add an option to aften so the user can specify the total number of input samples, overriding information from the file size or header if present. Then your program or script could get the duration from ffmpeg (or ffprobe) and multiply by the sample rate. If that option would be useful, I would consider adding it.

[Selur]

So sample rate * time in seconds = frame count ? If so you don't need to add such an option since I already got the duration in seconds and calculating the progress from the output I get when calling with '-v' as parameter isn't a problem. :)

Cu Selur

[Anima123]

Hi jruggle,

Any plan for introducing psychoacoustic model in your encoder? I am the one who have offered you some e-books at the hydrogenaudio forum.

Regards,

[tebasuna51]

So sample rate * time in seconds = frame count ? If so you don't need to add such an option since I already got the duration in seconds and calculating the progress from the output I get when calling with '-v' as parameter isn't a problem. :)

Cu Selur
Exactly:
sample rate * time in seconds = sample count

Frames, in compressed audio, can have many samples, for instance a frame in standard ac3 have 1536 samples.

[Selur]

Exactly:
sample rate * time in seconds = sample count

Frames, in compressed audio, can have many samples, for instance a frame in standard ac3 have 1536 samples.
when using '-v' aften shows each frame's stats assuming these are ac3 frames:
(sample rate * time in seconds)/1536 would provide me the number of frames aften will show for piped input, right?

Cu Selur

[LigH]

Look at the number closely! 1536 = 0600h.

[Selur]

hups, thanks for the correction! (fixed it in the above post)
-> it's working !!

[jruggle]

Any plan for introducing psychoacoustic model in your encoder? I am the one who have offered you some e-books at the hydrogenaudio forum.

Eventually. I got it working once but I wasn't satisfied with it. I have very limited time, but it's still on my todo list.

[Vincent Vega]

can't find an explanation for encoder output log, like these lines i get in megui:

progress: 100% | q: 333.5 | bw: 38.0 | bitrate: 192.0 kbps
progress: 100% | q: 284.0 | bw: 38.0 | bitrate: 192.0 kbps

what does q and bw (bitwriter?) mean? somewhere i can read about it?

[Abradoks]

what does q and bw (bitwriter?) mean? somewhere i can read about it?
Quality and bandwidth. Try "aften -longhelp".

[Vincent Vega]

thanks man

[video_magic]

Just found this at HA.org...and it sounds promizing. :)


Wisodev's builds :: http://win32builds.sourceforge.net/aften/index.html
For whose who prefer GUIs instead of command lines, I've made a small one (http://kurtnoise.free.fr/index.php?dir=Aften/&file=AftenGUI-1.4.zip).


In this first post, I have noticed that the wizodev link is a 404. Could you update to point to:
http://win32builds.sourceforge.net/

That seems current, thanks.

[LigH]

Please note:

http://sourceforge.net/projects/win32builds/files/
As of 2008-05-10 0:00:00 GMT, this project is no longer under active development.

And http://kurtnoise.free.fr is just as dead.

So - where is a really "current" alternative?

[TFM_TheMask]

Try http://code.google.com/p/wavtoac3encoder/

[video_magic]

Please note:

http://sourceforge.net/projects/win32builds/files/


And http://kurtnoise.free.fr is just as dead.

So - where is a really "current" alternative?


Oops, I meant that is current as in it doesn't give a '404 page', but the builds are quite old you are right.

[tebasuna51]

In http://code.google.com/p/wavtoac3encoder/downloads/list the aften-git-20091226-icl11_1_054 is the last build I know.

[wisodev]

Version 4.6 and later 4.7 of WAV to AC3 Encoder have been released today:

Changes:
- Fixed: Issue 5: Importing .mux files in ANSI not possible
- Added support for avisynth in AMD64 (64-bit) builds using http://code.google.com/p/avisynth64/.

Downloads:
http://code.google.com/p/wavtoac3encoder/downloads/list

[Yobbo]

THANK YOU, Wisodev! I always look forward to updates for WAVtoAC3Enc!!! It's a tool in my kit that I use regularly!

[MrVideo]

Version 4.6 and later 4.7 of WAV to AC3 Encoder have been released today

OK, what does this program do that aften doesn't?

[MrVideo]

Version 4.6 and later 4.7 of WAV to AC3 Encoder have been released today

I got an e-mail that showed a response that you gave to my query, yet it appears that the response was deleted.

All of my audio work is done via the CLI, in scripts that I have written to automate the process of stuff that I do.

[wisodev]

OK, what does this program do that aften doesn't?

Its GUI version.

[wisodev]

I got an e-mail that showed a response that you gave to my query, yet it appears that the response was deleted.

All of my audio work is done via the CLI, in scripts that I have written to automate the process of stuff that I do.

But my GUI has other goal then CLI version:
- It has optimized aften builds.
- Supports Avisynth scripts under Win32 and x64 OS.
- Can be used to automate several jobs from command-line using presets.
- Has MUX wizard.
- Supports unicode file names.
- You can access all aften options in one place and tweak quickly.

[MrVideo]

But my GUI has other goal then CLI version

Thanks for the info.

Silly me, I should have also followed the link in you signature.

[DVB-Freak]

Hello wisodev!

Is the project dead?
The site on google don't work anymore.


Best regards
DVB-Freak

[Blue_MiSfit]

encwavtoac3 5.0 is out now http://www.videohelp.com/tools/EncWAVtoAC3

[sneaker_ger]

Too bad AviSynth input still does not work properly.

[LigH]

That works in BeHappy and MeGUI (which uses the engine derived from BeHappy).

http://behappy.codeplex.com/

http://forum.doom9.org/showthread.php?t=104686

[sneaker_ger]

Thanks for the info. Didn't know BeHappy is still being developed (probably mixed it up with BeSweet).

[tebasuna51]

http://code.google.com/p/wavtoac3encoder/

still work for me.

[sneaker_ger]

Yes, the site seems to work again. He probably reactivated it or something.

[Brazil2]

Does anyone know why AC3 files encoded with Aften are not showing any information in MediaInfo ?

All other AC3 tracks I have, extracted from DVD / BD and samples from this forum, are showing details in MediaInfo like this:
General
Complete name : F:\Test\ac3test.ac3
Format : AC-3
Format/Info : Audio Coding 3
File size : 3.20 MiB
Duration : 1mn 0s
Overall bit rate : 448 Kbps

Audio
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Duration : 1mn 0s
Bit rate mode : Constant
Bit rate : 448 Kbps
Channel(s) : 6 channels
Channel positions : Front: L C R, Side: L R, LFE
Sampling rate : 48.0 KHz
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 3.20 MiB (100%)


But files encoded with Aften are showing this:
General
Complete name : F:\Test\ac3.ac3
File size : 1.57 MiB

I'm using aften-git-20091226-icl11_1_054-win32 (http://code.google.com/p/wavtoac3encoder/downloads/detail?name=aften-git-20091226-icl11_1_054-win32-bin.zip&can=2&q=), mostly through Foobar but I got the same results in a command line box, with these parameters:
-readtoeof 1 -pad 0 -exps 32 -s 1 -b 448 - %d

While muxed into an MP4 file with YAMB then the information becomes available:
General
Complete name : F:\Test\ac3.mp4
Format : MPEG-4
Format profile : Base Media
Codec ID : isom
File size : 1.57 MiB
Duration : 29s 326ms
Overall bit rate : 449 Kbps
Encoded date : UTC 2011-08-26 13:31:00
Tagged date : UTC 2011-08-26 13:31:00
Writing application : Yamb 2.1.0.0 [http://yamb.unite-video.com]

Audio
ID : 1
Format : AC-3
Format/Info : Audio Coding 3
Mode extension : CM (complete main)
Codec ID : ac-3
Duration : 29s 326ms
Bit rate mode : Constant
Bit rate : 448 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 44.1 KHz
Bit depth : 16 bits
Compression mode : Lossy
Stream size : 1.57 MiB (100%)
Title : AC-3
Encoded date : UTC 2011-08-26 13:31:00
Tagged date : UTC 2011-08-26 13:31:00

But it's gone again once the track has been extracted from the MP4.
However the files created by Aften are playing fine with Foobar and VLC.

Is it an Aften issue or a MediaInfo issue ? Any thoughts ?

[Kurtnoise]

but I got the same results in a command line box, with these parameters:
-readtoeof 1 -pad 0 -exps 32 -s 1 -b 448 - %d

why are you using -pad 0 -exps 32 -s 1 ? try without these switches...

[jruggle]

Does anyone know why AC3 files encoded with Aften are not showing any information in MediaInfo ?
...
Is it an Aften issue or a MediaInfo issue ? Any thoughts ?
My guess is MediaInfo since you say the files play correctly.

Just for kicks, add '-xbsi1 1' to your commandline and see if that changes what MediaInfo prints out.

[jruggle]

why are you using -pad 0 -exps 32 -s 1 ? try without these switches...
'-s 1' should not cause any issues since the block switch flags are in the audio blocks, not in the frame header. '-pad 0' would have no effect on this since it only changes how source audio data is passed to libaften. '-exps 32' should definitely not have any effect on this since it's just an algorithmic decision option.

The only thing I can think of right off-hand is that maybe MediaInfo expects the extended bitstream format. That is what most commercial AC-3 encoders produce, but it is certainly not required. Aften only uses it if one of the extended metadata options is set.

[Brazil2]

why are you using -pad 0 -exps 32 -s 1 ?
Because of this:
To convert a wav file 5.0 you can use WavToAc3Enc or eac3to with external Aften.exe encoder:

eac3to source50 stdout.wav | Aften -pad 0 -readtoeof 1 -exps 32 -s 1 -b 640 - output.ac3

where:
-exps 32 -s 1 -b 640
is the best quality (and slow encode) than Aften can offer.



try without these switches...
It's the same.



Just for kicks, add '-xbsi1 1' to your commandline and see if that changes what MediaInfo prints out.
No change either.

[jruggle]

It's the same.
...
No change either.
Hmm...not sure then. I am unable to reproduce the issue in Linux with MediaInfo v0.7.16. I get normal output. Could you provide a small sample?

[Brazil2]

I think I've found out the issue: it only happens when the source file is not 48 KHz. As I'm not doing any sampling rate conversion I ended up with a 44.1 KHz file which doesn't show any information in MediaInfo.

But all of the tests I've done with the resulting files at 48 KHz are showing detailed information in MediaInfo. And BTW same goes for AC3 files created with ffmpeg so there are chances it's a MediaInfo issue.

[jruggle]

I think I've found out the issue: it only happens when the source file is not 48 KHz. As I'm not doing any sampling rate conversion I ended up with a 44.1 KHz file which doesn't show any information in MediaInfo.

But all of the tests I've done with the resulting files at 48 KHz are showing detailed information in MediaInfo. And BTW same goes for AC3 files created with ffmpeg so there are chances it's a MediaInfo issue.
That's odd. I thought at one point that might be the cause, but 44.1kHz files also work fine for me with MediaInfo v0.7.16.

[tebasuna51]

Because of this:
...
-exps 32 -s 1 -b 640
...

Then for what you put: -exps 32 -s 1 -b 448 ?

Let the defaults for -b 448

No problem with AC3 streams, maybe a MediaInfo bug like put:
Bit depth : 16 bits
Lossy compressed streams don't have Bit depth, and we can't know the bitdepth of the source, can be 16, 24, 32, ... bits

@jruggle, please a new version with last improvements.

[Brazil2]

Weird, I got it working with some 44.1 KHz files but not with some other ones.
I've uploaded some samples here: http://www.multiupload.com/9Z1X01V9FF

Two of them are working fine with MediaInfo but the third one doesn't.

[tebasuna51]

... the third one doesn't.

No problem with your file:
File ........: D:\Internet\Aften\NotOK-Owner.ac3
Size ........: 1642300 bytes
----------------------------------------- First Frame Info
SampleRate ..................: 1 (44100 KHz)
BitRate .....................: 15 (448 Kb/s)
Version (bsid) ..............: 6 (Alternate sintax)
Bit Stream mode (bsmod) .....: 0 (main audio service: complete main, CM)
Audio coding mode (acmod) ...: 2 (2/0 - L, R)
Dolby Surround Mode .........: 0 (not indicated)
Low frequency effects channel: 0 (Not present)
Dialogue normalization ......: - 31 dB
RF atenuattion ..............: 0 (Not present)
Languaje ....................: 0 (Not present)
Audio Production Info .......: 0 (Not present)
CopyRight bit ...............: 0
Original bit ................: 1
Preferred stereo downmix mode: 0 (not indicated)
Lt/Rt center mix level ......: 4 (0.707, -3.0 dB)
Lt/Rt surround mix level ....: 4 (0.707, -3.0 dB)
Lo/Ro center mix level ......: 4 (0.707, -3.0 dB)
Lo/Ro surround mix level ....: 4 (0.707, -3.0 dB)
Xbsi2e ......................: 0 (Not present)
Additional Bsi ..............: 0 (Not present)
Block switch flags ..........: 0
Dither flags ................: 3
Dynamic Range Info ..........: 0 (Not present)
------------------------------------------- Rest of Frames
Frame SR BR PB BS BM AM LF DN DyRaCo Bytes Before Header
--------- -- -- -- -- -- -- -- -- ------ -------------------
2 1 15 0 6 0 2 0 31
3 1 15 1 6 0 2 0 31
4 1 15 1 6 0 2 0 31
5 1 15 0 6 0 2 0 31
6 1 15 0 6 0 2 0 31
7 1 15 0 6 0 2 0 31
8 1 15 0 6 0 2 0 31
9 1 15 0 6 0 2 0 31
10 1 15 0 6 0 2 0 31
11 1 15 1 6 0 2 0 31
12 1 15 1 6 0 2 0 31
13 1 15 0 6 0 2 0 31
...
833 1 15 1 6 0 2 0 31
834 1 15 1 6 0 2 0 31
835 1 15 0 6 0 2 0 31
836 1 15 0 6 0 2 0 31
837 1 15 0 6 0 2 0 31
838 1 15 0 6 0 2 0 31
839 1 15 0 6 0 2 0 31
840 1 15 0 6 0 2 0 31
841 1 15 0 6 0 2 0 31
842 1 15 0 6 0 2 0 31
--------------------------------------------- Revised Info
Total Frames ......: 842
Duration ..........: 29,328 seconds. ( 0 h. 0 m. 29,328 s.)
------------------------------------------------- End Info

But don't work fine with MediaInfo, is a MediaInfo bug.

[jruggle]

@jruggle, please a new version with last improvements.
I don't really have time for a release right now. I'm very busy improving the (E-)AC3 encoder in Libav. It's better than Aften now, and I will most likely no longer do additional improvements to Aften other than bug fixes.

[Brazil2]

No problem with your file:
----------------------------------------- First Frame Info
SampleRate ..................: 1 (44100 KHz)
BitRate .....................: 15 (448 Kb/s)
Version (bsid) ..............: 6 (Alternate sintax)
Bit Stream mode (bsmod) .....: 0 (main audio service: complete main, CM)
Audio coding mode (acmod) ...: 2 (2/0 - L, R)
Dolby Surround Mode .........: 0 (not indicated)
Low frequency effects channel: 0 (Not present)
Dialogue normalization ......: - 31 dB
RF atenuattion ..............: 0 (Not present)
Languaje ....................: 0 (Not present)
Audio Production Info .......: 0 (Not present)
CopyRight bit ...............: 0
Original bit ................: 1
Preferred stereo downmix mode: 0 (not indicated)
Lt/Rt center mix level ......: 4 (0.707, -3.0 dB)
Lt/Rt surround mix level ....: 4 (0.707, -3.0 dB)
Lo/Ro center mix level ......: 4 (0.707, -3.0 dB)
Lo/Ro surround mix level ....: 4 (0.707, -3.0 dB)
Xbsi2e ......................: 0 (Not present)
Additional Bsi ..............: 0 (Not present)
Block switch flags ..........: 0
Dither flags ................: 3
Dynamic Range Info ..........: 0 (Not present)
------------------------------------------- Rest of Frames
Frame SR BR PB BS BM AM LF DN DyRaCo Bytes Before Header
--------- -- -- -- -- -- -- -- -- ------ -------------------
2 1 15 0 6 0 2 0 31
3 1 15 1 6 0 2 0 31
...
841 1 15 0 6 0 2 0 31
842 1 15 0 6 0 2 0 31
--------------------------------------------- Revised Info
Total Frames ......: 842
Duration ..........: 29,328 seconds. ( 0 h. 0 m. 29,328 s.)
------------------------------------------------- End Info
Which tool are you using that gives this information ?

[Zenitram]

I think I've found out the issue: it only happens when the source file is not 48 KHz.

You're right: there was a bug in MediaInfo preventing analysis of some non-48 KHz file.

That's odd. I thought at one point that might be the cause, but 44.1kHz files also work fine for me with MediaInfo v0.7.16.

I added CRC check in newest versions, and this piece of code was buggy with non-48 KHz streams.

Corrected in latest development snapshot (https://sourceforge.net/projects/mediainfo/files).

[tebasuna51]

Which tool are you using that gives this information ?
The attached one, read some info here:
http://forum.doom9.org/showthread.php?p=1159994#post1159994

Last version LeeAudBi7.7z here (https://forum.doom9.org/showthread.php?p=424067#post424067)

[Brazil2]

The attached one, read some info here:
http://forum.doom9.org/showthread.php?p=1159994#post1159994
Thanks :) :thanks:

[Mug Funky]

@ tebasuna51:

is it possible that this tool could be used to patch an existing ac3 with data (such as DRC values) from another?

i suppose we'd have to guarantee the blocks are all aligned and all that, or is DRC frame based?

here's where i regret not learning C when i had time back in the day. it doesn't seem fair to ask for stuff without contributing something (or at least doing it myself from the source).

[tebasuna51]

is it possible that this tool could be used to patch an existing ac3 with data (such as DRC values) from another?

i suppose we'd have to guarantee the blocks are all aligned and all that, or is DRC frame based?

Yes each frame must have the same length and we can't insert data into a frame.

We can replace something like Dialog Normalization, the flag about Dolby ProLogic, ... and after recalculate the CRC's of the frame.

But we can't insert new data.

[xabregas]

Is there a way to cutoff frequency to lets say 80HZ and add more bass to audio??

[LigH]

Filtering and mixing audio is not the duty of an encoder, at least not to change the sound to personal preferences of a single listener, instead it should try to reproduce the sound with as little distortion as possible. But there is the exception of quality optimization (this is the mentioned "cutoff" of hardly noticable highest frequencies, usually above 12...16 kHz, which mostly disturb the encoding quality) and preparation of content with a specific purpose. The LFE channel would be designed to contain low frequencies only; but is intentionally named "Low Frequency Effects: It is usually not meant to contain only the filtered low frequencies of already existing full spectrum channels, but rather additional effects (especially regarding audio of cinematic movies). You may use the LFE mixdown to add existing low frequencies of present channels into the LFE channel; but it is (AFAIR) not intended to serve this purpose only.

[tebasuna51]

The cutoff frequency is for cut high frequencies, like LigH say, and is related with bitrate because save high frequencies need much more bitrate.

The low frequencies don't need be cut because don't need much bitrate, and always is saved like original without changes.

BTW please remember than Aften encoder is deprecated and replaced with ffmpeg (https://forum.doom9.org/showthread.php?p=1522077#post1522077) (supported by jruggle) encoder.
Please read some improvements here: https://forum.doom9.org/showthread.php?p=1794293#post1794293

[xabregas]

Thanks for the answers but i didnt explained the reasons behind why i wanted to do this.

As you might all be aware now amazon streams come with the new Dolby Digital Plus aka E-AC3. At first i tought i would get better audio and stopped purchasing itunes and started buying amazon. But the problem is i didnt get better. My soundcard can decode only AC3 and DTS. So i can`t seem to send E-AC3 trough spdif. I solved the issue at first by doing real time ac3 encoding of E-AC3 and sending through spdif and it worked but the audio was all wrong. Too strong on explosions and voices too low even with DDR enabled and sometimes the voices come strong and wrong mapped. I went to Amazon foruns asking if they could provide AC3 stream for all the episodes but they dont seem to be interested in doing that.

So how can i encode E-AC3 real time or not to AC3 and get the usual quality i was used to when i used the itunes audio? Or do i need to purchase every episode i already got from amazon on itunes just to have proper audio sending through?? I tought cutting off high frquencies and making dialogue normalizer to -31 would solve this? But aften only accept wav streams. How do i decode e-ac3 stream to wav the right way?

Thanks

[tebasuna51]

Like I say before forget Aften.exe

You can recode directly EAC3 to AC3 with ffmpeg with a batch file like:

rem Modify next line with the path to ffmpeg in your system
set FFMPEG=D:\Programa\Audio\0\ffmpeg
FOR %%I in (*.eac3) DO "%FFMPEG%" -drc_scale 0 -i "%%I" -acodec ac3 -ac 6 -ab 448k -center_mixlev 0.707 "%%I_.ac3"
pause

All eac3 in folder are converted to ac3 at bitrate selected (here 448 Kb/s)

[xabregas]

I am a bit noob to work with what you gave me but after testing many times and changing the paths i got it to work. Turns out the problem was in the .eac3. It worked with .ec3. :D

Thanks!!!

[GMJCZP]

Sorry if I'm posting in the wrong way but I'm trying to download the latest version of LeeAudioDib and I get an error message in Sendspace.

[filler56789]

Sorry if I'm posting in the wrong way but I'm trying to download the latest version of LeeAudioDib and I get an error message in Sendspace.

Give a try to this URL:

https://docs.google.com/file/d/0B0rPtrLUrGhIYmM1MDNhODktMzMxYi00YWJmLWFhMGYtY2NmYTk2YjJjZTkw/view?resourcekey=0-7F-yW80tmQTCKtzqlGKzYA

[GMJCZP]

Give a try to this URL:

https://docs.google.com/file/d/0B0rPtrLUrGhIYmM1MDNhODktMzMxYi00YWJmLWFhMGYtY2NmYTk2YjJjZTkw/view?resourcekey=0-7F-yW80tmQTCKtzqlGKzYA

That is the version of 2008, tebasuna upload today the version 6 by Sendspace here (https://forum.doom9.org/showthread.php?p=1522330#post1522330).

By the way, it seems to be 64 bit only and I need the 32 bit one.

Edit: problem resolved, here (https://forum.doom9.org/showthread.php?p=1944711#post1944711).