View Full Version : 6ch wav to 2ch wav - channels messed?
rahzel
17th August 2006, 08:02
i have a mp4 h.264 video, with AAC 5.1 audio. i've looked around, and i've tried about 3-5 methods to convert it to 6ch AC3 with no luck, so i decided to settle for 2ch stereo.
i extracted the aac with yamb, converted it to 6ch wav with aacdrop, then tried to convert that to 2ch mp2/ac3 with ffmpeggui with no luck; i get some kind of error.
i then tried using foobar to convert it to 2ch wav, and it worked, however, the voice is only coming from my left speaker, so i'm guessing the channels got messed up.
is there anyway i can correct this, or is there a better way to convert 6ch wav to 2ch?
tebasuna51
17th August 2006, 13:04
i have a mp4 h.264 video, with AAC 5.1 audio. i've looked around, and i've tried about 3-5 methods to convert it to 6ch AC3 with no luck
To transcode aac 5.1 -> ac3 5.1 use faad (http://www.rarewares.org/files/aac/faad.zip) (output wav int 16 bit WAVE_FORMAT_EXTENSIBLE and can be > 4 GB), and aften (http://forum.doom9.org/showthread.php?p=847746#post847746) (not need remapping, input can be > 4 GB WAVE_FORMAT_EXTENSIBLE)
converted it to 6ch wav with aacdrop,
Use faad instead aacdrop.
then tried to convert that to 2ch mp2/ac3 with ffmpeggui with no luck; i get some kind of error.
i then tried using foobar to convert it to 2ch wav, and it worked, however, the voice is only coming from my left speaker, so i'm guessing the channels got messed up.
You need a downmix (5.1 -> 2 chan) method. There are the ATSurround DSP plugin for foobar and the DSP "Convert 5.1 to stereo", but aren't Dolby Prologic compatibles.
If you want DPL downmix you can use BeHappy (open wav with BassAudio, apply DPL, Normalize and Encode to mp2/ac3) or WaveWizard.
Before use WaveWizard you need add a new ChannelMapping, open mappings.ini (at WaveWizard folder) with Notepad and add this 2 lines:
6->2#mix#inactive#Dolby Prologic II downmix
0.325 0.000 0.000 0.325 0.230 0.230 0.000 0.000 -0.282 0.163 -0.163 0.282
Then open the wav with WaveWizard, select Dolby Prologic II downmix, select Normalize at 100% and encode to mp2 (ac3enc with WaveWizard is not recommended).
rahzel
17th August 2006, 20:56
err, i've come across a few command line tools, and i have no idea how to use them. it opens a dos window, then just closes.
is there any reason why to use faad instead of aacdrop? to my understanding, aacdrop is a little gui for faad is it not? also, aacdrop did the job, or did it do something wrong?
ill give your methods a shot, thanks.
rahzel
18th August 2006, 08:34
behappy also didnt work... the voice is also coming from the left speaker.
i tried wavewizard, and the same thing happened, but it looks to me that wavewizard can remap the channels, but i have no idea what to do.
this is what i see... can anyone help me figure out what channels i should use for what?
rahzel
18th August 2006, 09:10
i plugged in 0.230 in ch 1 destination, set the rest to 0.00, and it seems to have worked. the voice is coming from both left and right. however, i wonder if its even stereo, or if the same channel is going to both speakers?.... it should be right shouldnt it? because the 2 ch's are different.
tebasuna51
18th August 2006, 13:21
behappy also didnt work... the voice is also coming from the left speaker.
i tried wavewizard, and the same thing happened,
Maybe your source are bad mapped. The original aac play ok?
but it looks to me that wavewizard can remap the channels, but i have no idea what to do.
this is what i see... can anyone help me figure out what channels i should use for what?
I can't see your image. Maybe something like:
Destination
ch 0 (Lt) ch 1 (Rt)
------ ------
Source ch 0 0.325 L
ch 1 0.325 R
ch 2 0.230 0.230 C
ch 3 LFE
ch 4 -0.282 0.163 SL
ch 5 -0.163 0.282 SR
Dolby Prologic II downmix
Or:
Lt = 0.325*L + 0.230*C - 0.282*SL - 0.163*SR
Rt = 0.325*R + 0.230*C + 0.163*SL + 0.282*SR
Dialogs are (normally) in C channel then after the downmix must be presents at Lt and Rt.
rahzel
18th August 2006, 18:46
yeah, thats almost exactly what i see, except ch 5 is 0.16, not 0.163.
even though im downmixing to dpl, do i still need to fill in all the channels?
also, how am i supposed to know whats what, and what those numbers mean? (ie, 0.230)
edit: another problem, the audio that i thought worked, there are gaps every 20-30 minutes or so where theres no sound for like a minute. im guessing its because i removed some of the channels.
tebasuna51
19th August 2006, 00:55
The numbers are the mix coefficients:
The new Right channel is the 32.5% of source Right channel plus the 23% of source Center channel and so on.
This is a tool to convert 6ch wav to 2ch at your convenience, but before you must know if your source channels are correct.
Maybe if you can send a source sample...
rahzel
19th August 2006, 01:16
what would be the best way to take a small segment from a 6ch wav file?
BigDid
19th August 2006, 02:09
what would be the best way to take a small segment from a 6ch wav file?
Hi,
Audacity?
http://audacity.sourceforge.net/about/features
Did
tebasuna51
19th August 2006, 11:45
Better use your aac source with BeSplit:
BeSplit -core( -input source.aac -prefix sample -type aac -a ) -split( 5 11 )
With this command line (in a folder with BeSplit and source.aac) you obtain a sample01.aac with 6 seconds (from second 5 to 11).
rahzel
19th August 2006, 21:32
ok, heres a sample. if you can't download it, get it here:
http://rapidshare.de/files/30022969/sample01.aac.html
the start stutters a bit for some reason, but i think it should do.
tebasuna51
20th August 2006, 03:26
Your sample is a quiet fragment without dialogs.
The max peaks are:
L -23 dB, R -24 dB, LFE -75 dB, C -32 dB, SL -36 dB, SR -32 dB
By volume seems correct mapped.
Maybe if you can select a fragment with the problematic dialogs.
rahzel
20th August 2006, 04:29
instead of this, can you explain how to convert it to 6ch ac3?
i've managed to get 6 mono wav files out of the 6ch wav file using besplit, and the total size is the same as the 6ch wav file (both 2.55gb).
i need to find out how to convert it to 6ch ac3 now. ill also need to change the sample rate to 4800 (its at 2400 now i believe).
tebasuna51
20th August 2006, 10:39
i need to find out how to convert it to 6ch ac3 now. ill also need to change the sample rate to 4800 (its at 2400 now i believe).
This is the aacdrop problem, low bitrate aac are converted to half frequency. Use faad instead, the sample01.acc is converted to 48 KHz by faad.
i've managed to get 6 mono wav files out of the 6ch wav file using besplit, and the total size is the same as the 6ch wav file (both 2.55gb).
If your wav is 24 KHz after decoded by faad must be > 5 GB, you need a NTFS partition with enough free space.
instead of this, can you explain how to convert it to 6ch ac3?
Put faad.exe, aften.exe and your source.aac at same folder and with Notepad make a new encode.bat file with this content:
faad source.aac
aften -b 448 source.wav output.ac3
pause
Or -b 384 if you want low ac3 bitrate.
Execute encode.bat and wait (be patient, faad can stop the % but continue working) until "Press a key to continue...".
rahzel
20th August 2006, 11:45
This is the aacdrop problem, low bitrate aac are converted to half frequency. Use faad instead, the sample01.acc is converted to 48 KHz by faad.
If your wav is 24 KHz after decoded by faad must be > 5 GB, you need a NTFS partition with enough free space.
i see... maybe ill just try what you said below.
Put faad.exe, aften.exe and your source.aac at same folder and with Notepad make a new encode.bat file with this content:
faad source.aac
aften -b 448 source.wav output.ac3
pause
Or -b 384 if you want low ac3 bitrate.
Execute encode.bat and wait (be patient, faad can stop the % but continue working) until "Press a key to continue...".
trying this now... its at 7%... man its slow.
by "faad source.aac", you want me to enter the source file name right? ie, if the source audio was audio.aac, i would put "faad audio.aac"?
btw, i REALLY appreciate your help... thanks.
rahzel
20th August 2006, 12:09
ok, something wrong happened... it stays at 7% for like 10 minutes, then it closes. the wav file is 5.11gb, but when i try to play the file, it says "the file is damaged". i also tried opening it in aften/faad, but it wont open.
i also renamed the aac file to source.aac to make sure.
tebasuna51
20th August 2006, 12:35
ok, something wrong happened... it stays at 7% for like 10 minutes, then it closes. the wav file is 5.11gb, but when i try to play the file, it says "the file is damaged".
Of course, correct wav only can be < 4 GB, but not problem for encoders/programs than accept this "damaged" wav. Your 5.11 GB wav is valid for Aften (at least with version I test (http://forum.doom9.org/showthread.php?p=850754#post850754))
i also tried opening it in aften/faad, but it wont open.
i also renamed the aac file to source.aac to make sure.
Faad make a wav with the same name than the aac, but the name is not problem, if your 5.11 GB is audio.wav then use a encode.bat like this:
aften -b 448 audio.wav output.ac3
pause
And say me if there are any error.
rahzel
20th August 2006, 19:45
sweet, i finally got it to work!
first of all, this is kind of irrelevent, but it's kind of bugging me i don't know if this is normal, but if i use tools like faad or aften, where you have to open them in a dos window, when i double click them, they just open and close right away. i have to go start -> run -> cmd (to open a dos window) drag the application file into dos, then type the command line to do the encoding... is this normal?
anyway, i manually typed "aften -b 448 (dragged source file into dos window) output.ac3"; i did not type "pause" like you asked, because i forgot. i played the file, and it seems like it was fine. should i re-encode the file, and add the "pause" like you asked me to? what does that do?
again, thanks a lot, tebasuna51. i've been trying to get this to work for days now (not constantly working, but you know...).
tebasuna51
20th August 2006, 20:11
anyway, i manually typed "aften -b 448 (dragged source file into dos window) output.ac3"; i did not type "pause" like you asked, because i forgot. i played the file, and it seems like it was fine. should i re-encode the file, and add the "pause" like you asked me to? what does that do?
Double click over a .bat file is open a cmd window and execute the commands like you type them.
Then 'pause' is only a pause before close the window and you can see any error message.
if i use tools like faad or aften, where you have to open them in a dos window, when i double click them, they just open and close right away
When this tools are executed without parameters send error message or help to correct usage, but the window is closed immediately and you can see nothing.
Use a .bat like:
faad
pause
And you can se the usage and after modify with the required parameters.
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