GUIDE: Converting stereo to 5.1 surround for FREE

[ursamtl]

Actually there are quite a few ways around. Some work better than others and some are more complicated than others. The results vary.

Anyway, if all goes well, I hope to have the first working Beta version of my 1-step, non-VST version of V.I available by the end of the this weekend. I just have some last-minute testing to do and a bit of error trapping to add. It won't be fancy, but it will work!

Regards,
Steve.

[Betchin]

Im not very good in all this encode stuff. I can't find out what is the easiest way to convert a simple mp3 to surround or 5.1

Is there a simple way?

[ursamtl]

If you check out the various threads on here, I'm sure you can find lots of instructions for encoding. Depends on what you want? Do you want to make DVDs with surround audio or surround CDs? Or perhaps you just want surround files you can play back on your PC. Most ways are actually a lot simpler than they may seem once you get the right procedure.

Whenever possible, I recommend not using an MP3 as a source. MP3 is a "lossy" compression format meaning that some of the original sound information was during the conversion to MP3. This is often the high frequency content that gives the audio a sense of "air" or the dynamic range. MP3 encoding is far better now than it was a few years ago, but it depends on a lot of factors. Even for supposed "CD quality" MP3s, some people can hear the difference.

If you have no other source then obviously, it's better than nothing but if you do have access to wave files or lossless sources (flac, ape or shn files) the end result will be better.

[raquete]

hi ursamtl
how i load one group of files(10 or more flfr,clfe and slsr)to encode with wavewizard like "batch file"?

thanks for one more great guide. :cool:

[ursamtl]

To be honest, I don't know. I've never tried batch work with ww. Perhaps you could post your question in the WaveWizard thread (http://forum.doom9.org/showthread.php?p=662555#post662555) or send a message to johnman. I'm at work right now so I can't do any testing on it.

Sorry,
Steve.

[ursamtl]

Just a note to let Plogue fans know that a new time-limited demo has just been released. This one works until Sept. 16, 2006. For those who purchased the early bird licence, this is simply version 0.92 and will work with your licence indefinitely.

[ursamtl]

Just thought I'd let people know that I updated the guide slightly to link to a new Foobar Bridge to Winamp plugin that's compatible with Foobar v9.x. You can find it at http://pelit.koillismaa.fi/plugins/redir.php?id=682. I'm very busy right now but as soon as I have time, I'll update the guide pictures and procedures. I also have some other tips for producing the most professional sounding 5.1 possible all with completely free and legal software.

Regards,
Steve.

[raquete]

also have some other tips for producing the most professional sounding 5.1...
most professional? :cool:

thank you Steve.

[Elektra999]

Hello Steve.

I hope anxious, your new guides and programs je je

:D :D :D

[Elektra999]

Je has to me very intrigued Steve je je :)

[ursamtl]

Hello Steve.

I hope anxious, your new guides and programs je je

:D :D :D

Not anything that new, just some tips I intend to write for getting the most professional sound possible from freeware tools. There are some things already in the guides, such as using 32 bit files, removing DC offset, using a limiter to avoid clipping, etc. As I read and study the whole field of sound production, I pick up things, so I hope to write some new stuff and update the guides.

[DSP8000]

Hi Steve,

glad to see that you're improving/updating your guide.
Just one suggestion, can you please in your updated version of the guide include vlevel for foobar2000. I find it very useful.
Also, of course Aften, free ac3 encoder that produces excellent output.
Keep up the good work.

Regards,
DSP8000

[ursamtl]

Thanks for the suggestion. I'll check it out. As far as I can tell from a quick Google, it's a compression plugin.

Given the whole loudness war, I usually stay away from compression except for limiting to prevent clipping. Compression can work wonders for a vocal or an individual instrument, but on a whole mix it can crush the life out of the music!

But, there are exceptions. Can you adjust the two channel individually or only as a pair? An individual per channel adjustment would be great for boosting the low LFE as I suggest with the Plogue Bidule layout and Classic Limiter, or for ensuring dialog clarity in a movie soundtrack by compressing the C. Unfortunately, the freeware method in this guide groups the C and LFE channels together so either a plugin with individual channel adjustments or doing separate passes for each would be necessary.

In any case, I'll check it out. Thanks.

Regards,
Steve.

[NorthPole]

@ursamtl

Have you looked at the mp3 surround upmix here http://www.all4mp3.com/info/mp3sx.html

I was wondering what you thought?

There is a command line utility located here http://www.all4mp3.com/tools/sw_fhg_demo.html

Thanks

[ursamtl]

Thanks for the tip. I'll check it out. As a rule, I'm not a big fan of MP3s other than as a convenient compression format for my portable player. For surround work, I always try to stick with lossless source files. I can hear the difference in clarity between MP3s and lossless even at higher bit rates. It's more a matter of dynamics in the highs than anything else. Cymbals, etc., just don't breathe as well. They sound constrained and squashed.

I also know in my tests when developing V.I that mp3s often do not convert well because the psychoacoustic tricks they use for compression get unmasked when one attempts to generate surrounds for example. I remember a couple of cases where I could hear a constant digital sample and hold sound in the surrounds. It wasn't in the 2-channel source but it was really apparent in the surrounds. It sounded like that phased, swooshing sound one hear when someone tries too hard with one of those spectral center extraction plugins, only worse!

Anyway, I'll check it out. The software is limited to Dec. 2006 and there doesn't seem to be any mention of what, if anything, will be available after that.

[raquete]

As a rule, I'm not a big fan of MP3s other than as a convenient compression format for my portable player. For surround work, I always try to stick with lossless source files. I can hear the difference in clarity between MP3s and lossless even at higher bit rates. It's more a matter of dynamics in the highs than anything else. Cymbals, etc., just don't breathe as well. They sound constrained and squashed.
:eek: :thanks: :cool:
my truly but dangerous comments about mp3 (fans excuse me) :
this is the first time that i read the true about mp3 in the internet(in thousands forums and places).
is exactly how i think and feel hearing... "it". (i don't like of ilusions)

your comments deserve a sticky in the top of the audio forum! :D

[NorthPole]

@ursamtl and @raquete

Thank, I didn't catch the software limit date.

Just looking for different options on upmixing movie soundtracks. Currently captured in stereo mp2 mode.

Unfortunately, I take a quality hit on the capture and decode from mp2 to wav and then I upmix from there.

Haven't struck on anything yet but will keep looking for better methods.

I like your vst plugins very much but I don't have plogue. The 3 step free method with foobar works good but is somewhat time consuming.
Btw, for AC3, if you use the aften encoder instead of ffmpeg or soft encode, you don't have to use wave wizard to remap the channels. However, you still have the issue of 3 different files to merge.
Still, very good results in the end.

[tebasuna51]

Btw, for AC3, if you use the aften encoder instead of ffmpeg or soft encode, you don't have to use wave wizard to remap the channels. However, you still have the issue of 3 different files to merge.
With new AviSynth v2.5.7 RC-1 (http://forum.doom9.org/showthread.php?t=116044) the WavSource() accept 32 bit Float and the wav output can be 32 bit Float, then the best method to ac3 encode may be with a Merge32.avs:
global OPT_AllowFloatAudio=True
f = WavSource("D:\YourPath\x_fLfR.wav")
c = WavSource("D:\YourPath\x_CLFE.wav")
s = WavSource("D:\YourPath\x_slsr.wav")
MergeChannels(f, c, s)
And after:
bepipe --script "Import(^D:\YourPath\Merge32.avs^)" | aften.exe -b 448 - x.ac3

[raquete]

another great post (or fast tutor) tebasuna, is bookmarked!
:)

thanks!

[NorthPole]

@tebasuna51

Good idea...Thanks.

[NorthPole]

@ursamtl

Do you know of any vst host programs that can create the 3 different stereo wave files from a command line in a batch file?

I was thinking about trying to generate the 3 different wave files with your plugins within a batch file and then pipe it into an encoder.

[ursamtl]

Unfortunately, to my knowledge, there are no command line hosts. I had a similar idea a few months ago and did a search around the internet. There are various commandline programs for merging or splitting 3 stereo waves once you have them.

[NorthPole]

@ursamtl

Thanks for the reply...I kind of thought it was a long shot.

[ursamtl]

Yeah it's too bad. I thought I read somewhere something about AVISynth supporting VST plugins and thought that would be fantastic but I couldn't find anything else out.

[newhaven]

hi,
is there a way to get more bass out of each channel using this method, or even just the lfe channel?

any ointers or lessons are appreciated-----newhaven

[ursamtl]

You can always use EQ, but one of the best ways to increase deep bass is to compress the bass. If you can find a free multiband compressor, try compressing just the bass frequencies. As with all effects, do it subtly. A little bit goes a long way. Then leave it and come back the next day to see if it still sounds good. Ears tend to become less sensitive as we work on audio on a given day and we all tend to boost or cut things more than necessary. Then the next day, what sounded good one day sounds bad the next.

[raquete]

one of the best ways to increase deep bass is to compress the bass. If you can find a free multiband compressor, try compressing just the bass frequencies.Steve,
seems very interesting but i don't understood how to do and basses are my "beach"...cool basses=cool sound,i like so much.
could you explain me in simple words? (if possible)

[newhaven]

steve,
when you are compressing the bass frequencies, at which point in the process that started this thread , are you doing so? also can you recommend a multiband compressor, free or not?
thanx --newhaven

[ursamtl]

Here are some free mulitband compressors.

http://www.kvraudio.com/get/1413.html
http://www.kvraudio.com/get/1360.html
http://www.kvraudio.com/get/1047.html
http://www.kvraudio.com/get/950.html
http://www.kvraudio.com/get/2000.html

You could try compressing the bass in the original stereo source file, or else the front L and R. You could try splitting the CLFE file and compressing the LFE channel only. This will give you a bit of really deep bass. If I had more time, I'd write a guide, but unfortunately, I'm very busy with my work right now.

Regards,
Steve.

[newhaven]

steve,
thanx for the pointers--will play on my own.

newhaven

[old-hack]

Steve,

I've run through this thread top to bottom. Can you confirm that the settings in your guide (at the beginning of this thread) for using wavewizard and conversionbatcher contain all of the correct steps & settings?

I'm trying to batch process my Bidule mono files using wavewizard and something goes funny. I don't want or need to convert anything - just want wavewizard and/or conversionbatcher to process the 6 mono files to Surcode.

Thx.

[old-hack]

Here is exactly what I'm doing.

1. Create 6 mono files using Plogue Bidule.
2. Add them to Wavewizard.
3. Created a correct channel mapping for Surcode.
4. Put checkmark in new Surcode Channel Map.
5. Put checkmark in "Enable channelmapping".
6. Under Conversion Batcher menu option select Surcode.
7. Configure Surcode for 6 channels & assign correct coding.
8. Put checkmarks in Send jobs to batcher & Start batcher.
9. I have no idea what to put in Preferences, so I tried
a. Put checkmark in Enable channel mapping.
b. Put a checkmark in Stream manipulation & select Merge
c. Output format = Wave PCM.
d. No other checkmarks.

Here's the error message I'm getting

"CB error on the merged/stitched output. The configuration of conversion batcher requires 6 mono streams."

If I use ac3dll it works fine and processes all 6 mono files into a merged wave and ac3 file.

What am I doing wrong?

Thx.

[tebasuna51]

What am I doing wrong?

Repeat the same post on three threads. See http://forum.doom9.org/showthread.php?p=912603#post912603

[Chainmax]

UrsaMtl, do you think it would be possible to update the guide for use with FooBar2K v0.9.x and replacing AC3Encoder.dll with AftenGUI?

[ursamtl]

Sorry I haven't had time to do this yet. My work has been very hectic over the past few months. I'll try as soon as I can. In the meantime, if you follow the guide as is, it should work for you.

[Chainmax]

Oh, there's no rush at all. I once managed to to it right, but a second try yielded no results. I can always go back to v0.8.3 though, so it's ok :).

[ursamtl]

Don't give up on 9.x. The method for surround works very much the same way as with 0.8.3.. You just have to account for the couple of changes to the Preferences dialog box.

[Chainmax]

Just because Nero Wave Editor can only save to 8bit or 16bit doesn't mean it can't read 32bit files properly, right? Because if that's the case, then I think the reason why the stereo enhancement didn't work is because apparently Wider Boy Pro can't output 32bit files, as setting every output option to 16bit in Foobar seems to have solved the issue.

[tebasuna51]

EDIT: Read also this new method (http://forum.doom9.org/showthread.php?p=1011274#post1011274)

Really there are changes with Foobar2000 v0.9.4.2 and plugins, only work with 16/24 bits int, never with 32. The method can be still valid because 24 bit is enough for Dynamic Ac3 Range.

Also we can use Aften to encode to ac3 from Foobar and AviSynth to merge the 3 stereo files. Let me propose the method:

a) INSTALL
- Foobar2000 v0.9.4.2 (http://www.foobar2000.org/)
- V.I Suite version 1.11 Installer (http://www.stevethomson.ca/vi/) (Remember the installation folder, C2)
- VST Host Winamp Bridge (http://www.savioursofsoul.de/Christian/VST/dsp_vst.exe) by Christian Budde (Remember the installation folder, C3)
- Bridge plugin for winamp DSP plugin (http://pelit.koillismaa.fi/plugins/redir.php?id=682). Uncompress the .dll in ...\foobar2000\components folder.

And also we need

- AviSynth v2.5.7 (http://sourceforge.net/project/showfiles.php?group_id=57023&package_id=72557)
- foo_input_avs (http://workspaces.gotdotnet.com/behappy) Uncompress the .dll in ...\foobar2000\components folder. Needed to open avs in Foobar.
- Aften (http://win32builds.sourceforge.net/aften/index.html) Ac3 free encoder. Uncompress Aften.exe in any folder, for instance in G:\St_Surr, we can use like working folder (C1).

b) CONFIGURE
- We need also a Mezcla.avs in the working folder with this content (use Copy and Paste in Notepad for instance):
f = WavSource("G:\St_Surr\flfr.wav")
c = WavSource("G:\St_Surr\clfe.wav")
s = WavSource("G:\St_Surr\slsr.wav")
MergeChannels(f,c,s)

- Foobar2000, firts we verify in File -> Preferences -> Components the plugins Winamp DSP Bridge and AVS input:

http://img237.imageshack.us/img237/2992/vifoocompop4.png (http://imageshack.us)

After we need configure Aften in Foobar, File -> Preferences -> Converter -> Add New, and fill this screen:

http://img248.imageshack.us/img248/4668/vifooaftenpl1.png (http://imageshack.us)

We see 'AC3 (AFTEN)' in the list 'Encoding Presets', also we can set 'Preferred bith depth' to 24 (max. supported).

http://img338.imageshack.us/img338/9651/vifoodepthzt2.png (http://imageshack.us)

c) PROCESS
We open, for instance, Track03 in Foobar and:

http://img114.imageshack.us/img114/8826/vifooconvap7.png (http://imageshack.us)

1) Mouse right click and Convert -> Convert to... to open 'Convert Setup'
2) In 'Convert Setup' select 'Encoding Presets' WAV, check 'DSP Processing' and click in (...) to open 'DSP Settings - Converter'
3) We put and select 'Winamp DSP Bridge' in 'Active DSP' list (also we can use before a Resampler or/and a Limiter like last DSP), click in 'Configure Selected'
4) First time we need fill 'Winamp plug-in path' with (C3), check 24 bit like 'Fixed point conversion parameters', select 'VST Host DSP v1.0 for Winamp' and click in 'Show interface window'
5) Here we need load the 3 dll's, like is explained in first post.
6) Also the V.I settings is explained already.
7) We need create the 3 wav stereo files in the folder and with the names established in Mezcla.avs.
8) Last pass is encode to ac3, we need only load Mezcla.avs in Foobar and convert:

http://img114.imageshack.us/img114/9209/vifootoac3dd1.png (http://imageshack.us)
Now we select 'AC3 (AFTEN)' in 'Encoding Presets' and uncheck 'DSP Processing' not needed now. Ok and select the final ac3 file name.

EDIT: Sorry, changed AviSynth 2.5\Plugins folder with the correct: foobar2000\components folder.

[ursamtl]

Tebasuna,

Excellent addition! Thanks for your work on this. I'll edit the opening message of the thread to mention this update.

Regards,
Steve.

[Chainmax]

Excellent addition indeed, muchas gracias tebasuna51 :) http://smilies.vidahost.com/otn/wink/thumb.gif.

Does anyone kow why does 32bit not work anymore? I want to use foobar with the Wider Boy Pro VST plugin to create a stereo file out of a mono one and needed 32bit throughout the whole process as the source file is very crappy and I want the absolute best possible quality output.

[tebasuna51]

There are a new DSP component for Foobar2000 (foo_dsp_vst.dll) to load VST plugins.
Is in beta stage but can simplify very much the method described here to obtain the 5.1 upmix, because can work directly with VI.dll instead with the stereo ones.

Now we need:

a) INSTALL
- Foobar2000 v0.9.4.3 (http://www.foobar2000.org/)
- V.I Suite version 1.11 Installer (http://www.stevethomson.ca/vi/) (Remember the installation folder, C2) (Temporal link (https://www.sendspace.com/file/bunpiq))
- foo_channel_mixer.7z (for 0.9). (http://www.skipyrich.com/foobar/foo_channel_mixer.ptml) Uncompress the .dll in ...\Foobar2000\components folder.
- VST host (foo_dsp_vst) (http://pelit.koillismaa.fi/plugins/redir.php?id=864). Uncompress the .dll in ...\Foobar2000\components folder.
- Aften (http://win32builds.sourceforge.net/aften/index.html) Ac3 free encoder. Uncompress Aften.exe in any folder, for instance in G:\St_Surr (C1).

b) CONFIGURE
- Foobar2000, firts we verify in File -> Preferences -> Components the VST Bridge and Channel Mixer.
After we need configure Aften in Foobar, File -> Preferences -> Converter -> Add New like in precedent post. (http://forum.doom9.org/showthread.php?p=991707#post991707)
EDIT: The 'Preferred bit depth' can be 32.

c) PROCESS
We open a stereo file in any format supported by Foobar and:

http://img230.imageshack.us/img230/1927/st51foovi1zq5.gif (http://imageshack.us)

1) Mouse right click and Convert -> Convert to... to open 'Convert Setup'
2) In 'Convert Setup' select 'Encoding Presets' AC3 (AFTEN) directly, check 'DSP Processing' and click in (...) to open 'DSP Settings - Converter'
3) First DSP to Active and Configure is 'Channel Mixer': General -> Output Channels 6, Upmix -> Mode Off. With these settings we convert the stereo in multichannel with FL-FR copied and the rest empty.

http://img217.imageshack.us/img217/9408/st51foovi2zo5.gif (http://imageshack.us)

4) Now we put and select 'VST Bridge' in 'Active DSP' list (also we can use before a Resampler or/and a Limiter like last DSP), click in 'Configure Selected'
5) First time we need fill 'Select VST Library' with (C2) and select the VI.dll plugin.
6) The V.I settings are explained already. V.I fill the empty channels and modify FL-FR
7) The ac3 file is generated directly.

EDIT:
- The 'Preferred bit depth' can be 32, and is the recommended setting to avoid conversions (at least Aften convert anything to 32 float to work internally).
- No problem with file sizes because don't exist intermediate wav files.

[ursamtl]

Thanks tebasuna! You're really coming up with some nice finds! I'll edit the guide to point to this post.

Regards,
Steve.

[Pookie]

I've been using the new foo_vst_dsp plugin for a month or so. Works great! You might see an error message the first time you load it as it looks for the location of the VST plugins. But from then on, no crashes, no problems. :)

[capricorn]

Hi Tebasuna51-

Thanks for your guide. However, I keep getting an write error that will not allow me to output the ac3 file.

I have followed your most recent guide (three posts up) and tried a different aften.exe compilation each time with no luck.

Any suggestions?

Regards,

cap

[tebasuna51]

However, I keep getting an write error that will not allow me to output the ac3 file.

I have followed your most recent guide (three posts up) and tried a different aften.exe compilation each time with no luck.

Can you obtain a wav 6 channel (selecting WAV in 'Encoding Presets')?

Can you convert a wav 6 channel to ac3 with Foobar and Aften?

[capricorn]

Hi-

I can create the 6 channel wavs from the encoder presets. However, I am unable to merge them together with the AVS file into an AC3 with Aften through foobar.

I tried going through all the settings and I'm still unable to work it through.

I also tried to play the three resulting wav files but no sound is output. That is a bit odd to me as well. Should I be able to hear something from those files?

cap

[tebasuna51]

I can create the 6 channel wavs from the encoder presets. However, I am unable to merge them together with the AVS file into an AC3 with Aften through foobar.
Use the method in post #92 instead the #89, really more easy.
You don't need avs file or three stereo wav files.

To convert anything (stereo or multichannel source accepted by Foobar) to ac3 you need configure Aften before with File -> Preferences -> Converter -> Add New, and fill the screen (see figure in post 89) Commandline Encoder Settings - Editing Preset:
Encoder: G:\your_path\aften.exe (Drive and path where you have the encoder)
Extension: ac3
Parameters: -v 0 -pad 0 -readtoeof 1 -b 448 - %d (and more parameters if you want)

Format is: lossy
Higest BPS mode supported: 32

Encoder Name: AC3 (AFTEN)
...
Try encoding something without DSP until work.

I also tried to play the three resulting wav files but no sound is output. That is a bit odd to me as well. Should I be able to hear something from those files?
This is a second problem. Of course you need hear the generated channels.

Use the method in post #92 configuring the two DSP functions and must work (at least for me work).

[capricorn]

Hi Tebasuna51-

I'll try the method in post #92. I've tried reconfiguring the aften.exe encoder by copying and pasting the parameters that you have and I get the following error message upon attempting to encode:

Error writing to file (Encoder has terminated prematurely with code 1; please re-check parameters) : file://C:\Documents and Settings\CAL\Desktop\audio\test.ac3

I have ensured that the aften.exe is correctly identified with the path. Also, the DSP was turned off.

Any suggestions?

TIA

cap

[tebasuna51]

I'll try the method in post #92. I've tried reconfiguring the aften.exe encoder by copying and pasting the parameters that you have and I get the following error message upon attempting to encode:

Error writing to file (Encoder has terminated prematurely with code 1; please re-check parameters) : file://C:\Documents and Settings\CAL\Desktop\audio\test.ac3

I have ensured that the aften.exe is correctly identified with the path. Also, the DSP was turned off.

Any suggestions?
I get the same error message when use a incorrect parameter line, for instance whith:
-v 0 -pad 0 -readtoe 1 -b 192 - %d
I get:
Error writing to file (Encoder has terminated prematurely with code 1; please re-check parameters) : file://C:\Documents and Settings\Toni\Mis documentos\stereoe.ac3

For test purpose you can let all defaults and use the more simple line:
- %d
Where '-' instruct Aften use the STDIN (Standard Input, pipe)
and '%d' is replaced by Foobar with the name of ouput ac3 file supplied in the next dialog box.

Maybe you use a old aften version without support for -pad or -readtoeof parameters.