GUIDE: Converting stereo to 5.1 surround for FREE

[ursamtl]

NOTE: THIS GUIDE WAS WRITTEN IN JANUARY 2006. RECENTLY, A NEW VST PLUGIN ADD-ON FOR FOOBAR2000 HAS APPEARED THAT GREATLY SIMPLIFIES THE PROCESS. IF YOU'D LIKE TO TRY IT, CHECK OUT THE FOLLOWING POST. FOR BACKGROUND AND OTHER APPROACHES, YOU CAN ALSO READ THE ORIGINAL GUIDE BELOW.

2007 GUIDE BY TEBASUNA51 (http://forum.doom9.org/showthread.php?p=1011274#post1011274)
2013 GUIDE (http://forum.doom9.org/showthread.php?p=1640451#post1640451)

Many of the stereo-to-surround conversion guides posted in this forum and elsewhere on the internet rely on Plogue Bidule or other VST hosting software such as Steinberg Cubase, Steinberg Nuendo, or Adobe Audition. These are all fine programs but they are not free. Bidule has a time-limited free version but eventually that will run out.
If you can afford Plogue's $75 US for an early bird license, let me say that in my opinion, this is the best value available. You get tremendous value for your money, one of the most interesting, unique and powerful audio programs available today. If you can't afford this, however, there is a way to convert stereo to 5.1 sound and do it for free, using legal software.

There may be other approaches, but this is one I'm proposing.

Here's what you need
V.I Stereo to 5.1 Converter Suite (http://stevethomson.ca/audio/guides/VI_Setup.exe)
Foobar 2000 version 9.0. (http://www.foobar2000.org)
Bridge plugin for winamp DSP plugin (http://pelit.koillismaa.fi/plugins/redir.php?id=682) NEW FOR FOOBAR v0.9x
VST Host Winamp Bridge by Christian Budde (http://www.savioursofsoul.de/Christian/VST/dsp_vst.exe)
MultiFXVST (Optional-for chaining VST effects) French site (http://www.ctaf.free.fr/dokuwiki/doku.php?id=meslogiciels) or Google translation of site (http://translate.google.com/translate?u=http%3A%2F%2Fwww.ctaf.free.fr%2Fdokuwiki%2Fdoku.php%3Fid%3Dmeslogiciels&langpair=fr%7Cen&hl=en&ie=UTF-8&oe=UTF-8&prev=%2Flanguage_tools)
johnman's wavewizard (http://tinyurl.com/d7lqg)

NOTE: THE FOLLOWING GUIDE WAS WRITTEN USING FOOBAR V0.83. THE BASIC PROCEDURE WITH VERSION 0.9 IS ALMOST THE SAME BUT WITH SOME CHANGES. FOR THE BASIC METHOD OF USING THE V.I PLUGINS, READ THE GUIDE BELOW. FOR THE FOOBAR2000 V.9.4 AND HIGHER PROCEDURES, INCLUDING AC3 ENCODING WITH AFTEN, SEE THIS POST (http://forum.doom9.org/showthread.php?p=991707#post991707).

As for why these programs and plugins: Foobar allows a user to right-click and save a 32-bit file to disk as a wave file without having to play it back in real time. Similar to Plogue Bidule's offline mode, this can save a lot of time over VST hosts that can only process while playing a file. Foobar also has built in dithering, limiting, resampling and convolution. Why not Winamp? Because to my knowledge, its diskwriter plugin only saves 16-bit files. If you prefer to use it, go ahead!
Foobar does not support VST plugins, nor does it support Winamp plugins. However, if you add the Bridge plugin for winamp DSP plugin and then in turn use it to load Christian Budde's VST Host Winamp Bridge, you can then host a VST plugin.
If you want to chain more than one VST plugin, you can load MultiFXVST into the VST Host Winamp Bridge. It works! However, for limiting, reverb convolution (impulse responses), resampling and dithering, you can use Foobar itself. Just be sure your settings for writing files have the Use DSP box checked.

Procedure Overview
You will process a stereo file three times, each time using a different V.I companion plugin from the fLfR, CLFE, and sLsR plugins to write processed files to disk. This will create three stereo files. Then you can use wavewizard to create 6 mono files or one 6-channel file for encoding. If you need a freeware 5.1 encoder, wavewizard comes with a free AC3 encoder.

Part 1: Foobar and plugins
Install Foobar. Install all files for both Bridge plugins in Foobar's Component subdirectory. This includes all files such as the msvcr71.dll that comes with the Foobar2WinampPlugin. Important: The installation program for the Winamp VST bridge installs to the Winamp/plugins folder. If you don't have Winamp, let it create and install to that folder anyway and then manually move the dsp_vst.dll file to your Foobar/Component directory.
You can check to be sure the bridge is installed by starting Foobar and pressing Ctrl+P. In the Preferences dialog box,
select Component libraries at the very top of the left tree and find "foo_dsp_Foobar2WinampPlugin.dll."
http://stevethomson.ca/audio/guides/foo-1.gif

Expand the Components branch of the tree on the left and select Diskwriter. Setup Foobar to write 32-bit files by choosing 32 from the Preferred bit depth dropdown list and then check the box next to use DSP in the Processing[b] section.
http://stevethomson.ca/audio/guides/foo-2.gif

Select [b]DSP Manager in the tree on the left. Use the <= and => buttons on the right to move DSPs from the Available DSPs list on the right to the Active DSPs list in the middle. Be sure that the Advanced Limiter DSP is at the very bottom (use the Up and Down buttons to re-arrange the list order.
http://stevethomson.ca/audio/guides/foo-3.gif

Expand the DSP Manager branch of the tree and select Winamp DSP plugin. Select the VST Host DSP v.10 for WinAmp in the middle Plugins pane and Use the On ==> button to move it to the list on the right.
http://stevethomson.ca/audio/guides/foo-4.gif

Click the Show Plugin Interface button if the VST Bridge does not open with the last plugin used in it. If it's the first time it's run, click the space to the right of VST Plugin: and choose Load DLL... from the menu that appears. Navigate to the folder containing the VST plugin dlls you wish to use and select flfR.dll. Adjust set it as desired. Note that you can play a file in Foobar while adjusting. Just move back and forth between the two windows.
http://stevethomson.ca/audio/guides/foo-5.gif

When you're ready to write a processed file, stop Foobar from playing the file. Select the file you wish to process in Foobar, then right click on the stereo file in Foobar and choose Convert to save as a wave file. Name your file in such a way that you can identify it later as the front channels.
http://stevethomson.ca/audio/guides/foo-6.gif

Once this is complete, go to the Bridge window and click the dll name to the right of VST Plugin:. If you cannot find the Bridge window, click the Show Plugin Interface button in Foobar's Preferences dialog box. Load CLFE into the VST bridge and set it as desired.
Right click on stereo file in Foobar and choose Convert to save as a wave file. Name the file so that you can identify it later as the center and LFE channels.
When complete, load sLsR into the VST bridge and set it as desired.
Right click on stereo file in Foobar and choose Convert to save as a wave file. Name the file so that you can identify it later as the rear surround channels.
When finished, go back to the bridge and click the text to the right of VST Plugin: and choose Reset to clear the bridge for the next time it's used.

Part 2: Merging or splitting files
Note: you can use other utilities to combine or split the files. For example, CDP Multi-Channel ToolKit at http://www.bath.ac.uk/~masrwd/mctools.html or Besweet as documented elsewhere in this forum. If you're going to split the files, it's a good idea to combine them first because some encoding programs that require 6 mono files will only work if the files are all the same length.

Here's how to merge or split the files with Wavewizard.
Load the three files into wavewizard.
http://stevethomson.ca/audio/guides/ww-1.gif
Choose Preferences from the Edit menu or press Ctrl+F4. Select the options you want for Stream manipulation according to your encoding software's requirements (1 6-channel file or 6 mono files).
http://stevethomson.ca/audio/guides/ww-2.gif
Choose Conversion Batcher from wavewizard's Edit menu or press Ctrl+F3. Important: Make sure the two boxes are not checked next to Send jobs to batcher and Start batcher when finished. You will use the Conversion Batcher later, but not yet!
http://stevethomson.ca/audio/guides/ww-6.gif
Click the Convert button to carry out the conversion. You should end up with one merged 6-channel file or 6 mono files, depending on the settings you choose in the previous step.

Part 3: Encoding
The subject of this guide is how to end up with a surround sound file using completely free (and legal) software. Unless you own commercial AC3 or DTS encoding software, there are only a couple of options for encoding to AC3 files. One option is to use wavewizard with ac3enc.dll, as I explain here.
Go to the directory where you installed wavewizard and in the ConversionBatcher subdirectory, find and run ConversionBatcher.exe. Click the Configure button and then the Programs 2 tab at the top. You need to set it up with the path of your AC3 encoder.
http://stevethomson.ca/audio/guides/ww-4.gif
Close ConversionBatcher and run wavewizard.
Choose Conversion Batcher from wavewizard's Edit menu or press Ctrl+F3. Choose the AC3 Encoder.dll at the bottom of the dropdown list. Be sure to check both boxes next to Send jobs to batcher and Start batcher when finished.
http://stevethomson.ca/audio/guides/ww-3.gif
IMPORTANT: Choose Channel mapping (F2) from the Edit menu. Be sure that the box for 6 -> 6 Softencode and ac3enc is checked as well as the box for Enable channelmapping. This ensures the final ac3 file will have the right channel order.
http://stevethomson.ca/audio/guides/ww-5.gif
Click OK to apply the channelmapping settings and return to the main window.
Click the Convert button to carry out the conversion.

For now, experiment and see if you can get it to work. I tested on several files and got it to work each time.

An extra bonus: stereo playback enhancement
I might add that the fLfR plugin by itself can enhance stereo playback if you use it by itself and turn the front ambience up near its maximum. This won't give you the full V.I effect, but depending on your source material it can have you looking for speakers that don't exist! :) ).


Happy surrounding!
Steve.

[daphy]

Hiho,

as you may have noticed: needfulthings (http://dhost.info/needfulthings/index.php) is under construction

Edit: problems solved!

[ursamtl]

Thanks daphy. I was going to put it on my site this morning until you got needfulthings back up and running. I'll change the link in the guide.

[johnman]

Thats a really nice guide you made


If you don't have one, use the AC3Enc.dll that came with the program


There isnt any ac3enc.dll included in the ww distribution. IIRC one can be downloaded in the ww thread on this forum.

[newhaven]

hi all,
may seem like a stupid question. i have installed foo_dsp_Foobar2WinampPlugin.dll in the components directory of foobar. how do i use this to load christians budde's vst host winamp bridge?

thanks for your time--newhaven

[ursamtl]

No it's a valid question. Be sure you copy Christian's dsp_vst.dll file to the same folder as foo_dsp_Foobar2WinampPlugin.dll, the Foobar2000 Component folder. Then restart Foobar and when you check the DSP Manager in the Properties tree, selecting Winamp DSP Plugin (see step 6 of the guide) should display VST Host DSP v.10 for WinAmp in the left side of the Plugins section of the dialog box. This is Christian Budde's bridge plugin. You just have to activate it and then load a VST dll file. This is because the Foobar2WinampPlugin searches for and lists all Winamp plugins it finds in its directory. A neat trick I've discovered is that if you make a copy of Christian's dsp_vst.dll, rename it to something like dsp_vst_2.dll and load it in the same directory, the Foobar2WinampPlugin lists two instances of it. You can then load activate both and load a different VST into each one!

[newhaven]

ok,
when i double click on the setup icon for christian's bridge, the install box comes up and states: please select your winamp path below (you will be able to proceed when winamp is detected) (install button in lower right corner is greyed out). for my destination folder, i select the component folder of foobar. the ok button stays greyed out, and i cannot proceed any further?
not sure what i'm doing wrong----i don't have winamp installed on the pc? also in the foobar winamp plugin folder i downloaded is a file called msvcr71.dll.
does this go into the components folder also--i saw no mention in the instructions.
thansk again--newhaven

[ursamtl]

Actually sorry, it's been awhile since I downloaded and installed Christian's plugin so I forgot that it tries to install to the Winamp/plugins folder. Since I also have Winamp installed, I just let it go ahead and then copied the dsp_vst.dll file manually to my Foobar/Component folder. As for the msvcr71.dll, yes put it in the Foobar/Component folder as well. Thanks for bringing this up. I'll update the guide right away.

Regards,
Steve.

[ursamtl]

Just a note to those who downloaded the software and tried this guide, the sLsR v1.0 plugin I released on January 14 contained a problem with reduced ambience. I've fixed this and released version 1.1. I think you'll find this greatly improves the surround field. Check the message at http://forum.doom9.org/showthread.php?p=773084#post773084 for download links.

Regards,
Steve.

[newhaven]

steve,
when i start foobar up. i get a pop up console that says: Failed to load DLL: foo_dsp_Foobar2WinampPlugin.dll, reason: Unable to load DLL. I can't get rid of this i have tried uninstalling foobar and putting all the DLL's back in the component folder, but i still get the error message?? any ideas?

thank you--new haven

[ursamtl]

I haven't seen that error before. When I put the foo_dsp_foobar2WinampPlugin.dll in my Foobar/Component directory, it worked ok.

Anyone else have this problem? You might also check the foobar forum at http://www.hydrogenaudio.org/forums/index.php?act=SF&s=&f=28. I'll let you know if I find anything out.

[newhaven]

Steve,
i was able to resolve the issue with foobar. the link provided for the winamp plugin above, has 4 selections. 1 of the files contained a third DLL (msvcp71). put this in the components folder and no more problems loading.
i am currently using wavewizard to process the files from your wonderful proggy to 6 monostreams that i want to convert using surcode dvd dts encoder (not using conversion batcher). each of the 6 streams has a 0 & 1 after the title (i.e. center & LFE channels ch0,center & LFE channels ch1, front channels ch0,front channels ch1, rear channels ch0,rear channels ch1)
can you possibly tell me what each ch# properly corresponds with (example--center & LFE channels ch0 is the center channel, front channels ch0 is the front left,rear channels ch0 is the left rear and so on?)
than you for your time! --newhaven

[ursamtl]

Glad to hear you got the Foobar problem resolved. In Foobar Diskwriter preferences, it's possible to have the program prompt you for a filename by checking the Ask before writing checkbox. This way, you can give you generated files descriptive names. What I used during my tests was adding "flfr" to the end of the filename for the fronts, "clfe" for the centre/lfe and "slsr" for the surrounds. Then you can place them in the order flfr, clfe, slsr in Wavewizard.

Steve.

[newhaven]

Steve,
i did what you suggested (CLFE,FLFR,SLSR) in foobar originally. my confusion comes when i load the 3 files into wave wizard. i am using a surcode dvd dts encoder----the only way i can load the files in Surcode is by having wavewizard convert the 3 WAV's into 6 mono streams (which Surcode accepts). after converting the 3 WAV's (c&LFE,flfr,slsr) to 6 mono streams i have the following:
c&LFE ch.0
c&LFE ch.1
flfr ch.0
flfr ch.1
slsr ch.0
slsr ch.1

what are the 0's and 1's? (examples: c&LFE ch.0=center
c&LFE ch.1=sub bass & LFE
flfr ch.0= left front
flfr ch.1= right front
slsr ch.0=surround left
slsr ch.1=surround right )
do you know what i listed in the examples is correct ?

thanks again, sorry if i'm a bit "green" at this:sly:
newhaven

[ursamtl]

These look to be correct, yes. Give them a try.

No need to apologize for being green at this. We all were not so long ago! :)

[newhaven]

Steve,
thanks for the kindness and patience, but MOST of ALL, THANK YOU FOR THIS GUIDE!!!!!!!! i have successfully used your method twice now, and to say i am pleasantly surprised at the final quality of the product is an understatement . WOW! is more like it.:D :D :D

thanks again, see ya around the forums ---newhaven

[ursamtl]

Hey I'm happy it's turning out well for you. I continue to be amazed myself with how good these conversions turn out.

Thanks for the feedback.
Steve.

[mfsn]

I followed the steps of your guide "Converting stereo to 5.1 surround for FREE" and was able to generate the 6-channel WAV file by merging 3 stereo (2 ch) files with wavewizardv0.54b. Then I burned a CD that didn't play at all on my home theater DVD. Could it be a problem with the ac3enc.dll???

How can I create 5.1 WAV files that will play on my DVD using the FREE method??

Thanks.

[ursamtl]

Go back and recheck everything. I know it took my a few tries before I got the process working correctly. I found it helpful to use CDRWs until I got things working.

One of the most common problems people run into is sampling rate. If you're burning a surround CD, the audio file has to be at a sampling rate of 44.1 kHz. If it's for a DVD soundtrack, it has to be at 48kHz. People often think because a surround CD will be played back on a DVD player, they need to convert to 48kHz. Not so. The other most common problem is the hook-up between the DVD player and the receiver. It has to be a digital SPDIF or optical connection. Once the data goes through the left and right audio connections, it's already been converted to analog.

A surround CD does a sort of "double trickery" on your system. First, it fools your DVD player into thinking it's a regular audio CD by being at 44.1kHz. Then the data going into the decoder has an AC3 identifier embedded, so the decoder thinks it's receiving the audio from a DVD and thus decodes it as such.

You could retrace your steps and save files every step of the way. since you've installed foobar2000 to do this method, you can use it to play back the 6 channel wave file to see if it's making sounds (if you have a 5.1 soundcard/speaker setup, you can use the Directsound of Kernel Streaming outputs in foobar to hear all 6 channels in surround). Foobar also plays back AC3 files, so you can check them as well.

Don't give up. You'll get it!

Regards,
Steve.

[mfsn]

I have changed the the sampling rate to 44.1 kHz but now when I open the file it looks like an ordinary PCM audio file with the same amplitude/frequency modulations as the original stereo files. Instead, shouldn't I expect to see bursts of data separated by spaces (zeros) when I visualize the file?

[ursamtl]

I have changed the the sampling rate to 44.1 kHz but now when I open the file it looks like an ordinary PCM audio file with the same amplitude/frequency modulations as the original stereo files. Instead, shouldn't I expect to see bursts of data separated by spaces (zeros) when I visualize the file?

Unless you stretched the data out tremendously in the app you're using to view it, no you shouldn't see spaces. How does the file sound when you play it back?

[mfsn]

I'm definitely able to create all three WAV files FLFR, CLFE, SLSR in 32-bit floating-point and play them without any problem. Then I run wavewizard on these files and it indicates a successful conversion, generating the merged WAV file.

However, when I open the logging window from the Conversion Batcher checkbox and display the "Jobs log", I get the following message:

Starting CB
***** Starting ac3 encoder dll *****

***** ac3enc.dll is missing or invallid, skipping current job *****


I am using the ac3enc.dll that came with BeSweet and I just copied it into the ..\WAVEWIZARDV0.54B\CONVERSIONBATCHER subdirectory. Any idea why this is happening?

Thanks.

[ursamtl]

Did you set up the path to ac3enc.dll in wavewizard's configuration?

[mfsn]

Hey,

I was finally able to encode the 5.1 WAV file from 6 WAV files using BeSweetv1.5b31 with BeLight-0.22beta9 GUI and it worked painlessly. It's very nice too.

I did set up the path to ac3enc.dll correctly in the Wavewizard configuration but I think my old ac3enc.dll had a problem so I just decided to reinstall everything from scratch.

I'm now interested in finding out the formulas to extract FL, FR, C, LFE, SL, and SR channels, I basically know that C=(FL+FR)/2 gives a boost on vocals but don't isolate it totally, perhaps some extra band-pass filtering would help.

For the surround channels, SL=SR=(FL-FR) will cut the vocals out, but the result is not very clean. A 120 Hz low-pass filter for the LFE should be ok, I just don't know what to do with the FL and FR. Keeping them just like the originals doesn't make much sense to me.

Anyway, if someone knows of any good literature on this subject, it would be very helpful.

Thanks a lot.

[ursamtl]

I'm glad you got it working. It can be a really cool feeling to get these conversions working and discover all the sound information that's seemingly "hidden" in a stereo mix.

As for the formulas, there are no exact rules, but many stereo to surround conversions are based on some sort of manipulation of the sum and difference signals in the stereo signal. In V.I and its related plugins, the formulas are a bit more complex: over 300 connections among software modules, a dozen virtual speaker positions superimposed over an ITU 5.1 layout, bandpass filtering plus phase inversion for Movie Mode, 60Hz lowpass filtering for the LFE channel, and some other stuff.

If you want some references, check the "General Reading and Information" part of the Additional resources (http://forum.doom9.org/showthread.php?p=558760#post558760) section of the guide list. There are some really good resources there. Certainly the Sursound mailing list is a wealth of information.

Enjoy!
Steve.

[daphy]

I am using the ac3enc.dll that came with BeSweet and I just copied it into the ..\WAVEWIZARDV0.54B\CONVERSIONBATCHER subdirectory. Any idea why this is happening?

wrong version :scared:
see wavewizard thread (http://forum.doom9.org/showpost.php?p=696810&postcount=50) or follow the links at needfulthings (http://dhost.info/needfulthings/index.php) ;)

[aichan]

nice guide, stereo to surround for free.. :)

[stars]

i think there must be a fault somewhere in the final ac3 encoding
the sound channels have the woring order and cant be re mapped.
And the settings for the ac3.dll cant be changed...
I ended up with converting the audio into 6 mono channels and do the final encodning with besweet....

stars....

[ursamtl]

You're right. However, if you use wavewizard with the correct configurations for ac3enc.dll, it should take care of that for you.

[johnman]

"I am using the ac3enc.dll that came with BeSweet and I just copied it into the ..\WAVEWIZARDV0.54B\CONVERSIONBATCHER subdirectory. Any idea why this is happening?"

Wavewizard is only tested with ONE version of ac3enc. I also have had problems once with ac3enc which was supplied with a particular version of headac3he which somehow did not work. I asked DA and he gave me a newer version which can be downloaded from the ww thread here on doom9.

I know it would be more easy if the ac3enc was downloadable from within CB but for a couple of reasons i cant do it unfortunatly. Once i get the okay from DA i willl host it myself immediately.

[johnman]

http://forum.doom9.org/showpost.php?p=696810&postcount=49

link to the ac3enc

[stars]

You're right. However, if you use wavewizard with the correct configurations for ac3enc.dll, it should take care of that for you.


do you the mean the options in the channel mapping menu..:)

i will config for ac3enc and give it a try....

stars.....

[tebasuna51]

@ursamtl
I think your method to encode with WaveWizard-ac3enc need this correction:

4. Choose Conversion Batcher from the Edit menu and uncheck Sends Jobs to Batcher.

5. Click the Convert button to obtain a wav 6 channels Surround test 4 fLfR_Merge.wav (with channels in correct wav order).

6. Clear_List and load the wav 6 Surround test 4 fLfR_Merge.wav. Now you need:
- Edit menu Channel Mapping, select ac3 order for SoftEncode or ac3enc
- Edit menu Preferences, check Enable Channelmapping and uncheck Stream manipulation.
-Edit menu Conversion Batcher and check Sends Jobs to Batcher and Start batcher when finished.

7. Click the Convert button to obtain the encoded ac3.

@Johnman
Seems that Stream manipulation and Channelmapping don't work together, the merged file in pass 5 is always in correct wav order with Channel mapping enabled or disabled.

[ursamtl]

tebasuna51,

Thanks for the feedback. I'll check it out but I did test the guide as it's written and the resulting surround files all had the same correct channel order as well. You might double-check your settings from the beginning, perhaps there's something you did differently earlier on that made it necessary for you to modify the method at the end to get good results.

Regards,
Steve.

[tebasuna51]

@ursamtl
If you load into WaweWizard:
Test_fLfR.wav
Test_CLFE.wav
Test_sLsR.wav
when you click Convert, WaveWizard make a intermediate wav 6 file with correct wav order FL_FR_C_LFE_SL_SR, and is send to ac3enc, but ac3enc need a wav ordered FL_C_FR_SL_SR_LFE then only the FL channel is in the correct place.

Maybe Johnman can confirm you this issue.

[ursamtl]

Ok, thanks for bringing this up tebasuna51. In fact, after testing this evening, I discovered that I had forgotten to add the step about enabling channelmapping to my guide. Do not disable sending the jobs tro the batcher as you suggest. Simply choose Channel mapping (F2) from the Edit menu. Be sure that the box for "6 -> 6 Softencode and ac3enc" is checked as well as the box for "Enable channelmapping." This ensures the final ac3 file will have the right channel order. I tested this with three stereo waves loaded into ww in the ITU 5.1 order (fLfR, CLFE, sLsR) and the resulting ac3enc-encoded ac3 file had the right channel order when I played it back through a 5.1 system. I've edited the guide and added a new screen grab.

Thanks for your help.

Regards,
Steve.

[tebasuna51]

Simply choose Channel mapping (F2) from the Edit menu. Be sure that the box for "6 -> 6 Softencode and ac3enc" is checked as well as the box for "Enable channelmapping." This ensures the final ac3 file will have the right channel order.
Yes, I tried this first time, but don't work for me.

In my previous post I say to Johnman:
"Seems that Stream manipulation and Channelmapping don't work together, the merged file in pass 5 is always in correct wav order with Channel mapping enabled or disabled."

For that I think we need 2 pass, first generate the wav 6 (always in correct wav order), and after open this wav 6, apply the remapping and encode.

Sorry.:)

[ursamtl]

Yes, I tried this first time, but don't work for me.

In my previous post I say to Johnman:
"Seems that Stream manipulation and Channelmapping don't work together, the merged file in pass 5 is always in correct wav order with Channel mapping enabled or disabled."

For that I think we need 2 pass, first generate the wav 6 (always in correct wav order), and after open this wav 6, apply the remapping and encode.

Sorry.:)

2 passes is exactly what it does for me. First it create the merged file, then it sends it to Conversion Batcher.; If you set it up the way I explain in the guide, you will end up with a correctly encoded AC3 file.

Perhaps the problem is that you are looking at the merged file and expecting it to be in the remapped order. It is not. Only when it gets passed to the encoder does the channel remapping take place. If you take the merged file and try to encode it yourself manually, you will end up with the wrong channel order in your AC3s. If you let ww take care of it for you, it will do the necessary remapping. This is one reason why I like johnman's wavewizard so much. I've used it on quite a few projects and it has never let me down!

Keep trying!
Steve.

[tebasuna51]

Perhaps the problem is that you are looking at the merged file and expecting it to be in the remapped order. It is not. Only when it gets passed to the encoder does the channel remapping take place. If you take the merged file and try to encode it yourself manually, you will end up with the wrong channel order in your AC3s. If you let ww take care of it for you, it will do the necessary remapping. This is one reason why I like johnman's wavewizard so much. I've used it on quite a few projects and it has never let me down!
Yes, I like wavewizard so much too. The remapping function "6 -> 6 ac3 order for SoftEncode or ac3enc" (and other issues about SoftEncode) is my very little contribution to Johnman's project.

But, seems you don't understand the wavewizard behavior :). Wavewizard don't work sending to ac3enc the wav file in pipe mode, need always a intermediate wav file to send to encoder. And, if the last merged file is in wav order, the ac3 output is wrong mapped.

I know work in pipe mode is in Johnman's mind, but AFAIK the last beta version 0.54b don't still work so. I test this (including the last ac3enc.dll from this post) with a channel_test and can confirm you: if the last wav is not ordered FL_C_FR_SL_SR_LFE, the ac3 output is wrong mapped.

In order to avoid disk usage (the two wav6 32bit can be really big) I can propose another method using BeHappy (pipe mode) to encode the three wav (lFlR, CLFE and sLsR) to ac3 with the free encoder ffmpeg (equivalent to ac3enc).

I know, is not a easy way because BeHappy is in developing stage, but can be useful for somebody:

Software required
- Avisynth 2.5.6a (http://www.avisynth.org/) installed.
- BeHappy: (http://workspaces.gotdotnet.com/behappy)Download last version and unzip to a folder at your choice.
- ffmpeg: (http://kurtnoise.free.fr/ffmpeg.7z)Unzip ffmpeg.exe to BeHappy folder.
- ffdshow installed and properly configurated: Audio Decoder -> Codecs -> Uncompressed -> All supported (to open 32bit float from avisynth with DirectShowSource). Check also filters and output configuration.
- AvsSource.extension: You need create a new file in BeHappy folder with the name "AvsSource.extension" and content (copy and paste to Notepad for instance):
<?xml version="1.0"?>
<BeHappy.Extension xmlns:xsd="http://www.w3.org/2001/XMLSchema" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xmlns="http://workspaces.gotdotnet.com/behappy">
<AudioSource Name="AvsSource" UniqueID="0aa78710-aafd-11da-a746-0800200c9a66">
<Script>Import("{0}")</Script>
<SupportedFileExtension>avs</SupportedFileExtension>
</AudioSource>
</BeHappy.Extension>

For other BeHappy usage you can read Basic Help for BeHappy (http://forum.doom9.org/showthread.php?p=779783) or BeHappy development (http://forum.doom9.org/showthread.php?t=104686)

BeHappy usage to encode 3 stereo wav (fLfR, CLFE and sLsR) to ac3 with ffmpeg.
1) You need make a avs file, (YourName.avs) with the content (put your exact path and filenames):
flfr = DirectShowSource("g:\Test fLfR.wav")
clfe = DirectShowSource("g:\Test CLFE.wav")
slsr = DirectShowSource("g:\Test sLsR.wav")
MergeChannels(flfr, clfe, slsr)

2) Run BeHappy.exe and open (in Source section) YourName.avs. Check the source mode is "AvsSource"

3) Select the Destination method "ffmpeg AC3" and configure (Button "...") the desired bitrate. Select the Destination filename and press Add to Job Control.

4) Go to Job Control tab and press Start.

[ursamtl]

Ok thanks again for your input. You're right. The channel order was wrong when running ww with both the merging and encoding at the same time. I was using a regular 6-channel V.I-processed version of one song to test. It was difficult to really ascertain where the channels went because the ambience still seemed quite good. I did some testing this morning with completely separate songs in each channel and this way, I confirmed the channel orders. I've rewritten the guide to split the job into two passes manually. This should do the trick.

Your Avisynth, BeHappy & ffmpeg approach looks good; I'll give it a try. Another option that could work well is HeadAc3He using the latest alpha version available on daphy and @ndy's www.needfulthings.com. This includes a modified version of ac3enc.dll that seems to solve the problem with low volume that's plagued the regular ac3enc.dll that's been used with Besweet and ffmpeg. I'll try and write a guide for it sometime soon as well.

Regards,
Steve.

[dimzon]

This includes a modified version of ac3enc.dll that seems to solve the problem with low volume

Is it possible to apply same modification to regular ffmpeg???

[tebasuna51]

Is it possible to apply same modification to regular ffmpeg???
I think all recent ac3enc.dll (for BeSweet, for WaveWizard) are similar to ac3enc.dll from HeadAC3he v0.24-a13 by Dark Avenger 15.12.2004. Also ffmpeg do the same volume output (50%) like the dll's.

But only HeadAC3he have a parameter "Gain" send and applied for the dll. Then if we have a source normalized 100% and put 6 dB (x2) Gain the ac3 output go to 100% also.

Then the possible midification is only multiply by 2 the signal at the correct point.

[dimzon]

I think all recent ac3enc.dll (for BeSweet, for WaveWizard) are similar to ac3enc.dll from HeadAC3he v0.24-a13 by Dark Avenger 15.12.2004. Also ffmpeg do the same volume output (50%) like the dll's.

But only HeadAC3he have a parameter "Gain" send and applied for the dll. Then if we have a source normalized 100% and put 6 dB (x2) Gain the ac3 output go to 100% also.

Then the possible midification is only multiply by 2 the signal at the correct point.
Can You create feature request @ ffmpeg development site? (your english is much more better)
http://sourceforge.net/projects/ffmpeg/

[tebasuna51]

Can You create feature request @ ffmpeg development site?
Done.

[ursamtl]

Just thought I'd let everyone whose following the thread know that I've tried testing this guide with Foobar v0.9 (which is now the only one available on the Foobar web site) but v0.9 does not support most v0.8x plugins, including the ones mentioned above. Fortunately, those who don't have v0.83 can still find it at Really Rare Wares (link in the guide). Too bad, there are a lot of great plugins that won't work in the new version.

[Betchin]

I don't get something, Can someone Explain the part of installing the bride plugins in foobar. And where can I get that foobar2winampplugin??? I do not really get that part :(

[ursamtl]

I don't get something, Can someone Explain the part of installing the bride plugins in foobar. And where can I get that foobar2winampplugin??? I do not really get that part :(

It's quite simple. In the directory where you install foobar, there's a subdirectory called Components. Simply put the bridge plugin files in there then restart foobar. As for the files themselves, the links are in this (http://forum.doom9.org/showthread.php?p=768141#post768141) message. Be sure to use foobar v0.83 only. Don't use v0.9; it won't work).

Give these a try and if you can't get things to work, let us know and we'll try and get you up and running.

Good luck!
Steve.

[Betchin]

I don't want to doublepost but figured out what I did wrong now (I confused the vstplugin with the dsp). Though there is one slight problem...the bridge plugin winamp dsp is not a valid link. It immediately refers me to some .txt
and I have tried to search it up on google though without any results ... :(

[ursamtl]

The author seems to be rewriting it for v0.9 and has taken the old version down in the meantime. I don't have the file on my computer right now as I'm at work. Perhaps someone else might send it to you. Otherwise, try Googling a bit more. Some of those asian sites may have it for download. Try clicking on the Google "Translate this page" link. In the meantime, I'm working on another totally free way to produce these surround files using the V.I algorithm. It'll be much simpler than this guide (only 1 step) and will produce the same results. Just give me some time to put a couple of things together and you'll have it.

Regards,
Steve.

[Betchin]

that would be really nice ! Could you keep us up with the information when you have finished it xD?

I think this is one hard way to convert stereo to surround. I was looking over the internet for some other solutions but found none.

I wish u luck with your work !

[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.

[capricorn]

Hi Tebasuna51-

I've tried the parameters you supplied with just - %d with no luck. I've also tried all the versions of aften.exe on the kurtnoise site as well. Which version of aften.exe should I be using?

Thanks for your help.

cap

[tebasuna51]

I've tried the parameters you supplied with just - %d with no luck. I've also tried all the versions of aften.exe on the kurtnoise site as well. Which version of aften.exe should I be using?

The last rev521, but with only defaults (- %d) all must work.

[capricorn]

Hi tebasuna51-

It didn't work but I'll try reinstalling all the programs and components again and see what happens.

cap

[Hyper_Thread]

Kurtnoise's builds don't support pipes (rev521), so use Wisodev's builds (http://win32builds.sourceforge.net/aften/index.html). i'm using the SSE3_MT build, it works fine.

[tebasuna51]

Kurtnoise's builds don't support pipes (rev521), so use Wisodev's builds (http://win32builds.sourceforge.net/aften/index.html). i'm using the SSE3_MT build, it works fine.

No problem with Aften rev521 pipe with Foobar or AviSynth (Bepipe, BeHappy or Soundout), Windows XP SP1, .NET Framework v2.0, Pentium 4, 2.4GHz.

Try with the aproppriate Wisodev 0.07 version and put your comments (hard and soft used).

[tebasuna51]

With Windows Vista Home Basic (32 bits) rev521 don't work in pipe or command line mode.

All Wisodev v0.07 32bits versions works for me with a Pentium Dual Core.

[tebasuna51]

Sorry, confirmed by Kurtnoise13 theirs builds don't work with multithreading processors.

Changed the Aften link to Wisodev's builds.

[Hyper_Thread]

Ok, problem solved...I'v been using graphedit+ffdshow to upmix my audio streams, but this method produce much better surrond sound (with tuned Aften's options, off course)
Thanks again to you & ursamlt for this great guide.

[shadowhaze]

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

I had a similar problem when I first used your conversion method. I copied the command line encoder settings parameters from your first guide post (which you reference in the updated guide) -v 0 -pad -readtoeof 1 -dnorm27 -dynrng 1 -b 448- %d I now use -v 0 -pad 0 -readtoeof 1 -b 576 - %d and it works fine. For some reason, removing "-dnorm27" and "-dynrng 1" fixed the problem for me.

Like everyone else :thanks: for posting this guide. Once you get everything installed and set up, it's very easy, fast and the results are great.

[tebasuna51]

I had a similar problem when I first used your conversion method. I copied the command line encoder settings parameters from your first guide post (which you reference in the updated guide) -v 0 -pad -readtoeof 1 -dnorm27 -dynrng 1 -b 448- %d I now use -v 0 -pad 0 -readtoeof 1 -b 576 - %d and it works fine. For some reason, removing "-dnorm27" and "-dynrng 1" fixed the problem for me.
The correct parameters are:
-v 0 -pad 0 -readtoeof 1 -dnorm 27 -dynrng 1 -b 448 - %d
Really the only required is -readtoeof 1 if the equivalent multichannel wav is > 4GB.
You are free to use others parameters (execute Aften -h to a complete list of parameters).

[NorthPole]

@tebasuna51

Havent followed this thread in a while and just though I would check it since there is a new version of foobar out. WOW...a 6 channel vst hosting option....Fantastic. I had been using the ATsurround plugin but I think this is probably better. Thanks


BTW... If anyone is interested, there is a vst host plugin for the free Audacity sound editor

Links is http://audacity.sourceforge.net/help/faq?s=install&item=vst-enabler

[jkowtko]

Hi,

This surround conversion is pretty neat.

However I am having trouble getting LFE signal from the resulting AC3 files. I don't know if it's the conversion process or the audio driver I'm using (BASS library with bass_ac3.dll).

For conversion I used the foo_channel_mixer, the VST bridge, VI.dll surround converter, and AFTEN x86 for the encoding. Everything sounds great except for no LFE.

If I am doing something wrong on the conversion or have to set up VI or Aften differently to get LFE, please let me know. However I've tried sample AC3 files from others and don't hear any LFE from them either.

Or, if you think it's my playback system and can send me an AC3 sample audio clip with a clear and loud LFE channel on it, I'll try it out on our PA system to see what it does. (Fyi, the PA plays 6 channel WMA just fine, LFE works great.)

Alternately, if AC3 is problematic in this area and there is a 6 channel WMA encoder, I'll take that as an option :)

Thanks. John

[Kurtnoise]

Now we need:


- V.I Suite version 1.11 Installer (http://www.stevethomson.ca/vi/)
this link doesn't work for me...Does anybody can provide a fresh link please ?

[tebasuna51]

this link doesn't work for me...Does anybody can provide a fresh link please ?

Yep, seems the Steve space don't work.

I can offer my copy here: VI_Setup.zip (http://www.sendspace.com/file/mqdwm1)

[daphy]

this link doesn't work for me...Does anybody can provide a fresh link please ?

site is up again :rolleyes:

[Sakuya]

Sorry for bumping an old thread. I just tried this alternative and found that it sounds pretty good. But I noticed that for a certain song, there's a noticeable clipping in the surround and it sounds blarey. The original was a FLAC file and was not like this. In foobar, I used the Advanced Limiter but it didn't do much. Is there a way to limit it?

Also, did anyone else have problems for certain songs with lots of choruses where the main vocal is coming from all the speakers rather than primarily the front? It sounds more like a concert hall effect than 5.1. I know there is a certain limit since we're converting stereo to surround but do the V.I. settings help in pushing the main vocals to the center channel more?

[newhaven]

I havent been around the forums much due to real life. I was pleasantly surprised to find this old post.

The links to download the install files for the Foobar2000 v0943 method by Tebasuna are not all working.

If anyone would like the install files (which will take a little digging and experimenting with on the net) please PM or email mail me at:
newhaven1074@yahoo.com.

I have loved this method since its inception many years ago. Special thanks to Steve Thomson & Tebasuna for this wonderful thread!!

Newhaven

[tebasuna51]

Now there are a better VST plugin for Foobar2000, the foo_channel_mixer is not needed now and is more easy to configure.

Tested with last Foobar2000 v1.2.9

a) INSTALL
- Foobar2000 v1.2.9 (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/y9mbl1))
- VST 2.4 adapter (http://www.hydrogenaudio.org/forums/index.php?showtopic=84947) (foo_vst). Uncompress the .dll in ...\Foobar2000\components folder.
- Aften (obsolete use now ffmpeg (https://forum.doom9.org/showthread.php?p=1907333#post1907333)) Ac3 free encoder. Uncompress Aften.exe in any folder, for instance in G:\St_Surr (C1).

b) CONFIGURE
- Foobar2000, in File -> Preferences -> Components -> VST plug-ins -> Add -> C2\VI.dll.
- Configure Aften encoder like before (http://forum.doom9.org/showthread.php?p=991707#post991707) with: Mouse right click over a track -> Convert -> ... -> Convert Setup -> Output format -> Add new -> Encoder -> Custom...

c) PROCESS
Open a stereo file in any format supported by Foobar and:
1) Mouse right click and Convert -> ... to open 'Convert Setup'
2) In 'Convert Setup' -> Output format -> select AC3 (AFTEN) directly, select Destination and click in Procesing to open DSP Settings.
3) Active the VI plugin and click in Configure Selected.
4) The V.I settings are explained already.
5) Now there are a new option, the output channels, put always '6' (when LFE is not checked the LFE channel is empty but always present).

[newhaven]

Thanks Tebasuna, much easier!

[gildardo]

I'm sorry, but I just don't get it... some screen shots may be missing from previous posts, as I in the link http://forum.doom9.org/showthread.php?p=991707#post991707 I see no screen in b) Configure after "... and fill this screen:" also, I don't see where "4) The V.I settings are explained already."

I apologize for nagging, but after weeks of doing a ton of Google searches on a project, my mind is completely shot, and I just can't find where things are "explained"... so please don't be angry

I also apologize, as this is my first post on doom9, after quite a bit of lurking around and assimilating information -- which has always been a lot of help, so thank you all...

[filler56789]

I'm sorry, but I just don't get it... some screen shots may be missing from previous posts, as I in the link http://forum.doom9.org/showthread.php?p=991707#post991707 I see no screen in b) Configure after "... and fill this screen:" also, I don't see where "4) The V.I settings are explained already."

Well, those images WERE hosted at the infamous imageshack, and the original URLs are pretty dead by now...

[gildardo]

Dang... :(

Thank you filler56789

Edit, sorry for the commotion

Figured it out --- the info was all there, but I was burned out from reading and couldn't see it --- it works PERFECTLY, I just used the presets listed by tebasuna51 a few posts above the most recent guide (muchas gracias por la guía, caballero!). It was just to put them in the presets for the encoder options in foobar.

I am now a happy camper :)

[zdoe]

ok, picking up an old thread, but...

goal - get to 5.1 from stereo (for DCP). WAVs, as that's not lossy. are you still around, @tebasuna51?

seemingly all installation steps work ok, but as a result i get a 2 channel WAV, and nowhere is there an option to get more channels out. if i try to do AAC, aften compressor crashes.

what gives?

[tebasuna51]

but as a result i get a 2 channel WAV, and nowhere is there an option to get more channels out.

I tried with a recent version of Foobar (1.3.7) and work fine for me with the same versions than before of V.I Suite version 1.11 and VST 2.4 adapter.

Can you configure the VI filter like is show in the image attached here (https://forum.doom9.org/showthread.php?p=1640451#post1640451)?