BeHappy - AviSynth based audio transcoding tool (UPD 19-07-2006) - Page 5

dimzon · 14th February 2006, 12:23

Quote:

Originally Posted by grokwik

I have noticed that nero aac encode works with or without neroipp.dll without any difference. Is this dll really necessary ?

It's really necessary if you does not have Nero installed

JoeBG · 14th February 2006, 19:44

I cant get work
- upmix (is it still not supportet) (invalid arguments to function "SuperEq"
- normalize (nothing happens)

Do I need some special software packages?

tebasuna51 · 14th February 2006, 20:05

- Upmix is in experimental stage by NorthPole, and need Sox Audio Effect Filter ( http://forum.doom9.org/showthread.php?t=104792 ), see the post:
http://forum.doom9.org/showpost.php?...8&postcount=66

- Normalize works fine (amplify the signal until the required level).

JoeBG · 14th February 2006, 20:50

Quote:

Originally Posted by tebasuna51

- Upmix is in experimental stage by NorthPole, and need Sox Audio Effect Filter ( http://forum.doom9.org/showthread.php?t=104792 ), see the post:
http://forum.doom9.org/showpost.php?...8&postcount=66

Works fine now, many thanks

Quote:

Originally Posted by tebasuna51

- Normalize works fine (amplify the signal until the required level).

What means "amplify the signal until the rquired level

I told the tool to normalize to 100 %. Is this not ok?

tebasuna51 · 15th February 2006, 01:23

Quote:

Originally Posted by JoeBG

What means "amplify the signal until the rquired level
I told the tool to normalize to 100 %. Is this not ok?

I say "required level" only because you can select other value than 100% with the Configure button.

But, is true, the Normalize() function don't work after the upmix because do a second pass (the first to know the max value) to amplify the sound. And some Sox filters doesn't support restarts (crash), like say the documentation.

I have the same problem with the Dynamic Range Compession function and the compand Sox filter.

NorthPole · 16th February 2006, 01:43

Well here is the latest revision on the upmix extension

Code:

<?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">
  <AudioDSP UniqueID="9579E57B-2D27-4583-99A4-921718E25B41">
    <Plugin>
      <MultiOptionDSP Type="BeHappy.Extensions.MultiOptionDSP, BeHappy">
        <TitleFormatString>5.1 Upmix - {0}</TitleFormatString>
        <ScriptPrologue>
	# Store clip in variable
	Stereo_{2} = convertaudiotofloat(last)
        </ScriptPrologue>
        <Option>
          <Name>Audio with mostly dialog (ie. Comedy, Drama)</Name>
          <Value>
	# Profile to use with audio sources that has mostly mono content. 20ms delay and -2db attenuation on surround
	front_{2} = stereo_{2}.soxfilter("filter 20-20000")
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.794,-0.794)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.794,-0.794)
	cc_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),fl_{2},1,-1),mixaudio(front_{2}.GetRightChannel(),fr_{2},1,-1),0.224,0.224)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.562")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.631,-0.631)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.631,-0.631)
	sl_{2} = DelayAudio(sl_{2},0.02)
	sr_{2} = DelayAudio(sr_{2},0.02)
          </Value>
        </Option>
        <Option>
          <Name>Audio with a mix of sounds (ie. Action, Adventure)</Name>
          <Value>
	# Profile to use with audio sources that more range of sound content. 20ms delay and -2db attenuation on surround
	front_{2} = stereo_{2}.soxfilter("filter 20-20000")
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.668,-0.668)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.668,-0.668)
	cc_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),fl_{2},1,-1),mixaudio(front_{2}.GetRightChannel(),fr_{2},1,-1),0.398,0.398)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.447")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.531,-0.531)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.531,-0.531)
	sl_{2} = DelayAudio(sl_{2},0.02)
	sr_{2} = DelayAudio(sr_{2},0.02)
          </Value>
        </Option>
        <ScriptEpilogue>
	# Return result
	MergeChannels( fl_{2}, fr_{2}, cc_{2}, lfe_{2}, sl_{2}, sr_{2})
	ConvertAudioTo16Bit()
        </ScriptEpilogue>
      </MultiOptionDSP>
    </Plugin>
  </AudioDSP>
</BeHappy.Extension>

changes are as follows:
- removed the frequency adjustment option because the quality was not very good.
- optimized sox filter usage to speed up execution.
- created 2 different sound profiles to use on different types of audio.

Notes on Usage:
1. You need to have the avisynth sox filter in the avisynth plugin subdirectory. See tebasuna51s' post above for the link.
2. Normalize you audio before upmix.
3. Choose a profile to upmix with from the 5.1 upmix config button.

If after upmixing, you can't hear the center channel dialog because the L & R channels are drowning it out, then try the other profile.
Also, you can add your own profile or change the above ones as needed.

@dimzon

Small feature request...

When you add a file to convert, the name of the destination file is shown as the same as the input file (excluding the dot extension) which is good. But when you add a 2nd file to convert, the destination file name doesn't change (it remains the same as the first). Therefore, you have to type in the destination file name.

Not a big deal if its a problem to change... I hope I explained that clearly enough?

iceborne · 16th February 2006, 03:03

does downmixing dts to DLP II wav really work in behappy? wanted to know cause i like to trancode DLP II wav to itunes's aac format.

could you also clarify between DPL II and DPL II(lfe)?
which one is equivalent to "-azid( -s dplii -c normal -L -3db )"?

tebasuna51 · 16th February 2006, 17:53

DPL II is equal to "-azid( -s dplii )" (copied matrix).
DPL II(lfe) equivalent to "-azid( -s dplii -L -3db )".
The "-azid( -c normal )" switch is for Dynamic Range Compression not yet implemented in BeHappy (problem with compand Sox Filter and Normalice see: http://forum.doom9.org/showpost.php?...5&postcount=60 ).

If you have ffdshow like default directshow filter to play dts files you can use DirectShow input instead NicDtsSource, and use ffdshow Audio Decoder Configuration to set:
- Codecs -> Dtd - libdts, Dynamic range compression (Checked)
- Mixer -> Dolby Prologic II, LFE (Unchecked)
-Output -> Output sample - only checked 16 bit integer, Don't use WAVEFORMATEXTENSIBLE ...(Checked)

This is equivalent to "-azid( -s dplii -c normal -L -3db )". After use the Normalice DSP function in BeHappy and select your desired encoder.

I test this method with ac3 source, I hope works also with dts.

Edit: I make a test with dts source:
- Same volume output with NicDtsSource and with ffdshow (drc checked and drc unchecked) (?).
- The output length with NicDtsSource is 2:06.688 (like input) and with ffdshow is cut at 1:50.851 (?)

iceborne · 16th February 2006, 23:39

from this thread
http://forum.doom9.org/showthread.php?s=&threadid=27936

Quote:

"Pro Logic II with LFE (Not recommended by Dolby)"

... so i shouldn't even be using "-azid( -s dplii -L -3db )" anyways, correct?

Quote:

For downmix you ever have to use DRC because more acoustic power will be concentrated in less channels!

what is frank trying to say? not use dynamic range or to use dynamic range on downmix?

tebasuna51 · 17th February 2006, 01:55

See also http://forum.doom9.org/showthread.php?t=57988 with the definitive matrix used in azid 1.9 and in DPL II in BeHappy.

Quote:

Originally Posted by iceborne

... so i shouldn't even be using "-azid( -s dplii -L -3db )" anyways, correct?

From "214_Mixing with Dolby Pro Logic II Technology.pdf" :
"There are other concerns when adding an LFE signal to the mix. If the LFE is simply redistributed within the other channels of the mix, they will usually be subject to some low-frequency bandpass filtering. This filtering causes phase shifts of the LFE signal. When they are acoustically added within a room, these phase shifts are fairly subtle and often go unnoticed. However, when they are electronically added together with the five main channels in the encoder, they may produce less than desirable results at certain frequencies. For this reason, it is recommended that the LFE signal not be used in a Dolby Pro Logic II downmix unless it contains unique information that is not repeated in any of the five main channels."
Is your choice, possible artifacts/lose unique info (my choice is don't include LFE).

Quote:

Originally Posted by iceborne

"For downmix you ever have to use DRC because more acoustic power will be concentrated in less channels!"
what is frank trying to say? not use dynamic range or to use dynamic range on downmix?

I think not use DRC. But isn't my opinion.

You must use DRC to decrease the original signal Dynamic Range if:
1) The source have a wide Dynamic Range (dialogs at -30 dB and explosions at 0 dB).
2) You have a bad/normal audio equipment without support for this wide range (noise with dialogs or explosions distorted) or you don't want disturb the neighbors.
3) You want reencode to a format without support for DRC in play time (mp3, aac, ...)

The DRC is independent and must be do before any downmix.

JoeBG · 17th February 2006, 13:04

Hi,

upmix to 5.1 aac works great now for me.

Many thanks.

What not works is AC3. Do I need any special plugins? ffmpeg or something like this? Can someone give a small HowTo? Thanks

dimzon · 17th February 2006, 13:21

Quote:

Originally Posted by JoeBG

Hi,

upmix to 5.1 aac works great now for me.

Many thanks.

What not works is AC3. Do I need any special plugins? ffmpeg or something like this? Can someone give a small HowTo? Thanks

You must place ffmpeg.exe into BeHappy folder
Warning! Some ffmpeg versions has buggy stdin support - such versions doesn't work with BeHappy. This version works fine:

Code:

ffmpeg version CVS, build 3211776, Copyright (c) 2000-2004 Fabrice Bellard
  configuration:  --enable-theora --enable-mp3lame --enable-libogg --enable-vorbis --enable-faad --e
nable-faac --enable-xvid --enable-x264 --enable-mingw32 --enable-a52 --enable-dts --enable-pp --enab
le-gpl --enable-memalign-hack --enable-amr_nb --enable-amr_wb
  built on Sep  9 2005 04:00:51, gcc: 3.4.4 (mingw special)

NorthPole · 18th February 2006, 02:29

I found some additional information on upmix profiles from a couple of other threads

http://forum.doom9.org/showthread.ph...997#post442997

http://forum.doom9.org/showthread.ph...760#post558760

So I added them to the upmix extension:

Code:

<?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">
  <AudioDSP UniqueID="9579E57B-2D27-4583-99A4-921718E25B41">
    <Plugin>
      <MultiOptionDSP Type="BeHappy.Extensions.MultiOptionDSP, BeHappy">
        <TitleFormatString>5.1 Upmix - {0}</TitleFormatString>
        <ScriptPrologue>
	# Store clip in variable
	Stereo_{2} = convertaudiotofloat(last)
        </ScriptPrologue>
        <Option>
          <Name>Audio with mostly dialog (ie. Comedy, Drama)</Name>
          <Value>
	# Profile to use with audio sources that have mostly mono content. 20ms delay and -3db attenuation on surround
	# Note: the center channel is very weak for this profile
	front_{2} = stereo_{2}.soxfilter("filter 20-20000")
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.794,-0.794)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.794,-0.794)
	cc_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),fl_{2},1,-1),mixaudio(front_{2}.GetRightChannel(),fr_{2},1,-1),0.224,0.224)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.596")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.562,-0.562)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.562,-0.562)
	sl_{2} = DelayAudio(sl_{2},0.02)
	sr_{2} = DelayAudio(sr_{2},0.02)
          </Value>
        </Option>
        <Option>
          <Name>Audio with a mix of sounds (ie. Action, Adventure)</Name>
          <Value>
	# Profile to use with audio sources that have a wider range of sound content. 20ms delay and -3db attenuation on surround
	# Note: General purpose profile
	front_{2} = stereo_{2}.soxfilter("filter 20-20000")
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.668,-0.668)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.668,-0.668)
	cc_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),fl_{2},1,-1),mixaudio(front_{2}.GetRightChannel(),fr_{2},1,-1),0.398,0.398)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.447")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.473,-0.473)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.473,-0.473)
	sl_{2} = DelayAudio(sl_{2},0.02)
	sr_{2} = DelayAudio(sr_{2},0.02)
          </Value>
        </Option>
        <Option>
          <Name>Gerzen Profile</Name>
          <Value>
	# Gerzen approach Profile modified with 20ms delay and some attenuation on surround
	front_{2} = stereo_{2}.soxfilter("filter 20-20000")
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.885,-0.115)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.885,-0.115)
	cc_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel,0.4511,0.4511)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.5")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.668,-0.668)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.668,-0.668)
	sl_{2} = DelayAudio(sl_{2},0.02)
	sr_{2} = DelayAudio(sr_{2},0.02)
          </Value>
        </Option>
        <Option>
          <Name>Farina Profile</Name>
          <Value>
	# Farina/Sursound approach Profile M=L+R, S=L-R, c=0.75, L' = (1-c/4)*M+(1+c/4)*S, C' = c*M, R' = (1-c/4)*M-(1+c/4)*S
	# also added with 20ms delay and some attenuation on surround
	front_{2} = stereo_{2}.soxfilter("filter 20-20000")
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
	fl_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.500,0.500),mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.500,-0.500),0.8125,1.1875)
	fr_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.500,0.500),mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.500,-0.500),0.8125,-1.1875)
	cc_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel,0.375,0.375)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.5")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.668,-0.668)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.668,-0.668)
	sl_{2} = DelayAudio(sl_{2},0.02)
	sr_{2} = DelayAudio(sr_{2},0.02)
          </Value>
        </Option>
        <Option>
          <Name>Multisonic Profile</Name>
          <Value>
	# Multisonic approach Profile modified with 20ms delay and some attenuation on surround
	front_{2} = stereo_{2}.soxfilter("filter 20-20000")
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),1,-0.5)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),1,-0.5)
	cc_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel,0.5,0.5)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.5")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.668,-0.668)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.668,-0.668)
	sl_{2} = DelayAudio(sl_{2},0.02)
	sr_{2} = DelayAudio(sr_{2},0.02)
          </Value>
        </Option>
        <Option>
          <Name>Sound On Sound Profile</Name>
          <Value>
	# SOS approach Profile with 20ms delay and some attenuation on surround
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
	fl_{2} = stereo_{2}.GetLeftChannel()
	fr_{2} = stereo_{2}.GetRightChannel()
	cc_{2} = mixaudio(stereo_{2}.GetRightChannel(),stereo_{2}.GetLeftChannel,0.5,0.5)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.5")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.668,-0.668)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.668,-0.668)
	sl_{2} = DelayAudio(sl_{2},0.02)
	sr_{2} = DelayAudio(sr_{2},0.02)
          </Value>
        </Option>
        <ScriptEpilogue>
 	# Return result
	MergeChannels( fl_{2}, fr_{2}, cc_{2}, lfe_{2}, sl_{2}, sr_{2})
	ConvertAudioTo16Bit()
        </ScriptEpilogue>
      </MultiOptionDSP>
    </Plugin>
  </AudioDSP>
</BeHappy.Extension>

JoeBG · 18th February 2006, 05:13

@ dimzon

I took the one from 15.02.2006 from celtic druid, works very good

@ northpole

I´ll try it immediatly

NorthPole · 19th February 2006, 00:59

Sorry about so many revision on this upmix extension...

This pretty much finishes what I have been able to come up with so here it is

Code:

<?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">
  <AudioDSP UniqueID="9579E57B-2D27-4583-99A4-921718E25B41">
    <Plugin>
      <MultiOptionDSP Type="BeHappy.Extensions.MultiOptionDSP, BeHappy">
        <TitleFormatString>5.1 Upmix - {0}</TitleFormatString>
        <ScriptPrologue>
	# Store clip in variable
	stereo_{2} = convertaudiotofloat(last)
	front_{2} = stereo_{2}.soxfilter("filter 20-20000")
	back_{2} = stereo_{2}.soxfilter("filter 100-7000")
        </ScriptPrologue>
        <Option>
          <Name>Balanced Center Channel Profile</Name>
          <Value>
	# Balanced approach L' = (L-R)*0.668  R' = (R-L)*0.668  C' = (L-L')*0.398 + (R-R')*0.398
	# Profile for general purposes to use with audio sources that have a wide range of sound content
	# Attempts to balance sound levels of L', R' and C' with common mono content in center only
	# 20ms delay and -2db attenuation on surround  SL' = (L-R)*0.531  SR' = (R-L)*0.531
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.668,-0.668)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.668,-0.668)
	cc_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),fl_{2},1,-1),mixaudio(front_{2}.GetRightChannel(),fr_{2},1,-1),0.398,0.398)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.500")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.531,-0.531).DelayAudio(0.02)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.531,-0.531).DelayAudio(0.02)
          </Value>
        </Option>
        <Option>
          <Name>Decreased Center Channel Profile</Name>
          <Value>
	# Decreased center approach L' = (L-R)*0.668  R' = (R-L)*0.668  C' = (L-L')*0.224 + (R-R')*0.224
	# Profile to use for older movies or videos with mostly mono content. 
	# Same as balanced approach but with significally reduces C' power
	# 20ms delay and -2db attenuation on surround  SL' = (L-R)*0.531  SR' = (R-L)*0.531
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.668,-0.668)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.668,-0.668)
	cc_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),fl_{2},1,-1),mixaudio(front_{2}.GetRightChannel(),fr_{2},1,-1),0.224,0.224)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.596")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.531,-0.531).DelayAudio(0.02)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.531,-0.531).DelayAudio(0.02)
          </Value>
        </Option>
        <Option>
          <Name>Increased Center Channel Profile</Name>
          <Value>
	# Increased center approach L' = (L-R)*0.596  R' = (R-L)*0.596  C' = (L-L')*0.473 + (R-R')*0.473
	# Profile to use with audio sources where quieter dialog or lots of loud sound effects.
	# Same as balanced approach but with slightly higher C' power and slightly lower L' and R' power
	# 20ms delay and -2db attenuation on surround  SL' = (L-R)*0.473  SR' = (R-L)*0.473
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.596,-0.596)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.596,-0.596)
	cc_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),fl_{2},1,-1),mixaudio(front_{2}.GetRightChannel(),fr_{2},1,-1),0.473,0.473)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.473")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.473,-0.473).DelayAudio(0.02)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.473,-0.473).DelayAudio(0.02)
          </Value>
        </Option>
        <Option>
          <Name>Gerzen Profile</Name>
          <Value>
	# Gerzen approach  L' = (0.885*L)-(0.115*R)  C' = (L+R)*0.4511  R' = (0.885*R)-(0.115*L)
	# Modified by adding 20ms delay and some attenuation on surround  SL' = (L-R)*0.531  SR' = (R-L)*0.531
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.885,-0.115)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.885,-0.115)
	cc_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel,0.4511,0.4511)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.5")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.531,-0.531).DelayAudio(0.02)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.531,-0.531).DelayAudio(0.02)
          </Value>
        </Option>
        <Option>
          <Name>Farina Profile</Name>
          <Value>
	# Farina/Sursound approach  M=L+R, S=L-R, c=0.75, L' = (1-c/4)*M+(1+c/4)*S, C' = c*M, R' = (1-c/4)*M-(1+c/4)*S
	# Modified by adding 20ms delay and some attenuation on surround  SL' = (L-R)*0.562  SR' = (R-L)*0.562
	fl_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.500,0.500),mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.500,-0.500),0.8125,1.1875)
	fr_{2} = mixaudio(mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.500,0.500),mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.500,-0.500),0.8125,-1.1875)
	cc_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel,0.375,0.375)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.5")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.562,-0.562).DelayAudio(0.02)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.562,-0.562).DelayAudio(0.02)
          </Value>
        </Option>
        <Option>
          <Name>Multisonic Profile</Name>
          <Value>
	# Multisonic approach  L' = L-(0.5*R)  C' = (L+R)*0.5  R' = R-(0.5*L)
	# Reduced above formula by 10% to eliminate potential clipping on L' and R'
	# Modified by adding 20ms delay and some attenuation on surround  SL' = (L-R)*0.596  SR' = (R-L)*0.596
	fl_{2} = mixaudio(front_{2}.GetLeftChannel(),front_{2}.GetRightChannel(),0.9,-0.45)
	fr_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel(),0.9,-0.45)
	cc_{2} = mixaudio(front_{2}.GetRightChannel(),front_{2}.GetLeftChannel,0.45,0.45)
	lfe_{2} = ConvertToMono(stereo_{2}).SoxFilter("lowpass 120","vol -0.5")
	sl_{2} = mixaudio(back_{2}.GetLeftChannel(),back_{2}.GetRightChannel(),0.596,-0.596).DelayAudio(0.02)
	sr_{2} = mixaudio(back_{2}.GetRightChannel(),back_{2}.GetLeftChannel(),0.596,-0.596).DelayAudio(0.02)
          </Value>
        </Option>
        <ScriptEpilogue>
 	# Return result
	MergeChannels( fl_{2}, fr_{2}, cc_{2}, lfe_{2}, sl_{2}, sr_{2})
	ConvertAudioTo16Bit()
        </ScriptEpilogue>
      </MultiOptionDSP>
    </Plugin>
  </AudioDSP>
</BeHappy.Extension>

Changes:
- Tweaked most of the surround levels to try and match the particular profile.
- Nuked the sound on sound profile (the last one) because it was just the alot of same sound from the L, R & C (not much channel or dialog separation).
- Changed the names on the first 2 profiles to better explain what they are trying to do.
- Added another profile to increase the center channel levels.
- Added some notes, etc.

Thanks again dimzon for this slick audio convertion tool.

dimzon · 19th February 2006, 12:40

@NorthPole
Thanx for Your investigations.
I just wrote a tool, but You and tebasuna51 fill it life

By the way - future of BeHappy
I'm really waitng when MeGUI refacoring done. Probably BeHappy will be assimilated by MeGUI completly (currently it share same code but MeGUI has subset of BeHappy functionality)

JoeBG · 19th February 2006, 19:54

Quote:

Originally Posted by dimzon

@NorthPole
Thanx for Your investigations.
I just wrote a tool, but You and tebasuna51 fill it life

By the way - future of BeHappy
I'm really waitng when MeGUI refacoring done. Probably BeHappy will be assimilated by MeGUI completly (currently it share same code but MeGUI has subset of BeHappy functionality)

Please leave it as a standalone tool

dimzon · 19th February 2006, 20:29

Quote:

Originally Posted by JoeBG

Please leave it as a standalone tool

Why? Why does not combine them with MeGUI (preserving BeHappy functionality/flexibility)

JoeBG · 20th February 2006, 20:08

Quote:

Originally Posted by dimzon

Why? Why does not combine them with MeGUI (preserving BeHappy functionality/flexibility)

OK, this makes sense.

I just want to be able to use it without MeGUI for quick encoding decisions.

dimzon · 1st March 2006, 14:45

Quote:

Originally Posted by tebasuna51

The solution is make alternate curves with normalized output. I'm work about this.

Hi! Any progress with it?

14th February 2006, 19:44	#82 \| Link
JoeBG stupid Join Date: Sep 2004 Location: Cologne Posts: 638	I cant get work - upmix (is it still not supportet) (invalid arguments to function "SuperEq" - normalize (nothing happens) Do I need some special software packages? __________________ cu Joe ------------------------ freedom is just another word for nothing left to loose

16th February 2006, 03:03	#87 \| Link
iceborne Registered User Join Date: Aug 2003 Posts: 43	does downmixing dts to DLP II wav really work in behappy? wanted to know cause i like to trancode DLP II wav to itunes's aac format. could you also clarify between DPL II and DPL II(lfe)? which one is equivalent to "-azid( -s dplii -c normal -L -3db )"? Last edited by iceborne; 16th February 2006 at 05:51.

16th February 2006, 17:53	#88 \| Link
tebasuna51 Moderator Join Date: Feb 2005 Location: Spain Posts: 6,915	DPL II is equal to "-azid( -s dplii )" (copied matrix). DPL II(lfe) equivalent to "-azid( -s dplii -L -3db )". The "-azid( -c normal )" switch is for Dynamic Range Compression not yet implemented in BeHappy (problem with compand Sox Filter and Normalice see: http://forum.doom9.org/showpost.php?...5&postcount=60 ). If you have ffdshow like default directshow filter to play dts files you can use DirectShow input instead NicDtsSource, and use ffdshow Audio Decoder Configuration to set: - Codecs -> Dtd - libdts, Dynamic range compression (Checked) - Mixer -> Dolby Prologic II, LFE (Unchecked) -Output -> Output sample - only checked 16 bit integer, Don't use WAVEFORMATEXTENSIBLE ...(Checked) This is equivalent to "-azid( -s dplii -c normal -L -3db )". After use the Normalice DSP function in BeHappy and select your desired encoder. I test this method with ac3 source, I hope works also with dts. Edit: I make a test with dts source: - Same volume output with NicDtsSource and with ffdshow (drc checked and drc unchecked) (?). - The output length with NicDtsSource is 2:06.688 (like input) and with ffdshow is cut at 1:50.851 (?) Last edited by tebasuna51; 16th February 2006 at 18:54.

17th February 2006, 13:04	#91 \| Link
JoeBG stupid Join Date: Sep 2004 Location: Cologne Posts: 638	AC3 question Hi, upmix to 5.1 aac works great now for me. Many thanks. What not works is AC3. Do I need any special plugins? ffmpeg or something like this? Can someone give a small HowTo? Thanks __________________ cu Joe ------------------------ freedom is just another word for nothing left to loose

18th February 2006, 05:13	#94 \| Link
JoeBG stupid Join Date: Sep 2004 Location: Cologne Posts: 638	@ dimzon I took the one from 15.02.2006 from celtic druid, works very good @ northpole I´ll try it immediatly __________________ cu Joe ------------------------ freedom is just another word for nothing left to loose

14th February 2006, 20:05	#83 \| Link
tebasuna51 Moderator Join Date: Feb 2005 Location: Spain Posts: 6,915	- Upmix is in experimental stage by NorthPole, and need Sox Audio Effect Filter ( http://forum.doom9.org/showthread.php?t=104792 ), see the post: http://forum.doom9.org/showpost.php?...8&postcount=66 - Normalize works fine (amplify the signal until the required level).

19th February 2006, 12:40	#96 \| Link
dimzon BeHappy/MeGUI developer Join Date: Oct 2003 Location: Moscow, Russia Posts: 1,727	@NorthPole Thanx for Your investigations. I just wrote a tool, but You and tebasuna51 fill it life By the way - future of BeHappy I'm really waitng when MeGUI refacoring done. Probably BeHappy will be assimilated by MeGUI completly (currently it share same code but MeGUI has subset of BeHappy functionality) __________________ BeHappy - AviSynth-based audio transcoding tool Audio encoding via AviSynth On2 VP7 is great in quality but it is unusable for long-term video backup puposes! Sincerely Yours, MCPD/MCTS

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