TimBoz
31st August 2005, 04:57
Stumbled on a post from 2003 about settings for FX (filters) to apply to channels created from stereo.
Obviously the end result depends on what is required.
The original thread was:
Orignal Post (http://forum.doom9.org/showthread.php?t=64561)
Further settings given by MaroonMike were:
Here is what I did using Goldwave starting with a 32bit stereo WAV file:
FL = 100% left channel, mono 32bit convert
FR = 100% right channel, mono 32bit convert
Center = 50% left and 50% right mix (I actualy used a command called
left/right channel average mix to do this), mono 32bit convert
LFE = 50% left and 50% right mix (left/right channel average mix), lowband filter at 120hz, volume reduction by 2%, mono 32bit convert
SL = 100% left channel, highband filter at 100hz, slight Hall reverb effect added, mono 32bit convert
SR = 100% right channel, highband filter at 100hz, slight Hall reverb effect added, mono 32bit convert
This sounded pretty good on my system. I will eventually hear it on others to see if it is a good template.
I answered the thread with:
My Method
Aimed at creating a general acceptable 5.1 production from stereo sound to play on an average DVD player with an average surround sound system in an average room - there is no such thing but you know what I mean - I don't want to spend a week perfecting the sound so much so that I am sick of it and never want to hear it played again.
It sounds okay and is based on MaroonMike's suggestions. I use Goldwave with a similar procedure which decompresses and then I use PCM only, compression is only done again for the final 5.1 ac.3 mix which I believe minimises quality loss.
To maximise video & audio sync I usually use Goldwave to extract audio from the source AVI and resave it at signed 16 bit PCM (CD quality?). I then use AVIMux to patch the original AVI and the CBR audio together which allows editing of the new AVI with sound in VirtualDub.
Using Goldwave to open the new AVI file I extract the audio. From here I:
Cleanup original audio:
Effect>Filter>NoiseReduction>Preset>Hiss removal
Effect>Resample to 48000khz (if not current format)
Effect>Volume>Maximise>Preset>Full Dynamic range
Save as STEREOMIX.wav PCM signed 16 bit stereo
The settings I use are below, though a little time spent creating presets with the Effect Chain tool will speed up the process though should be placed in the same order of processing.
Create 5 channels:
Left front & rear channels (from STEREOMIX.wav):
Front:
Effect>Stereo>ChannelMix>Preset>MonoLeft
Effect>Filter>Equaliser>Preset>Equal loudness
File>Save as>left.wav PCM signed 16 bit mono
Rear:
Effect>Filter>Low/Highpass>select Highpass radio button>initial cutoff 100hz
Effect>Reverb>OK
File>Save as>rearLeft.wav PCM signed 16 bit mono
Right front & rear channels (from STEREOMIX.wav):
Front:
Effect>Stereo>ChannelMix>Preset>MonoRight
Effect>Filter>Equaliser>Preset>Equal loudness
File>Save as>right.wav PCM signed 16 bit mono
Rear:
Effect>Filter>Low/Highpass>select Highpass radio button>initial cutoff 100hz
Effect>Reverb>OK
File>Save as>rearRight.wav PCM signed 16 bit mono
Centre channel (from STEREOMIX.wav):
Effect>Stereo>ChannelMix>Preset>Double Vocals
Effect>Stereo>ChannelMix>Preset>MonoMix>OK
Effect>Volume>Change>-8db
Effect>Filter>Equaliser>Preset>Equal loudness
Effect>Bandpass/stop>From:150hz To 12000hz
File>Save as>centre.wav PCM signed 16 bit mono
I then use Premiere Pro to mix the 5 wav files and auto-generate the LFE and create the ac3 5.1 audio. I have found that Surcode gets 'stuck' if the wav files are not of the exact same time length (to milli seconds) so use Goldwave to Trim or Add silence so that this is the case before using these 5 files with Premiere. Again Surcode has an upper limit on bitrate which is why 16 bit has been used as a format for each of the surround channels.
I then import the created ac3 5.1 audio to Encore, delete the original soundtrack and place the ac3 in its place for the soundtrack timeline.
Ideally I would like to remove vocals from all but the centre.wav, though I think this removes some of the quality/dynamics.
Obviously the end result depends on what is required.
The original thread was:
Orignal Post (http://forum.doom9.org/showthread.php?t=64561)
Further settings given by MaroonMike were:
Here is what I did using Goldwave starting with a 32bit stereo WAV file:
FL = 100% left channel, mono 32bit convert
FR = 100% right channel, mono 32bit convert
Center = 50% left and 50% right mix (I actualy used a command called
left/right channel average mix to do this), mono 32bit convert
LFE = 50% left and 50% right mix (left/right channel average mix), lowband filter at 120hz, volume reduction by 2%, mono 32bit convert
SL = 100% left channel, highband filter at 100hz, slight Hall reverb effect added, mono 32bit convert
SR = 100% right channel, highband filter at 100hz, slight Hall reverb effect added, mono 32bit convert
This sounded pretty good on my system. I will eventually hear it on others to see if it is a good template.
I answered the thread with:
My Method
Aimed at creating a general acceptable 5.1 production from stereo sound to play on an average DVD player with an average surround sound system in an average room - there is no such thing but you know what I mean - I don't want to spend a week perfecting the sound so much so that I am sick of it and never want to hear it played again.
It sounds okay and is based on MaroonMike's suggestions. I use Goldwave with a similar procedure which decompresses and then I use PCM only, compression is only done again for the final 5.1 ac.3 mix which I believe minimises quality loss.
To maximise video & audio sync I usually use Goldwave to extract audio from the source AVI and resave it at signed 16 bit PCM (CD quality?). I then use AVIMux to patch the original AVI and the CBR audio together which allows editing of the new AVI with sound in VirtualDub.
Using Goldwave to open the new AVI file I extract the audio. From here I:
Cleanup original audio:
Effect>Filter>NoiseReduction>Preset>Hiss removal
Effect>Resample to 48000khz (if not current format)
Effect>Volume>Maximise>Preset>Full Dynamic range
Save as STEREOMIX.wav PCM signed 16 bit stereo
The settings I use are below, though a little time spent creating presets with the Effect Chain tool will speed up the process though should be placed in the same order of processing.
Create 5 channels:
Left front & rear channels (from STEREOMIX.wav):
Front:
Effect>Stereo>ChannelMix>Preset>MonoLeft
Effect>Filter>Equaliser>Preset>Equal loudness
File>Save as>left.wav PCM signed 16 bit mono
Rear:
Effect>Filter>Low/Highpass>select Highpass radio button>initial cutoff 100hz
Effect>Reverb>OK
File>Save as>rearLeft.wav PCM signed 16 bit mono
Right front & rear channels (from STEREOMIX.wav):
Front:
Effect>Stereo>ChannelMix>Preset>MonoRight
Effect>Filter>Equaliser>Preset>Equal loudness
File>Save as>right.wav PCM signed 16 bit mono
Rear:
Effect>Filter>Low/Highpass>select Highpass radio button>initial cutoff 100hz
Effect>Reverb>OK
File>Save as>rearRight.wav PCM signed 16 bit mono
Centre channel (from STEREOMIX.wav):
Effect>Stereo>ChannelMix>Preset>Double Vocals
Effect>Stereo>ChannelMix>Preset>MonoMix>OK
Effect>Volume>Change>-8db
Effect>Filter>Equaliser>Preset>Equal loudness
Effect>Bandpass/stop>From:150hz To 12000hz
File>Save as>centre.wav PCM signed 16 bit mono
I then use Premiere Pro to mix the 5 wav files and auto-generate the LFE and create the ac3 5.1 audio. I have found that Surcode gets 'stuck' if the wav files are not of the exact same time length (to milli seconds) so use Goldwave to Trim or Add silence so that this is the case before using these 5 files with Premiere. Again Surcode has an upper limit on bitrate which is why 16 bit has been used as a format for each of the surround channels.
I then import the created ac3 5.1 audio to Encore, delete the original soundtrack and place the ac3 in its place for the soundtrack timeline.
Ideally I would like to remove vocals from all but the centre.wav, though I think this removes some of the quality/dynamics.