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Author Topic: Audacity (osx): How to mix two AUD sources?  (Read 20795 times)

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Offline intpseeker

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #60 on: August 28, 2016, 02:14:01 PM »
Definitely.  Though your first mix was most likely not "50/50."

The levels were probably somewhat different for each source and just loading them into Audacity and combining into one stereo track won't give you an even mix.

But to your question: at the very basic level, when you open files in audacity there is a gain slider for each file on the left side of the screen.  Its the one that just has the '+' and '-' symbols on both ends.  Above the 'L' and 'R' slider.

When listening to the show in Audacity just play with turning one source up or down and see how it sounds and what you like best.  Make sure the animated green bars in the top tool bar never show red.

ALL THAT SAID, have you been following the rules outlined in the beginning of this thread to align both sources?  The soundboard source and the omnis will almost certainly not be totally in sync and could cause an echo effect by the end of the set.  Thats a whole other can of worms that needs to be opened before you can get the right sound mix done.

I wasn't aware of/considering alignment at all, but when I began reading here, phasing came up and I checked the first kick drum spike and the two tracks were dead on. I just checked the last song,
and the spikes were still dead on. The soundboard and I were maybe 80-90 feet from the stage lip.
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Offline Life In Rewind

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #61 on: August 28, 2016, 02:23:07 PM »
Definitely.  Though your first mix was most likely not "50/50."

The levels were probably somewhat different for each source and just loading them into Audacity and combining into one stereo track won't give you an even mix.

But to your question: at the very basic level, when you open files in audacity there is a gain slider for each file on the left side of the screen.  Its the one that just has the '+' and '-' symbols on both ends.  Above the 'L' and 'R' slider.

When listening to the show in Audacity just play with turning one source up or down and see how it sounds and what you like best.  Make sure the animated green bars in the top tool bar never show red.

ALL THAT SAID, have you been following the rules outlined in the beginning of this thread to align both sources?  The soundboard source and the omnis will almost certainly not be totally in sync and could cause an echo effect by the end of the set.  Thats a whole other can of worms that needs to be opened before you can get the right sound mix done.

I wasn't aware of/considering alignment at all, but when I began reading here, phasing came up and I checked the first kick drum spike and the two tracks were dead on. I just checked the last song,
and the spikes were still dead on. The soundboard and I were maybe 80-90 feet from the stage lip.

You wont have drift - your 4 tracks are clock synced from the same recorder - drift is only an issue when you mix sources from different recorders.

Not sure how you got away without having delay issues - at 80 feet it should be VERY noticeable...be sure to zoom in a lot when checking...

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Offline voltronic

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #62 on: August 28, 2016, 05:24:07 PM »
Definitely.  Though your first mix was most likely not "50/50."

The levels were probably somewhat different for each source and just loading them into Audacity and combining into one stereo track won't give you an even mix.

But to your question: at the very basic level, when you open files in audacity there is a gain slider for each file on the left side of the screen.  Its the one that just has the '+' and '-' symbols on both ends.  Above the 'L' and 'R' slider.

When listening to the show in Audacity just play with turning one source up or down and see how it sounds and what you like best.  Make sure the animated green bars in the top tool bar never show red.

ALL THAT SAID, have you been following the rules outlined in the beginning of this thread to align both sources?  The soundboard source and the omnis will almost certainly not be totally in sync and could cause an echo effect by the end of the set.  Thats a whole other can of worms that needs to be opened before you can get the right sound mix done.

I wasn't aware of/considering alignment at all, but when I began reading here, phasing came up and I checked the first kick drum spike and the two tracks were dead on. I just checked the last song,
and the spikes were still dead on. The soundboard and I were maybe 80-90 feet from the stage lip.

You wont have drift - your 4 tracks are clock synced from the same recorder - drift is only an issue when you mix sources from different recorders.

Not sure how you got away without having delay issues - at 80 feet it should be VERY noticeable...be sure to zoom in a lot when checking...

Another way to check this is to put on headphones, pan one source hard left and the other hard right.  When they don't line up you'll start getting dizzy!  That's also a method I sometimes use to balance two tracks by ear.
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Offline achalsey

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #63 on: August 28, 2016, 10:52:14 PM »
Oooh, right.  My mistake.  Not drift by the end, but the two sources won't be aligned at the beginning.

The SBD feed should show up at your recorder basically immediately (electricity theoretically travels at the speed of light) while the mic source has to wait for the sound to travel through the air.  At 80-90' from the PA there definitely should be a discrepancy between the start of the SBD source and the mic source.

Offline danlynch

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #64 on: August 29, 2016, 12:15:39 AM »
General rule is one millisecond per foot between the sources, i.e. mics 50 feet from the stage needs a shift of 50 milliseconds between AUD (slower) and SBD (faster) sources.  But the number can be adjusted by other factors, mainly the introduction of delay into the system by the FOH.  Its best to check the among various points in each source (usually using drum clicks) to check for the correct shift amount.



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Offline morst

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #65 on: August 29, 2016, 05:09:22 PM »
and the spikes were still dead on. The soundboard and I were maybe 80-90 feet from the stage lip.
Were the mics on the stage lip? That would explain the lack of much difference between the SBD feed which is also made of mics on stage...

otherwise, 80 feet is about 71 ms according to my favorite relevant widget:
mac kids, check this out: https://www.apple.com/downloads/dashboard/calculate_convert/soundreference.html
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Offline intpseeker

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #66 on: August 29, 2016, 10:08:40 PM »
and the spikes were still dead on. The soundboard and I were maybe 80-90 feet from the stage lip.
Were the mics on the stage lip? That would explain the lack of much difference between the SBD feed which is also made of mics on stage...

otherwise, 80 feet is about 71 ms according to my favorite relevant widget:
mac kids, check this out: https://www.apple.com/downloads/dashboard/calculate_convert/soundreference.html

The mics were a bout 2 feet from the soundboard, 10 feet in the air.

I have cherry-picked spikes throughout the 2 hour gig, including a spike very near  the end of the last song, and zoomed in, the two tracks are dead on. I thought that maybe
I somehow took two soundboards or two audience pulls when  combined them, but when separated, one is dry and one has applause and chatter.

It could be that I'm not hearing the echo, but the last song sounds pretty good.
Mics:        Akg 451 eb A51's, ck-1's, ck-2's, ck 8's
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Offline Life In Rewind

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #67 on: August 30, 2016, 08:03:32 AM »
and the spikes were still dead on. The soundboard and I were maybe 80-90 feet from the stage lip.
Were the mics on the stage lip? That would explain the lack of much difference between the SBD feed which is also made of mics on stage...

otherwise, 80 feet is about 71 ms according to my favorite relevant widget:
mac kids, check this out: https://www.apple.com/downloads/dashboard/calculate_convert/soundreference.html

The mics were a bout 2 feet from the soundboard, 10 feet in the air.

I have cherry-picked spikes throughout the 2 hour gig, including a spike very near  the end of the last song, and zoomed in, the two tracks are dead on. I thought that maybe
I somehow took two soundboards or two audience pulls when  combined them, but when separated, one is dry and one has applause and chatter.

It could be that I'm not hearing the echo, but the last song sounds pretty good.

Your distance from the soundboard is not very important...its all about the distance from the soundmakers and the mics.

The only thing that could account for this is - the soundman knew what you were doing and added the appropriate delay to your feed...Id say that is unlikely.

Similarly - if there were a set of delay stacks - and they were relatively even with the board - and you got that signal?

The other thing I thought of - you accidentally used the internal mics? Doesn't sound like it - but it would account for the lack of delay.

You can add this delay with the 70D - but you would have known if you did...not something you can set accidentally
« Last Edit: August 30, 2016, 08:08:48 AM by Life In Rewind »
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Offline morst

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #68 on: August 30, 2016, 02:53:52 PM »
You can add this delay with the 70D - but you would have known if you did...not something you can set accidentally
Unless you set it a while ago, and forgot about it?!
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Offline intpseeker

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #69 on: September 01, 2016, 05:40:52 PM »
and the spikes were still dead on. The soundboard and I were maybe 80-90 feet from the stage lip.
Were the mics on the stage lip? That would explain the lack of much difference between the SBD feed which is also made of mics on stage...

otherwise, 80 feet is about 71 ms according to my favorite relevant widget:
mac kids, check this out: https://www.apple.com/downloads/dashboard/calculate_convert/soundreference.html

The mics were a bout 2 feet from the soundboard, 10 feet in the air.

I have cherry-picked spikes throughout the 2 hour gig, including a spike very near  the end of the last song, and zoomed in, the two tracks are dead on. I thought that maybe
I somehow took two soundboards or two audience pulls when  combined them, but when separated, one is dry and one has applause and chatter.

It could be that I'm not hearing the echo, but the last song sounds pretty good.

Your distance from the soundboard is not very important...its all about the distance from the soundmakers and the mics.

The only thing that could account for this is - the soundman knew what you were doing and added the appropriate delay to your feed...Id say that is unlikely.

Similarly - if there were a set of delay stacks - and they were relatively even with the board - and you got that signal?

The other thing I thought of - you accidentally used the internal mics? Doesn't sound like it - but it would account for the lack of delay.

You can add this delay with the 70D - but you would have known if you did...not something you can set accidentally

I've worked with the sound guy, and when I asked if I could plug in, he was running to the stage and yelled back to use the rca outs, which I did. So he didn't have a lot to do with it.

I went through (all) the menus for the mics and it doesn't seem that I used the internals, mic+phantom was picked, plus the internals buried in the bottom of my bag, etc.

A question: If I had used the internals, why wouldn't I still have the lag from 80 feet away?
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Offline Life In Rewind

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #70 on: September 02, 2016, 09:50:34 AM »

A question: If I had used the internals, why wouldn't I still have the lag from 80 feet away?

The speculation was: you ran your mics on your stand (1/2) - and had accidentally run the internal mics.(3/4)  Those two sets of mics would be physically very close to each other - and wouldn't have any lag.

But - doesnt sound like thats what happened...

At this point - its a mystery!

Post a sample if you can - see if anyone hears anything...
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Offline intpseeker

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #71 on: September 05, 2016, 02:34:13 PM »

A question: If I had used the internals, why wouldn't I still have the lag from 80 feet away?

The speculation was: you ran your mics on your stand (1/2) - and had accidentally run the internal mics.(3/4)  Those two sets of mics would be physically very close to each other - and wouldn't have any lag.

But - doesnt sound like thats what happened...

At this point - its a mystery!

Post a sample if you can - see if anyone hears anything...


Well...that's a 'duh' moment on my part.

As it turns out the singer in the band noticed a bit of 'digital delay' on two of her songs in the middle of the gig. Wanna bet it was about 75 milliseconds worth?

Sounds like a touch of reverb.

I'm back to the drawing board figuring out how to align the tracks.

Thanks to all for the info. This has been a fun learning experience for me.

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                 Busman modded R-4
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Offline morst

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #72 on: September 09, 2016, 05:42:41 PM »
Well...that's a 'duh' moment on my part.

As it turns out the singer in the band noticed a bit of 'digital delay' on two of her songs in the middle of the gig. Wanna bet it was about 75 milliseconds worth?

Sounds like a touch of reverb.
HAHA! Awesome, ya found the problem, now you can solve it!

Go to the tracks with the delay, and use the original peak, not the delayed peak. Typically, short delay (less than 100 ms or so) is used as a thickening, or doubling effect and the peaks of the delayed signal might be nearly as high, or just as high as the original source signal (her voice.)

Longer delays are often used as an echo effect, and those are usually not mixed as loud as the original signal. Longer delay sometimes has multiple "returns" of the signal, typically at a decreasing level, to mimic a natural echo. If your echo came back louder than your voice in nature, it would sound weird, so too in music. But at very fast rates like you might have, returning just as loud is actually an option.

Heads up from the sound mixing dept...  :cheers:
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Offline Gutbucket

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #73 on: September 13, 2016, 05:11:49 PM »
I'll recap for folks following along who may be scratching their heads a bit.

When mixing multiple sources, the final resulting "combined" recording will benefit most when:
1) All sources to be mixed have been checked for time synchronization and correctly synchronized with each other if required.
2) Appropriate adjustments are made to each source individually, before the sources are mixed together.
3) The sources are mixed using the most appropriate level relationship, which may be constant throughout the recording or varied over time.
4) Appropriate adjustments are made to the resulting combined file.

I'll follow up with some comments on each of these points..
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Offline Gutbucket

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Re: Audacity (osx): How to mix two AUD sources?
« Reply #74 on: September 13, 2016, 07:13:37 PM »
On point 1)-

Time synchronization issues manifest in two ways:

A) Time-offset difference-  a difference in time-alignment of the same events as represented across multiple the sources.  On playback, the time-offset will remain constant throughout the recording, unless there also happened to be a clock-rate difference between recorders (and unless the source or microphones are moved while recording).   If recorded using a single multichannel recorder or multiple devices which share the same clock data, the time-offset at that at the end of the recording will be exactly the same as the time offset at the beginning of the recording.  There is offset, but no "drift".

B) Clock-rate differencee- a difference between the nominal sampling rate at which the recordings were made, and the actual clock rates at which the two sources were sampled, when recording the same events to separate recorders that do not share the same clock-data.  For example- two recorders writing 48kHz audio files will sample the analog audio signals very close to, but quite exactly 48,000 times per second.  In actuality, the clock of one recorder is going to run ever so slightly slower than the other.  Like a time-offset, such a clock-rate difference should remain constant throughout the recording as well (hopefully! If not it would be far more challenging to correct*).  On playback, that slight difference between recording clocks manifests as a cumulative "additive" time difference when both sources are reconstructed to their nominal 48kHz rate using a single playback clock.  If the two sources are time-aligned at the start, and then played back using single playback clock, the slight difference between their actual recording sample rates will cause them to slowly "drift" out of synchronization as the file plays.  The longer the recording, the more out of sync the two sources "drift" by the end.

It's frequently the case that only a time-offset exists and needs correction. When sample-rate difference also exists, there is going to be a time offset as well.


For concert tapers, a time offset between sources is most commonly caused by the difference in the speed of sound through air verses an electrical signal through a wire.  Mics placed out in the audience further away pick up a sound significantly later than mics on-stage.  But there are other sources of time-offset too. Even when recording simultaneously to several recorders which are "clock-linked" so as to operate using the same sample clock, the different recordings may not start and stop at exactly the same time.  A recordist may need to push record and stop separately on each machine.  The resulting sources will have been sampled at exactly the same clock-rate, but they will need to be time-aligned later.  Time-offset is corrected for by offsetting one source in time with respect to the other.  That can be achieved by starting playback of each at exactly the right time, with a delay line, or most commonly and most easily by shifting one source along the playback time-line with respect to the other in the audio editing program.  By ear, it's is most easily done by listening to segments containing sparse speech (stage announcements and banter) or sparse sharp transients sounds like a single clap, snare or drum hit, or the like.  Zooming in on the waveforms, one may be able to see and align the peak of the transient.  Either way, it's easiest when the event being used stands out clearly from the background sounds, and difficult to impossible during dense passages.  Although there are exceptions, ordinarily the goal is to shift one source in time with respect to the other until a transient representing the same event occurs simultaneously, sounding clear and concise with no echo or blur.

A sample-rate difference occurs when recording to two or more separate, non-clock-linked recorders.  In that case, each recorder digitally samples the signals it is converting based on it's own local clock.  The clock-rate difference between clocks may be pretty close or not be particularly close, but in either case, if the recording runs long enough there will be some measurable difference between them, in addition to the time-offset difference between the starting and stopping points of the recordings.  This "drift" is sometimes so minimal it doesn't require correction.  Other times it causes a significantly audible effect which grows more severe until it becomes a blur and eventually an obviously discernible echo.    There are multiple ways of correcting for this.  One older, less than optimal method was to divide both sources into single song files, then individually align each song at it's beginning. The hope being that the time-offset "drift" is not significant enough to become audible before each song ends and the next begins.  The more appropriate and modern way of addressing the problem requires modifying the time-basis of one of the sources with respect to the other.  In effect, the length of one source is either "stretched" or "shrunk" to match that of the other.  In some software that's done via entering a new time-length for the file, after doing some work with a calculator to figure out the correct value. In others it's via entering a percentage or +/- value into a stretch/shrink function, by dragging the duration envelope of one source, or by manipulating a parameter value for the the source object.
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