Follow Up...Vocal Mixing Approach

I've been kicking around possibilities in a different thread with some very helpful people
http://discussions.apple.com/thread.jspa?threadID=2386425&tstart=0
Now, I'd like to implement what AT THIS POINT, I think may be my best way to approach the issue of getting my vocal mixes good.
First, I am going to go back to using my mic the way I had in previous recordings. That is, using the large windshield that came with my SM7, that helped with a lot of plosives etc. Also, I will apply the bass roll off and NOT the presence boost. Presence boost seemed to give too harsh highs. I can EQ later if need be. Flat will be easier to start with. As far as pre tracking, this is about all i can do, since I know my recording levels and all are good, and i don't have any other equipment besides the SM7 and the DUET.
Next, I would like to know y'alls take on "NORMALIZE". I usually have not ever normalized my vocal tracks. The last one I had issues with a lot, I actually DID normalize, So, perhaps I WON'T in the future. I'm gonna open up a meter for monitoring. Prolly the Multimeter. I'm going to go in and for the sake of time, only do some manual edits to the volume envelope for the REAL bad guys, those that stand out clearly in my ears. Maybe do a quick check for anything that is REAL quiet too, boost that a little manually. Problem is I don't know how much to boost or lower. Can I see with the meter what the overall average db is, and see if I have a loud part that hits, say -6db and the average is around -12db, so I'd just lower that part by 6db?? and vice versa for quieter parts.
Now, the fader is controlled by the volume envelope so if I do automation I will lose control of the fader for my levels, so should I insert GAIN plug or just use an output gain on one of the inserts I'm gonna use anyways. Either way, at this point I will set the relative levels. Just a quick setting to start off with.
Now I believe I can apply this plugin
http://sounds.wa.com/audiounits.html
Thanks Ericksimon!!
Looks like a nice tool, similar to what I was looking for but not too cumbersome. I have not gotten a chance to try it out or see how to use it properly.
So at this point, I think I should have a really steady volume level going on.
Now I am not sure about the whole compression thing. I have come across several options. First though, do I want to do any EQing at this point??? before I add compression or after?? There are two points I would want to use EQ, one is to tame any out of control frequencies and 2 would be to shape the overall tone of the vocal. I suppose I could EQ for any bad frequencies first, then do compression and whatnot, then EQ at the end to slightly shape the sound if need be.
Anyways, compression. The Multipressor worries me a bit. I've used it with nice results on instruments and sampled audio, but not on vocals. Also, the downward expansion looks cool. I don't usually have any problem with too much noise, but I guess that's a problem i just never knew I had!!! I just try to track with as little gain as possible while still achieving a usable recording level. Then I get increase the gain later without having a lot of noise on the recorded track. Also, the multipressor would address my Highs in my voice jumping out at times. If I could just compress the Highs and not so much the lower frequencies. That might give me better results than JUST compression the whole spectrum equally.
The OTHER option, maybe, would be to use the DeEsser. I can compress a specific frequency range, but I don't think it has multiple bands. So, I couldn't really get any compression going on for the other bands. But I was thinking about the DeEsser in tandom with a Noise Gate instead of the Mulitpressor. This may be a little easier, but I think I might want to have a full spectrum compression, but be able to compress certain "trouble" frequencies more or less.
So I think I should be done at this point as far as levels and all that go, perhaps if I was to use EQ to color my track a bit, but that is gonna fall in with applying REverb and Delay or whatever else I want to add flavor. WHICH IS A WHOLE nother thread. After mixing about 13 tracks in logic now, I still can't find a setting that I'm comfortable with for my voice. Every recording seems different so every time i have to do something quite different. And every time i feel like it's the first time!!!
Anyways, how does my approach at this point look??
Holy crap,.....I didn't realize I had written that much. My god, If anyone replies you are truly a saint.

Bee Jay wrote:
Next, I would like to know y'alls take on "NORMALIZE".
Never do it. Absolutely pointless.
I think if it doesn't hurt anything I WILL because I think it helps me when doing manual edits so that I have a point of reference that is somewhat consistent throughout multiple audio regions
Now, the fader is controlled by the volume envelope so if I do automation I will lose control of the fader for my levels,
This is a good point - in general, send the output of a channel to a bus/aux channel. You can now automate one, and use the fader on the other to offset the levels - doesn't really matter which, although I prefer to automate the channel fader to smooth out the signal before hitting the aux wher the compressor is.
I am not real familiar with aux channels, I've tried using them but didn't understand the send amounts and how it was affecting the signal. If I put the send amount at 100 does 100% of the signal get passed to the fader then. So in your example, I would definitiely want to set at 100. As of now I simply used the gain control on the compressor or whatever plugin was last in the chain, I think that should not be any different in the sound....
The Multipressor worries me a bit. I've used it with nice results on instruments and sampled audio, but not on vocals
I wouldn't use it unless there is a definite reason to use it. At this point, you don't sound sure why you are using, and thus are unlikely to use it effectively. So don't over-complicate and use something simpler - there are plenty of good sounding compressors around, if you don't like Logic's.
well what I did is looked at the problem frequencies that I had corrective EQ'd and made that one band,,,,from say 700-1200, then made the other two bands as 0-700 and 1200-18000 or whatever. I didn't touch the low band, but I applied compression slightly different for each one of the other bands. I think it turned out much better than I expected, but I really don't know that I needed much compression anyways......the manual automation, rider plug and EQing had the signal going to the compressor pretty good already. It's amazing how much better a compressor sounds when the signal coming in isn't pure garbage!!!
I just try to track with as little gain as possible while still achieving a usable recording level.
What levels do you record at?
I believe the settings on my Duet are at about 56-58 usually when I record. This puts the input signal usually bouncing around between -24 to -12 db. I heard that was a good range and leaves lots of headroom. I use Ind. Monitoring level to crank it up so I can record at this level, otherwise it's too quiet to hear over the beat. I like to perform with lots of volume. These settings seem to work well, and NO i don't have a noise problem. Actually after alll the compression and everything was said and done. The "silent" parts between phrases or words were actually pretty much dead silent. So that seems to tell me my gainstaging is correct?
Also, the multipressor would address my Highs in my voice jumping out at times. If I could just compress the Highs and not so much the lower frequencies. That might give me better results than JUST compression the whole spectrum equally.
Maybe, but I still think it's overcomplicating things, although it's hard to say without knowing the recordings. if you have particular sections which are harsh, the common technique employed by the big guys is to split the vocal ("mult") into multiple channels for the appropriate parts, so you can EQ and treat the parts separately. Some producers will even go so far as to automate or mult individual syllables in a vocal phrase, although that is extreme.
yeah, haha, It already took something like 4 hours last night to do what I did. I broke down and manually automated nearly every phrase/syllable in the song!!! Boy, that was work. But, the results were quite apparent. Even taming the out of control stuff, it gave life to the stuff that used to get kind of lost in the mix. I been reading the manual to get some easier shortcuts and stuff for editing automation, so that should cut some time a bit. Also, I'm not gonna be recording much more backing tracks I don't think. If I do, I will probably just copy the automation from the first "Lead" track to the backing track. I have gotten a little more comfortable using reverb to help fatten up my vocals.
So I used manual automation edits. Rider Plugin. Corrective EQing. Multipressor. "Color" Eqing. And Finally SilverVerb.
Sounded much better than before. And really I only spent about an hour or 2 more than what I had originally spent battling with plugins that couldn't correct a bad input. So good in good out and you know the rest. And like I said. I accidentally erased most of the automation I did, then saved then quit. Didn't realize I did that. Oh well. I'm learning

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  • [Solved] System audio and microphone input, as one mixed stream?

    I'm using ffmpeg to record screencasts with this command:
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    It works really well for cases where I only need to record my voice ("Mic" capture works). However, there are situations where I would like to record both what I am saying, and what I am hearing. So, my question:
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    Last edited by Goran (2012-09-17 02:33:32)

    Since I don't have a Realtek ALC663 codec chip, the best I can do is provide guidance.
    While all the info you need is provided by the hda-analyzer program, it is confusing. What helped me (a bunch) to better visualize the "virtual wires" of my codec was to download it's datasheet.
    I found one for the Realtek ALC663 here:
    http://www.datasheetarchive.com/
    Just search on "alc663" in the search bar at the upper right, then I had to click on the PDF symbol in the search results, which then displayed just the first page of the datasheet. But by clicking on that first page, the whole document could be downloaded... at least that's how it worked for me. Otherwise, try something like using the "Advanced Search" function of Google to search for PDF files with some relevant search words.
    Then go to the Block Diagram page of that document. This is what really helped me understand/visualize the wiring. But to be redundant, this diagram should match the wiring shown in hda-analyzer... it's just more "human friendly".
    Notice the hexadecimal numbers that are inside the ovals (0Dh, 1Bh, etc). For all my different codec chips, these numbers have matched the Node IDs shown in hda-analyzer (and all other ALSA data on my systems), but that might not be true for all codecs, so be careful.
    Also notice that not all the "virtual wires" are shown. Just like in some electronic schematics, the document writers "cheated" by labeling a "virtual wire" and then using that label elsewhere in the document. For example, the "virtual wire" coming from Node 0Fh is labeled "Mono DAC". That "virtual wire" connects to the input of Node 17h on the wire labeled "MONO".
    So let's follow some signals... like the "front" two stereo channels...
    (1) The PCM digital audio signal will come in at Node 02h. This node seems to allow selection ("SRC"), convert from digital to analog ("DAC"), and volume control ("VOL") from -64 dB to 0 dB in 1 dB steps. This codec chip seems to support 44.1k, 48k, 96k, and 192k samples per second.
    (2) It then heads to Nodes 0Fh and 0Ch, but let's just follow the 0Ch mixer node. Where it then leaves with a label called "Front DAC" and connects to many other nodes (14h, 15h, 19h, and more).
    (3) But let's just look at Node 14h. The output (called "FRONT(Port-D)") would probably be connected to speakers if this was a laptop, or to jack(s) if this was a desktop. The input to this node has a mixer that mixes the "Front" and "Surr" channels, then it goes into an Input/Output amplifier ("I/Oa") before leaving the chip.
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    So... assuming that you wanted to loop the "front" signal path that we just followed back as input (capture)... let's follow that path...
    ALSA play to hw:?,? -> 02h -> 0Ch -> 14h -> ((22h -> 09h) or (23h -> 08h)) -> ALSA capture from hw:?,?
    So fire up hda-analyzer and verify that the path that you want is connected and unmuted and make the required changes if they aren't. Simple, right?... chuckle.
    In my case I used hda-analyzer to connect the front two stereo channels going to my "Line Out" into the capture input and found that I also needed to unmute some of the mixers/switches in that new path that were not available in alsamixer. Then I fired up alsamixer and played with it's controls until I got a good bouncy audio signal using:
    arecord -vv -f cd -V stereo -D plughw:0,0 /dev/null
    while simultaneously playing some tunes, to verify that it was working. Then told hda-analyzer to create a Python script of the changes. Then I used hda-analyzer to put the codec back to it's original condition and created another Python script.
    Now I had scripts to turn on the hardware loopback and to turn it back off again. Like I said in that previous thread, the changes will disappear on a reboot unless you jump through even more hoops.
    Obviously, you need to use the correct "plughw:X,Y" for your system (i.e. The correct sound card,device)... Hint: look at the output from "arecord -l". Like your codec, mine also had two capture devices to complicate matters, so I had to keep trying different alsamixer settings and/or "plughw:X,Y" parameters until things smoothed out.
    Now just fire up projectM-libvisual-alsa and enjoy the trip... or try your screencast quest.
    This is, by far, one of the hardest methods for loopback... but perhaps the most personally satisfying/frustrating, and with the fewest problems once it's working.
    Good luck.
    Last edited by pigiron (2012-09-17 00:05:18)

  • Dealing with Expired Certs in Mixed-Mode

    I have done a fair amount of research on this topic and while I have deployed mixed-mode clusters, I haven't had a situation quite like the one I need to contend with in the next couple of weeks. I am trying to be as prepared as possible and am looking for feedback on some procedures I am drafting.
    The situation:
    (4) node cluster (clustering over the WAN)
    TFTP is enabled on all nodes (that is going to change as a result of our assessment findings)
    Cluster is running in mixed-mode
    Most certificates on the Publisher node are expired
    tomcat cert
    ipsec cert
    host-name ipsec-trust cert
    call manager cert (callmanager.pem)
    CAPF cert
    CAPF trust cert
    One of the subscriber nodes is in the same boat as the Publisher node (they were deployed at the same time and were the first nodes in this cluster)
    The other two nodes (in a DR datacenter) have valid certificates  (until 2016) except for the publisher node server cert (which has expired)
    The publisher node and the subscriber node that has the expired certs were also installed without DNS being enabled (no domain and no DNS resolvers specified - therefore, I expect that DNS client was not enabled during install)
    It is worth noting the following:
    Customer enabled mixed-mode because one of the security folks got hot and heavy on encryption. However, they limited the scope to phones only. So, IP Phone to IP Phone == authenticated/encrypted. They have a Unity Connection system with secure ports and that is it. Gateways: no encryption. CCX, etc. == no encryption
    During discovery we also found that LSC distribution is fubar. Only a percentage of the phones are using LSC. Likely due to a flaw in the provisioning process. That will be addressed later.
    The version they are running is 6.1(3)  (base, no service releases)
    The goal: Get the present solution into a VMware environment running CUCM 9.1(2). Planning on doing the Jump Upgrade procedure (interim hop to 6.1.4).
    We found out about the certificate issues during our discovery phase. We have built in time to remediate the certificate issue.
    The plan (well, thus far). I am still pulling together my notes and trying to come up with a way to test an implementation plan off line so that I can avoid bricking the phones (they are spread all over north america).
    Here is the 10,000 foot view of the plan (obviously, the actually plan will be more detailed):
    Use BAT to disable phone security and uninstall LSC
    Security Profile mod
    Certificate Ops
    Reset phones
    DRS Back up
    Download/backup current certs
    Configure DNS
    set DNS domain name
    set DNS resolver (primary and secondary)
    Pub node:
    regenerate tomcat cert
    restart tomcat service
    regenerate ipsec.pem
    regenerate callmanager.pem
    regenerate capf.pem
    Sub node (repeat above)
    ?should we update the Subs not affected by the cert issue?
    Run the CTL client and update CTL
    Reboot servers
    Pub then Subs
    Phones will reset as a result of this process
    The customer has said that they are actually fine with the idea of going back to square one and start over with provisioning a secured (mixed-mode) cluster after the 9.1 upgrade. That would be great except that if I uninstall LSCs, change phones to non-secure, and use CTL client to change back to standard-mode, I still have the CTLs left on the phones. No way to bulk delete them in UCM. I am considering using something like UnfiedFX to help me get back to square 1. Right now, I consider this a plan B. Unless feedback to this thread and other research suggests a different tact.
    Thanks in advance for any assist.
    -Bill

    Hi William,
    You have a quite a few requirements here. Just to clear things up, there are two type of certificates, first is called "certificate trust", and the other is called "Certificate".  For the trust certificates such as Callmanager_trust you can just click on the certificate, make sure that it is expired, and then delete it. this has no impact on the phones. The other type of certificate is called "Certs", you will need to regenerate those certificates, This will regenerate the certificate and also recreates the new "CAPF-trust" or "CallManager-trust" certificates with new date/time ranges.
    Doing the above will not impact the phones are the services, however after regenerating the certificates, you will need to restart all the services related to this certificate, for example if you regenerate the tftp certificate, you will need to restart the tftp service on all the servers in the cluster. Same for the Callmanager and the Tomcat.
    Please note that whenever you regenerate the Call manager certificate, you will need to run the CTL client with the same Token you used when the server was changed to mixed mode.
    In General the below is the procedure to regenerate the certificate
    - log into the "Cisco Unified OS Administration" page of the publisher
    - choose Security>Certificate Management
    - click the link for the expiring certificate
    - click "Regenerate"
    - restart the service that uses the certificate
    That will regenerate the certificate on the publisher. Within the next
    10-15 minutes, the updated certificate will  be propagated
    to the subscribers.
    For more details you could refer to :
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/cucos/6_1_1/cucos/iptpch6.html#wp1040760
    http://www.cisco.com/en/US/prod/collateral/voicesw/direct_upgrade_procedure_for_cisco_unified_communications_manager_releases_6.1_2.pdf
    Hope this Helps!
    Regards,
    Karthik Sivaram

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