8 bit, Mono Voice Over Clarity!

Ok. First thank you for any help you can provide. Second I am a very new (amateur) user but pick up pretty fast.
I have read a few threads and know that most will suggest I do not use an 8 bit recording. However this is for loading to a phone system that dictates the file format and can not be changed. I however, record initially at 16 bit and then downsampled to 8. The issue I am having is quality and clarity. I expect some loss and have been able to use Noise Capture & Reduction. My main issue comes when using Hiss reduction. With the dialogue box up and using standard reduction preset, I preview the audio. It sounds clear enough to accept. When I accept the settings and apply it to the file, it is NOT the same quality heard in preview mode. any ideas?

Maybe "Switch" is what you need?
http://www.nch.com.au/switch/
That'll enable you to convert your soundfiles to 8khz since it's not supported by STP. However, you'll need to save your files as 8-bit before.
BIAS Peak is able to do this too. If you don't already have STP, you may want to take a look at it instead.

Similar Messages

Mono voice over has 1 track only is this the proper way to do it?

My dialogue has 2 tracks (A1/A2), has well has my music (A3/A/4). My voice over, that was recorded in Mono is on just one track (A5), is this fine or should I have 2 tracks also for this voice over?
Thank you.

That's not really a great way to handle mono audio, in my opinion. When you double audio tracks, you're literally doubling the audio; all your levels will be jacked up, because you're superimposing sound. What Shane said is what I understand to be the correct way — as far as there's ever a correct way to do anything — to handle mono audio. You pan your left and right stereo tracks to the left and the right, but you keep your mono tracks panned center (or automate the pan for things like foley effects on moving shots, stuff like that).
Among other things, this makes doing your final mix a lot easier, since you've got only the number of tracks you actually need. Every track you add creates more work when you finish your sound.

IPod shuffle voice over for battery status problem

The voice over function for battery status doesn't work on my new Shuffle 4g. The other voice over functions do work.
I'm using Windows 7 (64bit). I have Itunes 10.5 . I down loaded the 64 bit version, voice over kit installer

Hi Imyman,
Apple has determined that the Apple Headphones with Remote included with the iPod shuffle (3rd gen) may fail under certain conditions. If your Shuffle has a certain serial number you can change your Headphones following this procedure:
http://www.apple.com/support/headphones/replacementprogram/
Hope this will solve your problem as it did for me.

Voice Over Recording Edit view vs Multitrack View

Hi Guys, need some advice here.
I have always recorded Voice Overs in edit view with Audition 3.0 (Windows 7 64bit)
Recently I purchased a Creative sound card (X-Fi Titanium HD) which sounds great and records 24/96Khz with great results using several different microphones, mainly a EV RE20.
I have the main outputs of my mixer left and right coming out using 1/4 inch cables and going into the soundcards Aux left and right inputs via RCA..
In Auditions audio hardware setup edit view, there is only a choice for either Aux left or Aux right but not a mix of both as a input. (using Aux left)
Not using the mic input on the card because I wanted to use the RCA ins (Left/Right) on the soundcard and the mic input is a single 1/8 input which would require another Y 1/4 to 1/8 cable and the fact that I have had issues with 1/8 inputs before.
So, when I record Voice Overs in edit view, I think I"m really getting only half the signal (Aux left) which means I need to up the gain on the preamp more than I would like to get a good level.
Recently, I have been recording in multitrack view using Track 1 Aux left and Track 2 Aux right and then doing a mixdown to mono which results in a nice strong level without having to boost the gain to much.
So I guess my questions would be am I in fact only recording half the signal in edit view by just using Aux left
and if that is true, should I just continue to record in multitrack view using the setup I decribed above.
Any advice would be greatly appreciated.

gdg444 wrote:
I have always heard that if you just raised the level using gain you take the chance of also introducing more noise in the signal and I was wanting to avoid that.
The short and sweet answer to this is that once you've recorded the signal, the amount of noise is fixed. But what you have to do to optimise this is called 'gain staging'. In the case of your preamp, this invariably means that you set the gain so that the clipping indicator (I take it that it has one?) indicates that you've got it just lower than it takes to light it on the loudest sound you want to record. With these signals present, you then set the input level control on your sound device (not the same as gain) so that it records these signals at around -5 to -10dB, in a controlled environment. In a non-controlled environment you'd allow a bit more than that. This gap between the -10dB and 0dB is called 'headroom'.
So basically, you are recording the best signal to noise ratio you can get from your preamp at the best level you can with your sound device. And that, by definition, is optimum.

How to change the "defualt language" on Voice Over?

I am a bit behind on the hardware (iPad 1) but only recently discovered the accessability options such as VoiceOver. When I gave the language rotors a go, after going around once it said "default language; Australian English." I do not see any point in this, because:
1. This iPad was not bought in Australia nor New Zealand
2. I live in New Zealand (although I can grudgingly understand NZ English doesn't have its own option)
3. Everything is set around UK English.
I've been searching Google and Apple support threads about this, but it seems only I am troubled with this. Does anyone know how to change that?
If I spoke honestly, I just don't like the voice and would rather not have it!

Here is a list of supported Voice Over languages: http://support.apple.com/kb/HT3562
To change the language used for Voice Control, tap Settings > General > International > Voice Control.

Can you change the size subject field on the voice over for like reading the news paper

i been playing around with the functions on my mac and i discoverd "voice over" a little annoying cause it just verbalizes everything, but i thought it would be kinda cool to have it read me the paper in the morning while i drink coffee and get ready for the day. however it only reads a small suject field one at a time .i was wondering if you can expand the size of the subject field to encompass more text if you know what i mean

HI,
You cannot change the PIc size of the Buddy on the Buddy List or in the Messages (Chat) window.
In the View Menu you can turn off the Buddy Pics (Depending if the Buddy List or Messages window is the "Front" one when you use that Menu).
10:10 PM Thursday; September 20, 2012
Please, if posting Logs, do not post any Log info after the line "Binary Images for iChat"
 iMac 2.5Ghz 5i 2011 (Mountain Lion 10.8.2)
 G4/1GhzDual MDD (Leopard 10.5.8)
 MacBookPro 2Gb (Snow Leopard 10.6.8)
 Mac OS X (10.6.8),
 Couple of iPhones and and iPad
"Limit the Logs to the Bits above Binary Images."  No, Seriously

Voice Over Tool Lockup

I upgraded to Lion last week and now when I select Voice Over Tool, I get the beachball and the application just freezes. Is anyone else experiencing this problem? I have to do a force quit to get out. Any suggestions?
Thanks

I did try removing the pref file, but it made no difference. I was already at 4.0.1.
I bit the bullet, offloaded all my files to an external hard drive and wiped my Mac clean. I then reinstalled Snow Leopard. Everything is working fine.
Not sure what the hang up with Lion is on FCE. All the video editing works fine. Just causes the entire system to lockup while using Voice Over. Not sure if it's related to FaceTime but sneaking suspicion says maybe.
Thanks for the help.

Crackling and glitches when recording voice overs in premiere pro cc

Hi,
I am trying to monitor my voice overs with minimal latency in premiere pro. I know that lowering the buffer size reduces latency, but adds strain to the CPU.
The problem that I encounter is crackling and weird glitch sounds when I drop the buffer size to 256 within the ASIO settings. I find it odd that my CPU can't keep up seeing as how the resource monitor doesn't show 'CPU Usage' higher than 10%.
When I raise the buffer to 512, the crackling and weird sounds disappear, but there is a slight delay present in the headphones.
There are no plug-ins or effects in use.
Would anyone know why this is happening, or if there's anything that can be done?
Thanks!
Equipment being used:
Audio Technica AT-2020 (USB Mic)
i7 3770k @ 4GHz
32gb RAM
NVidia Geforce GTX 670
Win 7 Pro
256gb SSD
2TB HDD

Yes. The procedure I went through was the following.
1) Add Mono Track (Audio 3)
2) Click on Mic Icon in Audio 3, in the Audio Mixer.
3) Click on Record Button. Watch it flash.
4) Press the space bar to record.

Voice Over in Premiere Pro CS5

Hi there.
A quick question about doing voice overs. I am recording using a Tascam US-122, with mono on. I am recording a voice over onto Audio Track 1, but it records to normal volume on the left side, and barely any volume to the right.
What am I doing wrong?
Thanks,
David

You're right, we weren't on the same page, and now I understand!
So I added a mono track and yes, there is only one levels display. I recorded, then copied and pasted it to a second mono track and panned one hard left, the other hard right, and I got equal levels in both channels!
Ok, so problem solved.
But one last question.... I'm still not really understanding what I'm doing when I record in mono. The screenshot below shows the playback of the mono recording... with one Levels display for the mono track, Audio 3, and two Levels displays in the Master. And 2 channels are playing back in the Master. So when I play it back now, I'm getting sound in both channels.
I'm not sure why creating a mono channel, as you suggested, ended up with a recording that plays back in 2 channels on the Master.
Thanks,
David

Voice over Wireless with Cisco phones 7921 and 7925

Hello experts,
I made an wireless audit for a company.
They have 2 WLCs 5508 in HA mode, with APs 2602 for indoor and 1552. Version of the WLC : 7.6.120.0
At the end of the day we noticed that the roaming between indoor and outdoor access points is sometimes failing and results to a complete disconnection of the wireless phone (7921 or 7925) from the network. When people go from the indoor to the outdoor area, there is no problem. The problem comes when people are coming from the outdoor to the indoor.
Also, on the WLC, the power lvl of the outdoor APs are set to 1 ... Is it good or not ?
My question is, is there any known issue about Voice over wireless with WLC 5508-7.6.120.0 with APs 2602 and 1552 ?
Maybe it should be better to upgrade to 7.6.130.0 ?
Thanks in advance,
Alexis

Normally yes.
Is there a way to troubleshoot what's going on with the phones ? Maybe a "show client detail MAC address* on the WLC ?
Here are some logs when the phones are losing the network :
*Dot1x_NW_MsgTask_4: Apr 09 12:44:21.320: #DOT1X-3-INVALID_WPA_KEY_MSG_STATE: 1x_eapkey.c:957 Received invalid EAPOL-key M2 msg in START state - invalid secure bit; KeyLen 40, Key type 1, client 00:24:d7:83:56:dc
*apfMsConnTask_6: Apr 09 12:28:32.668: #APF-3-VALIDATE_CCKM_REASS_REQ_ELEMENT: apf_utils.c:2506 Could not validate the CCKM Reassociation request element.Received Timestamp deviation > 1sec in CCKM Info Element from mobile. Mobile:4c:00:82:85:6e:e1, AP:1
*Dot1x_NW_MsgTask_1: Apr 09 12:26:53.964: #DOT1X-3-INVALID_REPLAY_CTR: 1x_eapkey.c:445 Invalid replay counter from client 74:26:ac:63:8c:a9 - got 00 00 00 00 00 00 00 03, expected 00 00 00 00 00 00 00 04
*Dot1x_NW_MsgTask_1: Apr 09 12:26:53.929: #DOT1X-3-INVALID_REPLAY_CTR: 1x_eapkey.c:445 Invalid replay counter from client 74:26:ac:63:8c:a9 - got 00 00 00 00 00 00 00 02, expected 00 00 00 00 00 00 00 04
*apfMsConnTask_4: Apr 09 12:24:34.959: #APF-3-VALIDATE_CCKM_REASS_REQ_ELEMENT: apf_utils.c:2506 Could not validate the CCKM Reassociation request element.Received Timestamp deviation > 1sec in CCKM Info Element from mobile. Mobile:78:da:6e:f6:5f:89, AP:5
*Dot1x_NW_MsgTask_0: Apr 09 12:22:30.217: #DOT1X-3-INVALID_REPLAY_CTR: 1x_eapkey.c:445 Invalid replay counter from client 4c:00:82:85:1d:68 - got 00 00 00 00 00 00 00 03, expected 00 00 00 00 00 00 00 04
*Dot1x_NW_MsgTask_4: Apr 09 12:22:30.206: #DOT1X-3-INVALID_REPLAY_CTR: 1x_eapkey.c:445 Invalid replay counter from client 4c:00:82:85:b3:ac - got 00 00 00 00 00 00 00 03, expected 00 00 00 00 00 00 00 04
*Dot1x_NW_MsgTask_4: Apr 09 12:22:30.186: #DOT1X-3-INVALID_REPLAY_CTR: 1x_eapkey.c:445 Invalid replay counter from client 4c:00:82:85:b3:ac - got 00 00 00 00 00 00 00 02, expected 00 00 00 00 00 00 00 04
*Dot1x_NW_MsgTask_0: Apr 09 12:22:30.167: #DOT1X-3-INVALID_REPLAY_CTR: 1x_eapkey.c:445 Invalid replay counter from client 4c:00:82:85:1d:68 - got 00 00 00 00 00 00 00 02, expected 00 00 00 00 00 00 00 04
*Dot1x_NW_MsgTask_6: Apr 09 12:22:29.672: #DOT1X-3-INVALID_REPLAY_CTR: 1x_eapkey.c:445 Invalid replay counter from client 78:da:6e:f6:14:2e - got 00 00 00 00 00 00 00 03, expected 00 00 00 00 00 00 00 04
*Dot1x_NW_MsgTask_6: Apr 09 12:22:29.638: #DOT1X-3-INVALID_REPLAY_CTR: 1x_eapkey.c:445 Invalid replay counter from client 78:da:6e:f6:14:2e - got 00 00 00 00 00 00 00 02, expected 00 00 00 00 00 00 00 04
*apfMsConnTask_3: Apr 09 12:19:22.098: #APF-3-VALIDATE_CCKM_REASS_REQ_ELEMENT: apf_utils.c:2506 Could not validate the CCKM Reassociation request element.Received Timestamp deviation > 1sec in CCKM Info Element from mobile. Mobile:4c:00:82:85:6e:e1, AP:5
*osapiBsnTimer: Apr 09 12:13:36.031: #LOG-3-Q_IND: spam_lrad.c:53542 The system is unable to find WLAN 2 to be deleted

Standard Voice Over Effects

Hello,
Still a bit new to Sound Track Pro 2 and there seems to be a massive array of effects available and I was wondering if some more experienced users could help me.
I am recording a voice over for a rolling news type segment with one mail narrator. Are there a few effects out there anyone can recommend that they would normally use for this style of audio, that kind of BBC news reader effect.
I am aware that my query is quite general but any advice would be appreciated.
Mac OS X (10.4.10) 8 Core Mac Pro & Intel MacBook

no I don't think so, the guy is news and media commentator Ray Snoddy - I've seen him on many occasions and that's just his normal voice.
On a live news programme like that he wil be mic'ed up by the studio sound op with a lavalier mic and the settings through the mixer will be set to get as true as possible. Levels will be taken prior to broadcast for each speaker but that will be likely only to be for volume.
On a live programme like that, with many contributors, there's no time for fine tuning anything else. Even with the pre-recorded inserts, the only tweaking will be for levels - it will be assumed that the item has been recorded correctly and only if there was some problem with the recording will any attempt be made to correct it.
Live news just doesn't have the time.
Obviously the studio will be acoustically appropriate to start with.

Voice over tips?

Hi Everyone,
I am currently trying my hand at creating voice overs. I have a reasonable mic/preamp setup plugged into one of the higher end Sony Vaio laptops so I already get a descent raw audio file. However, that's exactly what it is: raw. I would be really grateful if any of you could give me some tips on making the audio (spoken male voice only) sound more professional. Perhaps not quite as deep and resonant as the film trailors but something towards that direction. At the moment I am experimenting with the compander and then I normalize the file. This has given fairly good results.
Your help will be much appreciated.
Many thanks

Nothing beats an acoustically dead or well treated room for that closeness. I don't have it, but the Reflexion filter from Selectronics is pretty good. It's a half barrel of heavy duty damping material mounted around the mike.
When I use a dynamic mike, I point it not directly at the mouth but to the tip of my chin or even from the side. This way I avoid plosives like heavy B's and P's. It also rounds off treble gently, emphasizing the bass in your voice slightly. This kind of 'natural EQ'ing' is preferable to digital filtering afterwards IMO.
Inside Audition I always remove any noise that might creep in, like fan noises etc. I also roll off all bass below circa 70 hZ. You must listen to your recording to find the correct frequency; it shouldn't affect the sound at all. The reason I do this is to remove rumble and deep resonances that will not be audible on most computer speakers anyway. My recordings ends up in the MP3 format, and low frequencies tend to 'steal' a lot of energy from the compression algorithm. When there aren't any really low frequencies, you free up resources so that your VO is treated better. This is especially evident with hard MP3 compression.
If I'm not the last hand on the file I will deliver my files without any dynamic or EQ treatments except perhaps normalizing to -3 db. If I make the final file I actually use some hard limiting too. Max is at -3 dB, because I've found that laptop speakers especially can break up at anything above that. I push up the volume a bit to get more presence and intelligibility on poor speakers. It depends on your target audience.
Nevertheless, you need one of those rare 'million-dollar-voice' guys in to get that particualar voice. Technology can't do it alone. The attitude of the talent/speaker will make a huge difference. Don't sit hunchback'ed and talk, pretend you are a teacher (in a quiet class!) and you want to say something important to all, without shouting. Singers in the studio are often told to sing with a smile - however artificial - because it improves their voice. Similar tricks for VO. Things like that...

Voice-over : setting prefs for length of pre-run?

Where would I set the prefs for the pre-run for Voice-over? I don't need the five seconds, just want to start right away.
Also, I notice that when I bring the original media (the recorded voice-over clip) from the Capture Scratch bin it's coming in as a stereo clip (one side of which is empty). Where could I set THAT particular pref, so it just comes in mono, as was the original recording (the original clip on the track on the timeline).
Sometimes I trash the original voice-over clip on the timeline and have to recover it from the Capture Scratch bin.
All ears,
Ben

Hi Nick,
What are the patch panel audio buttons? Haven't heard of them before (or I use them all the time but don't know they are called that).
Thank you for the preroll info. I'll just start my record. Aha... and that's why there was so much header on my audio clips. Mystery revealed.
You wouldn't know, off chance, how I can stay plugged into the discussion forums would you? In the earlier version of the forums I could sign in in the morning, and still be signed in at the end of the day.
Now it seems that if I leave the room, or the laptop goes into sleep mode, I've got to sign in again. And if I'm chasing down a particular response, like yours, if I have to sign in anew it takes me to the front page of the discussion, away from the actual discussion I'm trying to follow.
B

No Levels in Voice Over Meter with Presonus Firebox Preamp

I've searched a bit through this forum and it seems this has been a problem in the past for several people. Has anyone figured out why the Presonus Firebox Preamp (using a condenser mic with phantom power)will register sound in Garage Band...Soundtrack Pro...etc, but not in Final Cut Pro 5.
Final Cut Pro's VO utility identifies the Presonus Firebox, but no sound comes through at all...even if you try to record...say a few lines...and then play it back...still absolutely silent.
Help!!!
Mr. Pants

Check that the sample rate is set the same everywhere:
Presonus dialog
Audio MiIDI setup
Sequence settings
Audio/Video settings- A/V Devices - Playback Output - Options
and finally, Toolbench Voice Over dialog.

Cannot do Voice-over

I cannot get either my bluesnowball USB mike or an analog audio mike to work in premiere pro CS4 for voice over recording. My system, windows vista home edition (32 bit) recognizes and loads the drivers for the USB mike, but when I try to record voice overs I get no audio, only static in the audio track. Adobe gave me some advice, which I followed, but I still cannot get it to work. My audio card is a Soundblaster X-FI Xtreme Gamer mounted in an HP Pavilion Ultimate (AMD PHENOM 9600 Quad-core processor, and an NVIDIA GEFORCE 8800GT (512MB)). Any help or someone with a similar problem? - Jean.

I've just switched to PPro CS4 after years of being a Vegas user ... because Vegas Pro 8 is @#$%%^&.
I'm trying to configure PPro to record my Rode Podcaster USB mic for VO's ... Adobe Soundbooth can record the mic and playback the file via my Audigy 2 card by simply configuring Preferences / Audio Hardware / using the default device 'Soundbooth 2.0 WDM Sound' and selecting the Podcaster mic from the list of available inputs in 'Settings' and selecting the output to go to the Creative Speakers.
That's all good.
BUT THE SAME THING doesn't work in Premiere Pro. I do the same setup in Preferences ... but the 'Premiere Pro WDM Sound' default device doesn't see the Podcaster (why NOT? the Soundbooth one can? and I can't use the 'Soundbooth 2.0 WDM Sound' default device in Premiere Pro ... Why can't PPros WDM sound device see the MIC if Soundbooth's can???) ...
So I browse around the forums and find a reference to someone else who can't get a simple audio input working in Premiere Pro for their USB mic ... someone suggests downloading and installing good old ASIO 4 ALL ... man, I used to use that with Cubase years ago when nothing worked ... anyway ... did the download ... (BTW ... 'ASIO 4 All' looks all grown up now, very professional ... the old George Bush graphic for the 'simple' options and Albert Einstein graphic for the 'Advanced' options are gone ... probably since Obama got in ... replaced by a fancy interface with a piano keyboard instead) ... anyway, all of a sudden using 'ASIO 4 ALL' PPro can see the Podcaster USB mic and I can record on the timeline ... that's they way it should be ... if Soundbooth can do it straight away ... why can't PPro do it??????
BUT WAIT THERE'S MORE! ... just when things looked like they were going to work ... THE PLAYBACK DOESN'T WORK!!!!! ... I can record from the USB mic ... BUT I can't route the playback to my Creative speakers ... because the playback defaults to trying to squirt the sound out of my ASIO 4 ALL driver and back into my Podcaster USB mic as soon as I hit play on the timeline ... I SELECT the Creative Speakers for playback and as soon as I hit play ... it changes back to trying to make my Podcaster mic into a speaker ... THIS IS CRAP!!!!
HEY YOU APPLE GUYS ... DOES 'FINAL CUT' ACTUALLY ALLOW YOU TO MAKE A DECENT VIDEO ... COS I CAN'T AFFORD A QUANTEL IQ SYSTEM JUST YET ...
Darren Powell ... already disappointed with PPro ...

8 bit, Mono Voice Over Clarity!

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