Hey there musicvid10! Thanks for the response! So it sound like you're saying that after I render a video, it should sound exactly like it does when I'm playing it in the Vegas timeline, is that correct?

My apologies, what do you mean by the problem could be with the player? I use Windows Media Player to play the video - do you mean the problem could be with Windows Media Player?

Thanks again, I really appreciate your help!

Check your player volume and sound card internal levels.
What are your render settings in Vegas?

Yes, what are your render settings?
For instance, the normal DVD Dolby AC3 preset renders low level audio, anemic for a DVD>TV final destination IMO. Otherwise the audio render should be exactly .. or very close to that of the timeline PB. Try a PCM audio render... normally a 48kHz/16 bit stereo WAVE file. <.wav>
As was stated, your player or soundcard could need adjustment.

Thank you both for the responses! My apologies, I should have mentioned - I have two main render settings, but I have experienced this problem with both settings.

I always render using the Sony AVC/MVC Codec. Sometimes, I will use the 'Internet 1920x1080-30p' Template, which has the following audio settings: 128 Kbps, 48,000 Hz, 32 Bit, Stereo, AAC. Other times, I will use the 'AVCHD 1920x1080-60i' Template, which has the following audio settings: 192 Kbps, 48,000 Hz, 24 Bit, Stereo, AC3. It looks like the Audio format is in fact 'Dolby Digital AC-3 Studio.' Rraud, you mentioned that this format lowers the audio? Could that be the problem?

Rraud, I just tried rendering the audio using the 'Wave (Microsoft)' Codec; the settings were 48,000 Hz, 16 Bit, Stereo, PCM. And WOW, what a difference! Rendering in this format caused the audio to sound EXACTLY like the audio sounds on the Vegas timeline! A perfect match. Just to double-check everything, I re-rendered the same thing using the AVC Codecs I mentioned above, and sure enough, the music was about half as loud in the AVC-render compared to both the .wav render and the Vegas timeline. It's so strange - it's like the AVC renders recognizes that the music tracks are very loud, and it automatically lowers them drastically, while the .wav render just keeps it how it is.

Since this is the case, would that infer the the problem doesn't lie within my player and soundcard settings, but maybe within the Sony AVC templates I'm using?

Other question - would it be possible to render the video with the Sony AVC Codec I mentioned above while rendering the audio with the .wav render I mentioned above? (Since the .wav render worked perfectly, and since the Sony AVC Codec is great for my video)? I'm trying to figure out how to do that now, but I'm not having any luck (as mentioned, I'm an audio noob!).

Thank you both once again so much for your help, I really appreciate it!

AAC audio with AVC/h264 video is the norm, and is universally playable.

WAV or AC3 audio with AVC video won't play on very many delivery systems, including the web and older iThingys. It's more a matter of convention than practicality.

Why your AAC is lower volume than WAV is what we need to figure out. Can you make some new renders to confirm this is what is happening?

"Rraud, you mentioned that this format lowers the audio? Could that be the problem?".
- Not the format itself, the AC-3 pro default preset (at least in VP-9) is for theater PB . The normalization can be changed in the custom settings parameters.... That said, I do not use AC-3 for anything, other than for DVDs, where PCM audio causes a disc space issue.
I've never encountered any level issues with the MP4/AAC format, either in Vegas or third-party AAC encoders.

Thanks so much again for both of your replies! Unfortunately I am out of town at the moment away from my video-editing computer, but when I return next week I will definitely do some more tests by making new renders to confirm whether or not my AAC volume is lower than my WAV volume and report back (also, to specify, the problem primarily seems to be with the music, not the audio from my video - not sure why this is the case. It seems like when I render with AAC it balances the volume out - i.e. when the music should be getting much louder, it seems to automatically lower it so that it's just as loud as it was a moment ago). But in any case, I'll definitely respond early next week! Thank you both so much again, I sincerely appreciate your help!

Hello musicvid & rraud! Sorry for the delayed response, just got back home from out of town. I can now confirm that the AAC audio is in fact lower in volume (particularly with the music tracks) than the WAV renders. Still not sure why.

Rraud, you said, "The normalization can be changed in the custom settings parameters." I'm wondering if that's part of the problem. I am not familiar with the term 'normalization,' but does that adjust the audio to be a certain volume? Do you happen to know where in the custom settings parameters this could be changed? (I don't see anything but I could be missing it).

Again, it does seem like the AAC render tries to 'balance' the audio (it never lets the audio get too loud - it just 'stops' it from getting louder, even if the audio sounds extremely loud in the Vegas timeline). Thank you both once again so much for your help!

In the 'Dolby AC-3 Pro' encoder custom settings under the 'Preprocessing' tab, change the 'Dynamic Range Compression' Line and RF Mode settings to none. The encoded / PB level should then be close to whatever it's set within Vegas.. and about the same as a PCM/WAVE file. I don't recall the Dolby AC-3 "Studio' version has these user settable options.

FWIW, I normally set my overall mix levels to around -20 LUFS (integrated), a little louder than the 'recommended' EBU/ATSC broadcast loudness standard. This works out (for me) with light compression and peak limiting, peaks not exceeding -6.0 dBFS, VMMV depending on how heavy handed one is with the comp/limiters or lack thereof.

Hey rraud! Thanks for the detailed instructions on how to change the Dynamic Range Compression settings to none! That was very helpful to see, I followed those steps to check it out. Unfortunately, you are correct - these same user settable options do not appear to be available in the Dolby AC-3 Studio version. This is very unfortunate.

It seems very odd to me that, with all the options available in Vegas, I'm unable to render a Sony AVCHD video with audio other than the Dolby Digital AC-3 Studio version. I'm looking at my customizable template right now (under Sony AVC/MVC AVCHD 1920x1080-60i). Here are the only customizable options for Audio that I have:

- Audio format (the only option available is Dolby Digital AC-3 Studio)
- Sample rate (Hz): (the only option available is 48,000
- Bit rate (bps): (options available are 128,000 ; 192,000 ; and 256,000)
- Audio coding mode: (options available are 1/0 (C) ; 2/0 (L, R) ; 3/2 (L, C, R, Ls, Rs)

I have no idea what that last one even remotely means, haha. The other two options I have are that I can check off "Automatic gain control" or "Save data in Intel byte order (LSB first)."

Do you happen to know if changing any of these options help my case? If not, any other thoughts?

Thank you both so much again for the help!!!

By the way, I just noticed something that I haven't paid much attention to before. As mentioned, I'm very new at editing audio - I was previously unfamiliar with how to use the Master Bus, but I'm reading up on it a bit now. It looks like a decent amount of my audio is going over 0 db (I have some audio at 7.1 db). It seems like when I render with AC-3, it normalizes my audio so nothing goes over 0 db. When I render with WAV, the audio is just how it sounds in my timeline (with audio up to 7.1 db). Is there any way to turn this normalization off so I can have the audio sound just as it does in the timeline (even going up to 7.1 db)? Not sure if this new information helps or not! Thank you both once again so much!

> When I render with WAV, the audio is just how it sounds in my timeline (with audio up to 7.1 db).

I don't know how you are reading that, because it is IMPOSSIBLE to render a file exceeding 0dB... just like a bucket of water, once it is full, you cannot add any more.... Not even David Copperfield or Houdini could achieve that! :-)

"I'm unable to render a Sony AVCHD video with audio other than the Dolby Digital AC-3 Studio version."
Are you sure? I'm not familiar with Vegas 'studio' version, but AVC (.mp4) video normally uses the AAC audio codec, a lossy codec similar to AC-3 and MP3 but does not have the 'theater' normalization.
As I previously stated, only use AC-3 for DVDs... when PCM (.wav) audio creates a size problem.
As Christo correctly stated, it's impossible to render audio exceeding 0dBFS. However in some windows, Vegas and SF will additionally display the absolute over 0dBFS value in floating point 32bit. (the app's internal bit depth) The subsequent render would normally be 16 or 24 bit and subject to it's inherent 'limitations'.

If you render a > 0db signal in 32 bit float or 64 bit double, you will save all of the information. You won't get a hard clip like you would if you convert to a fixed point format.

This is what allows us to work internally - to a point - over 0db. (It makes the math easier to do the float to fixed point at the last stage before the hardware.)

However, since hardware is fix point d-a, it will not sound very nice. Some audio device drivers will accept floating point streams and allow gain reduction before they convert.

Not a good way to work, but it is there. Far better to keep a handle on your gain structure from start to finish.

Your final delivery should never exceed 0db on the master bus, and there are many schools of thought on what is the best, depending on whether you are sending to a mastering service and such. Your typical mastering engineer will want some head room to play with, and a broad dynamic range.

Peter

I should have stated that it is possible to render in the 32 bit FP mode and I occasionally do so for work in progress. But as Peter stated, playback would be a problem in most end users.
And in Cap C's case for a DVD, the DVD-Video specification only supports; PCM, DTS, MPEG-1 Audio Layer II (MP2), or Dolby Digital (AC-3) audio. And DTS is not a Vegas option AFAIK.

Wow, thank you so much everyone for all of the responses! I sincerely appreciate your help. In conclusion, it looks like I really just don't know what I'm doing when it comes to editing audio and that's my actual problem, haha. I have been video-editing for 10 years but audio-editing has always been my weakness.

It seems that my problem has been that many parts of my video go above 0.0 dB (and I'm assuming some kind of compressor or limiter was applied when I rendered my video to AC-3 perhaps?). It sounds like the solution is simply that I should try to make sure that my audio never goes above 0.0 dB?

I am completely unsure of how to use the Master Bus, and I'm not sure how to develop skills concerning how to have a good dynamic range in my videos (i.e. I want the audience to be able to hear all of the dialogue, but I also want the music to be loud enough to create emotion and suspense). I also have no idea how to use compressors or other audio-editing tools. Does anyone know of any good resources for learning how to edit audio like a pro? (i.e. how-to books, videos, etc.). I'd definitely be interested in checking them out!

Once again, thank you so much for all of the help!!

Your audio peaks should never exceed -1.0 dBFS. Anything above 0dBFS is distortion.
Your average (RMS) audio should be around -12 dB as a starting place.

I know this is a late post but it doesn't look like the issue was solved. The saddest thing about SVP is that it is designed to be open ended - to please as many people as possible. Issue: What works for me may not work for you!

Not once was the sound card type mentioned in this post. Like the all important best recommended video card for Vegas 12 (NVidia GTX570) there is also probably a recommended or dest fit sound card. If you think the standard Realtek audio is good enough you are mistaken. Vegas Pro is trying to do a good job but your OEM sound card is butchering it up. I have noticed the same issue with my renders and the problem was my sound card. You need to upgrade to a card made for music production that can handle what SVP (think Sound Forge) is trying to accomplish.

Think 24 bit, 44.1 kHz 192 bps or better. The Realtek that comes on many mobos is NOT adequate enough to meet the match of what SVP wants to put out. You probably have bit rates of 1,400 to 4K or more if using wav files. Realtek will color and compress your work in ways you don't want it to. OEM cards are designed to play DVD's, Blu-Rays and Gamer audio in that order. All finished and compressed and pre-coded.

You need to think of cards designed for music production, not music playing like Sound Blaster - it is for gamers. Been there. Name one SB card that says "Blaster for Video Editing". There are a million gamers for every A/V editor. That's where their money is.

And Peter: what sound card do YOU use? You say, "too many to mention" in your specs. Give me one that works well.

@VideoFreq

While a higher end audio device can make workflows better, I have heard - and created - many a find mix with even the lower end. It really depends on your signal path.

There are many blind tests out there that show even the most golden eared producers could not distinguish between the SB/Realtek audio hardware and devices costing $1000 of dollars.

To me the most important part of the chain is the speakers and amplifiers. $1000 sound devices connected to a $100 pair of speakers - well as they say, the weakest link in the chain.

Controlling levels and learning how to mix is the best set of 'equipment' you can have. Gain stages, FX, and other process can make or break a mix. No amount of expensive hardware will change that.

The SB/Reltek and related wont color anything if you don't use what is built in. However, just using the Wave or Mapper drivers can cause a hidden SRC. It is done by Windows. It is not terrible, but it can affect what you hear.

Devices that support ASIO are the best - in general - because the driver is independent of the OS and what it can do. I always recommend that a second ASIO supporting device is used. At the same time, keep the on board audio device active for Windows Sounds (don't point Windows at your expensive device, as you risk conflicts that are not worth the trouble.) I like to hook up some consumer speakers to the built in hardware so I can A/B my mix through what many end users will hear the audio on in the end.

One thing to be clear here:

The audio device has no affect on rendering (to a file) at all. It is out of the loop and not used during rendering. Makes it all that more important that you speakers and singal path are clean and reflect what you expect. Do not turn it up to "11". Make sure you really know your "room". That way when you do render you have a chance to produce what you heard in your final delivery. What you render to in format will affect what your results are like.

Personally - and this is NOT a recommendation - I like MOTU and RME. Not because they of their audio specs, but because of the array of i/o options they afford. They also tend to have rock solid ASIO drivers, and some really great utilities bundled. I also love the Lynx Studio hardware - $$$, but if you need the types of IO they offer, they are solid - and nobody does drivers better than they do.

Peter

In my experience, Echo cards are very stable and sound very good.
If you only need two in/out channels, the Echo MIA goes for around $100USD street. -10 and +4dB balanced/unbalanced in/out. Eight virtual output channels.

Not to sound silly, but my Zoom h4 recorder makes for a wonderful little audio interface. The drivers have been the best since the first revision. Has only essential bells and whistles.

I am way, way, WAY late to this game. And I don't have a solution, but I do have a thought. What the OP needs to do is separate reality from perception, to isolate where the problem might lie. To do this, I recommend a simple experiment. Mute all audio in the project except the music track. Render a short piece of the video where the volume levels of the music are sometimes louder, sometimes softer, as set by the audio envelope. Take that short piece you rendered and add it to the project. Put it on the timeline and look to see if there is visual evidence in the waveform of the changes in audio levels. If there is not, then something is wrong in the setup or rendering process within Studio. But if you do see visual differences in the waveform where the music was made softer then louder, play the clip from your timeline and see if it sounds like it should, with the levels going up and down at the right times and in the right amounts. If they do, then the problem is in playback -- your player or your system audio setup or something else -- but your Studio is not the problem.