In today's high-tech and fast-paced world, the tools we use make our jobs easier and more convenient. But be aware, fewer knobs and claims of "anyone can use it" are reason enough to worry.
Digital phone patch recommendations How many engineers have received a digital phone patch that, sonically, has little to be desired? I'll bet it's more times than the state of Florida can count accurately. When it comes to recording a voice or any audio source, there are many factors to consider. One is room acoustics. Studios spend untold thousands of dollars building rooms that have a neutral acoustic ambiance and excellent sound and vibration isolation. These studios add character to a recording and don't sound like back bedrooms or oversized closets. More and more, voice-over talents record themselves in their own homes, in recording spaces that lack a neutral ambiance. Back bedrooms offer little isolation from the outside world. This introduces unwanted rumble from ventilation systems and various other household noises. The method of delivery is most often via ISDN, a network audio transceiver. Unfortunately, too many of these recordings lack in sonic integrity.
Another source of sonic degradation is how the microphone and microphone pre amp are being used in conjunction with compression and equalization. Do not over-compress your recording, leaving it with little or no dynamic range. This leaves the recording sounding very dull. It is one thing to compress a musical instrument to make it "breathe" in time with the song, but this is not the effect you want on a narration track. When recording, it is usually best to lean toward light compression and equalization. You can always add more but you can't dial it back once it's been recorded.
Furthermore, be aware of overmodulating your microphone pre amp. If the signal is too hot at the input, then insert a pad at the microphone. An easy way to tell if the pre amp is being overmodulated is to see if the signal distorts or sounds very crunchy. Increasingly, some microphone pre amps are digital devices. They convert the analog signal from the microphone to a digital signal inside the pre amp. Working in the digital domain is a fantastic and very flexible method as long as everything is synchronous. This means that if your microphone pre amp is set to a sampling frequency of 48kHz, and that signal is passed through a digital mixer referenced to a sampling frequency of 48kHz, then when sending the signal to the digital phone patch device it too must be set to a sampling frequency of 48kHz. If any one of these devices is not set to the same sampling rate, or there is no sample rate conversion taking place, you will experience erroneous audio known as alias frequencies. You can recognize an alias problem by listening for harmonic distortion, clicks or pops. The recording will also sound very thin and metallic in nature. Avoid this problem by keeping all your digital devices synchronous to one another and properly convert to a single consistent sample rate.
It is in the best interest of the voice-over talent to record the highest quality sound possible. The sound quality of their voice is what sets them apart. If you can't record at a studio and must send the narration over a digital phone patch, then the suggestions noted should make for flawless recordings.
Open media framework preparation These days, more and more producers are purchasing and using nonlinear editing systems. Nonlinear systems are becoming much more affordable and easier to use for the everyday producer and so are proliferating in the post production world. Such systems allow video editors to export the audio tracks via Open Media Framework (OMF) to audio engineers. This efficient method of importing edit decision lists (EDLs) into a dedicated audio workstation eliminates hours of prep time in an audio studio, leaving more time for exciting and creative audio decisions. I'm sure you have heard the phrase "we will fix it in the mix." Yes, audio engineers have been fixing problems in the mix stage for years, and will continue to do so for many years to come. However, there are a few things that could help keep projects on time and under budget: Nonlinear editor operators should pay close attention to these audio details.
First of all, most field audio is recorded with two discrete audio tracks. One is usually a boom and the other a lavalier microphone. If that is the case, make sure the audio is loaded into the nonlinear editing system as two discrete audio channels. I know this sounds obvious, but it is a mistake most editors make. The audio from the source arrives at the mixing board on two discrete channels and the mixer sums the two channels together when loaded onto an editing system. Make sure the two channels are panned hard left and hard right and sent out a stereo bus to the input of the editor and everything will be great. Let's say your source is from a digital betacam and you are loading the audio into your editor digitally, but your editor has a sampling rate set to 44.1kHz. This will result in a sampling rate problem. You will need to sample rate convert the 48kHz source down to 44.1kHz or record your audio into the editor via the analog inputs. Always reference the tone of your source tapes before loading and be sure to match timecode frame rates. These are a few easy but often overlooked parameters to loading your audio correctly.
Second, when you prepare the tracks for the OMF, always deliver both sets of audio tracks per source if available. Let the audio engineer decide which sounds better for the final mix. Also, make sure to render all your fades and to add at least 60 frame handles for each edit. Be sure to match the timecode frame rate of the final event list with the timecode on the master tape. If all these details are met, you should have a perfect OMF.
All of the aforementioned details may sound obvious, but it's often the most obvious things that get overlooked.
