Microphones and Input Sources
Microphone selection and proper usage are critical to getting quality audio. Without digging into microphone technology and techniques, here are some things to consider when selecting and using microphones for podcasting, conferencing, and live streaming applications.
- Keep the talker-to-microphone distance as short as possible. This will provide the highest direct sound from the talker, which will minimize pick up of other sounds in the room. Try to achieve a mouth-to-microphone distance less than 20 cm.
- Use directional microphones. While directional microphones do not “reduce noise,” they are less sensitive to off-axis sounds, minimizing pickup of room noise.
- Consider dynamic microphones. Dynamic microphones excel at close-talked speech applications and have less handling noise and sensitivity than condenser microphones. For speech applications this is beneficial.
- Use quality shock mounts for microphones. Many conferencing applications are done with the participant seated at a table or desk. Any mechanical noises on the table can get picked up by the microphone if a shock mount is not available.
- Talk across the microphone or use an effective windscreen. Close-talked microphones are susceptible to “plosives” which can cause distortion.
- While headsets are one of the easiest ways to get usable, intelligible speech audio, few consumer headsets offer truly professional audio quality. For professional results, use professional microphones (such as an SM7B) with the MixPre.
Headphones and Monitoring
Let’s keep this suggestion simple: always use headphones or earphones when participating in a Zoom session. Think of headphones as tiny loudspeakers that are directly at the ear, breaking the acoustical coupling path between a local microphone picking up speech and returning audio from the Zoom session.
When possible, do not use the built-in loudspeakers on the computer or mobile device to playback session audio.Why? When session audio from the loudspeaker is picked up by the local microphone it is sent back to the session where it becomes echo. While conferencing software like Zoom has built-in echo cancellation and echo suppression software, these processes are destructive to the audio signal. The result is degraded audio quality. Also, the interactivity between participants is diminished through the action of the echo suppression gating action. The higher the loudspeaker is turned up, the more echo cancellation that is required, reducing audio quality and interactivity. If you have no choice but to use open loudspeakers, turn their level down as low as possible.
Large, over-the-ear headphones sound great, but don’t look very good on camera. There are many low profile earphones that are not as visible on camera, such as the type used by on-air news talent. Since many conferencing software like Zoom can use different send (microphone) devices and speaker (headphones) devices, simple Bluetooth headphones can be used for monitoring while still using the MixPre to send microphone signals to the session.
Room Acoustics and Noise Sources
Recording studios and production stages have highly controlled acoustics which improves overall speech quality. Most office and home environments have not been designed with acoustical considerations and can often contribute to echoey, noisy audio that is distracting for the listener and reduces speech intelligibility. In general, the larger, noisier, and more reverberant a room is, the less suitable it is for good quality conference audio.
In some cases room acoustics can be improved by moving to a different part of a room. If one end of a room has heavy drapes versus smooth walls, choose to be closer to the drapes. Other simple ways of improving room acoustics is to add surface area to the room. Hanging heavy drapes or carpet pads are a common technique to make portable voice over booths. These add surface area and reduce acoustical reflections.
Noise transmission from adjacent spaces is far more difficult to minimize than reducing reverberation. Isolation from noise requires physical decoupling between spaces. If the room is not suitable for good audio, consider changing locations.
In cases where room acoustics are poor and there are no other options, there are still ways to reduce its impact. Digital Noise Reduction (below) is an option. Also, reducing the subject-to-microphone distance improves the speech-to-ambient-noise ratio. Additionally using directional microphones will reduce the pick-up of off-axis ambient noise.
Digital Noise Reduction – NoiseAssist on the MixPre
Noise reduction is a process to reduce acoustical and electrical noise in an audio signal. While noise reduction algorithms are available in many DAWS for use on pre-recorded audio signals, real time noise reduction has historically required highly specialized hardware. In 2020 Sound Devices introduced NoiseAssist, a noise reduction plugin for the MixPre II Series and 8-Series.
NoiseAssist works by listening to audio signals and selectively eliminating what it interprets as noise, whether electrical or acoustical. The amount of noise attenuation is selectable in dB. NoiseAssist is an adaptive algorithm that continuously evaluates the audio signal and effectively removes continuous, non-speech noise-like signals. NoiseAssist works best when it is applied to audio signals with a high percentage of direct audio, but can be effective in high noise situations.
While NoiseAssist is an effective tool to reduce noise, its magic can only work so well. Microphone placement, selection, usage, and room acoustics have a gigantic impact on getting great audio.
What is Sidetone?
Have you ever been on a phone call talking loud because you don’t know if the other person can hear you? If a bit of your own voice is sent back to you, you won’t talk as loud. That return audio coming back to you is known as sidetone. Sidetone was inherent in the design of early telephone systems. Today sidetone is deliberately added to communication systems so that a talker modulates the level of their own voice.
Sidetone is important for talkers in conferencing systems since it 1) helps reduce talker fatigue and 2) allows the talker to self-regulate how loud they speak. Sidetone is especially important when the talker is wearing headphones.
Audio Delay – Sound and Picture Synchronization
Audio delay is important in applications where the MixPre is used for audio but the video comes from a video capture device connected to the computer. In those cases the video is often delayed enough that lip sync is noticeably off. Delaying the local audio at the MixPre compensates for the delay from the video and lip sync is restored.
When using separate hardware devices, like the MixPre, for audio and video capture cards for camera signals, each sends their signal on its own discrete path. In most cases, video signals require more processing and arrive at the computer later than the audio. This results in a loss of lip sync, which is noticeable and objectionable to viewers in the session.
To match the delay between sound and picture in a Zoom session, the MixPre Series has selectable output delay. When using video capture cards with external video cameras start by dialing in an output delay of 180 milliseconds. Adjust as necessary to achieve good lip sync.
Bus: a signal path where audio signals are summed, or combined.
Local Audio: the audio signal generated by the conferencing participant or talker.
Session Audio: an audio signal from all participants of a conference. Session audio from applications like Zoom is sent back to a participant as a mix minus, without local audio.
Talker: the individual speaking in a conference.