Conference and Corporate AV Sound: A Practical Guide

In conference settings, the primary goal is clear, effortless speech reproduction. Unlike music, speech has a narrower frequency range (roughly 100 Hz to 8 kHz) and relies heavily on consonant clarity for intelligibility. Factors like reverberation, background noise, and poor loudspeaker placement can degrade the signal-to-noise ratio (SNR) and reduce the Speech Transmission Index (STI).

A good rule of thumb is to maintain an SNR of at least +15 dB throughout the listening area. This requires careful system design: selecting loudspeakers with controlled directivity to minimise reflections off walls and ceilings, and using DSP to apply equalisation that reduces muddiness (often around 200-400 Hz) and boosts presence (2-4 kHz). SSOUNDS point-source loudspeakers, for example, are engineered with constant-directivity waveguides that keep coverage consistent even in acoustically challenging rooms.

For boardrooms and small meeting spaces, ceiling-mounted distributed systems are common. Here, the key is to ensure uniform coverage without hot spots or dead zones. Using multiple low-profile loudspeakers with wide, even dispersion (e.g., 120° conical) and spacing them according to the 4:1 rule (distance between speakers should be no more than four times the listener distance) helps maintain consistent levels and intelligibility.

Corporate AV often calls for distributed audio—multiple loudspeakers placed throughout a space rather than a single central cluster. This approach is ideal for long, narrow rooms, open-plan offices, or spaces with low ceilings where a central system would create uneven coverage. Distributed systems can be either low-impedance (8Ω or 4Ω) for short cable runs or 70V/100V constant-voltage for long distances and multiple speakers per amplifier channel.

When designing a distributed system, consider the critical distance—the point where direct sound and reverberant sound are equal. Beyond this, intelligibility drops. By placing loudspeakers closer to listeners (e.g., in the ceiling or on walls), you increase the direct-to-reverberant ratio, improving clarity. SSOUNDS offers a range of compact, high-output loudspeakers suitable for flush-mount or surface-mount installation, with built-in DSP presets that optimise for speech or music.

For larger conference halls or auditoriums, a hybrid approach may be best: a central left-right line array or point-source cluster for the main stage, supplemented by delay fills or under-balcony speakers to cover distant seats. SSOUNDS line array systems, with their precise vertical and horizontal pattern control, allow for seamless coverage from front to back without excessive spill onto reflective surfaces.

Hybrid events—where some attendees are in the room and others join via video conference—present unique challenges. The audio system must serve both audiences without compromising either experience. For remote participants, the primary concern is capturing clean, intelligible audio from the room (microphones, speakers) while minimising background noise and echo. For in-room attendees, the system must reproduce remote participants' audio clearly without feedback or comb filtering.

A dedicated mix is essential: one for the room (often a stereo or LCR feed) and one for the streaming platform (a mono or stereo mix with different EQ and dynamics). Use a digital mixing console with Dante or AES67 networking to route audio flexibly. SSOUNDS DSP amplifiers support Dante audio networking, allowing you to send separate mixes to different zones or to the streaming encoder with low latency.

Acoustic echo cancellation (AEC) is critical for the return audio from remote participants. Most modern DSP platforms include AEC processing; ensure that the reference signal (the audio going to the room speakers) is fed back to the AEC algorithm so it can subtract it from the microphone signal. For best results, use a dedicated AEC reference from the mixer's matrix output. Also, consider using a noise gate or expander on microphones to reduce ambient noise pickup, which can be distracting for remote listeners.

The choice of microphones directly impacts intelligibility and feedback margin. For panel discussions and presentations, gooseneck or boundary (PZM) microphones are common. Condenser microphones offer better sensitivity and transient response, but require phantom power and are more susceptible to handling noise. Dynamic microphones are more rugged and have a tighter pickup pattern, making them suitable for high-SPL environments or when multiple mics are open simultaneously.

Placement is key: keep microphones as close to the talker's mouth as possible (6-12 inches for goosenecks) and avoid placing them directly in front of loudspeakers. Use the 3:1 rule—if two microphones are used, the distance between them should be at least three times the distance from each mic to the talker—to minimise phase cancellation. For hybrid events, consider using a dedicated microphone for the room audio and a separate one (e.g., a headset or lavalier) for the stream to allow independent level control.

Wireless microphones offer flexibility but require careful frequency coordination, especially in crowded RF environments. Use a spectrum analyser to find clear frequencies, and always have backup frequencies and batteries ready. SSOUNDS works with leading wireless brands to ensure seamless integration with its loudspeaker systems.

Once the system is installed, proper tuning is essential. Start by setting the gain structure: adjust amplifier input sensitivity so that the mixer's nominal output (0 dBu) produces the desired SPL without clipping. Use pink noise and an SPL meter to set levels for each zone, aiming for a flat response at the listening position. Then apply EQ to correct for room acoustics—cutting problematic resonances rather than boosting to avoid feedback.

For speech, a high-pass filter at 80-100 Hz removes low-frequency rumble and reduces muddiness. A gentle presence boost (2-4 kHz) can improve clarity without sounding harsh. Use a real-time analyser (RTA) or measurement software like SMAART to verify the system's frequency response and coherence. SSOUNDS loudspeakers come with factory presets that provide a solid starting point, but room-specific tuning is always recommended.

Finally, test the system with actual speech—have someone speak at a normal level and walk through the room. Listen for dead spots, echoes, or sibilance. Adjust delay times for distributed speakers to ensure they align with the main source (within 10-20 ms for speech). For hybrid events, also test the remote audio path: call in from a laptop and verify that the AEC is working and levels are balanced.

Modern corporate AV systems should be built on a networked backbone. Dante and AES67 allow audio to be routed over standard IP networks, simplifying cabling and enabling easy expansion. SSOUNDS amplifiers and DSP processors support these protocols, making it straightforward to add zones, integrate with video conferencing codecs, or connect to third-party control systems (e.g., Crestron, Extron).

Consider scalability from the start: choose amplifiers with enough channels and power to handle future additions, and run extra network drops to key locations. Use a centralised DSP that can be reprogrammed remotely for changing room configurations. With SSOUNDS' cloud-based monitoring and control, you can adjust levels, EQ, and routing from a tablet or laptop, ensuring the system remains flexible as needs evolve.