Broadcast Audio and Streaming Standards Flashcards

What is 'Dolby Atmos' and how does it differ from traditional channel-based surround audio in broadcast and streaming?

Answer: Dolby Atmos is an object-based audio format where sound objects carry position metadata, allowing renderers to place sounds in 3D space dynamically on any speaker configuration, unlike fixed-channel formats

Dolby Atmos is an object-based spatial audio format where individual sound objects (with 3D position metadata) are rendered dynamically to the available speaker layout — from headphones (binaural) to home theater to Atmos cinema arrays. Unlike channel-based surround (5.1, 7.1), Atmos scales to any configuration without remixing.

What is the purpose of 'loudness metering' during a live broadcast production?

Answer: To monitor signal levels and ensure program loudness stays within broadcast specification (e.g., -24 LKFS for ATSC) in real time to avoid regulatory non-compliance

Loudness metering during live broadcast allows engineers to monitor integrated, short-term, and momentary loudness alongside true peak levels in real time. This ensures the program consistently meets broadcast loudness standards (ATSC: -24 LKFS, EBU: -23 LUFS) and prevents regulatory non-compliance from the CALM Act or EBU R 128.

What does 'AC-4' (Dolby AC-4) add over the existing AC-3/EAC-3 codec for next-generation broadcast?

Answer: AC-4 supports immersive audio (object-based and channel-based) including Dolby Atmos, improved coding efficiency at lower bitrates, and personalization features (dialogue enhancement, loudness personalization)

Dolby AC-4 is the next-generation broadcast codec designed for ATSC 3.0 and streaming. It delivers improved compression efficiency (higher quality at lower bitrates than EAC-3), supports Dolby Atmos immersive audio (both channel and object based), and enables personalization features such as dialogue clarity enhancement and loudness control per listener.

In OTT (over-the-top) streaming audio delivery, what does 'audio fingerprinting' technology accomplish?

Answer: It creates a compact acoustic signature of an audio track that allows content identification, rights management, and automatic metadata matching without requiring embedded metadata

Audio fingerprinting generates a compact perceptual hash (fingerprint) of audio content that uniquely identifies it regardless of compression or minor processing. Used by platforms like YouTube (Content ID) and Shazam, it enables automatic content identification, rights management, royalty tracking, and metadata association.

What is the WebRTC audio processing pipeline designed to handle in browser-based streaming applications?

Answer: Automatic echo cancellation, noise suppression, and adaptive bitrate Opus encoding for real-time bidirectional audio communication over browser-based peer-to-peer connections

WebRTC includes a built-in audio processing pipeline with acoustic echo cancellation (AEC), noise suppression, automatic gain control (AGC), and Opus codec encoding/decoding. It enables real-time peer-to-peer audio (and video) communication directly in browsers and native apps without plugins, used in Google Meet, Zoom web, and similar platforms.

What is 'SMPTE timecode' used for in broadcast audio production, and what format does it use?

Answer: A standardized time address signal (hours:minutes:seconds:frames) embedded in or alongside video and audio to synchronize equipment and provide a common reference for editing and delivery

SMPTE timecode is a standardized address format (HH:MM:SS:FF — hours, minutes, seconds, frames) used to synchronize audio recorders, video editors, mixing consoles, and delivery systems in broadcast production. It allows frame-accurate synchronization across all devices in a production chain and is essential for multi-machine editing and post-production sync.