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Apple ProRes is a high quality, "visually lossless" lossy video compression format developed by Apple Inc. for use in post-production that supports video resolution up to 8K.It is the successor of the Apple Intermediate Codec and was introduced in 2007 with Final Cut Studio 2. [1]
The quality the codec can achieve is heavily based on the compression format the codec uses. A codec is not a format, and there may be multiple codecs that implement the same compression specification – for example, MPEG-1 codecs typically do not achieve quality/size ratio comparable to codecs that implement the more modern H.264 specification.
Videos streamed using traditional formats such as progressive download and RTSP have a common challenge; any given video must be encoded at a specific target bitrate (e.g., 500 kbps) – and that is the bitrate regardless of the access network over which it is delivered.
VP9 is an open and royalty-free [1] video coding format developed by Google.. VP9 is the successor to VP8 and competes mainly with MPEG's High Efficiency Video Coding (HEVC/H.265).
The data compression software for encoding into ALAC files, Apple Lossless Encoder, was introduced into the Mac OS X Core Audio framework on April 28, 2004, together with the QuickTime 6.5.1 update, thus making it available in iTunes since version 4.5 and above, and its replacement, the Music application. [8]
High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio defined as an MPEG-4 Audio profile in ISO/IEC 14496–3. It is an extension of Low Complexity AAC (AAC-LC) optimized for low- bitrate applications such as streaming audio .
Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors.
The Lyra codec is designed to transmit speech in real-time when bandwidth is severely restricted, such as over slow or unreliable network connections. [1] It runs at fixed bitrates of 3.2, 6, and 9 kbit/s and it is intended to provide better quality than codecs that use traditional waveform-based algorithms at similar bitrates.