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Bink uses a wavelet-based compression algorithm optimized for game video sequences. It supports resolutions up to 4K and can encode at bitrates from 500 kbps to 200 Mbps. The codec is designed for efficient decompression, leveraging multithreading and SIMD instructions on modern CPUs.
WavPack is a free and open-source lossless audio compression format and application implementing the format. It is unique in the way that it supports hybrid audio compression alongside normal compression which is similar to how FLAC works.
The 'Music' category is merely a guideline on commercialized uses of a particular format, not a technical assessment of its capabilities. For example, MP3 and AAC dominate the personal audio market in terms of market share, though many other formats are comparably well suited to fill this role from a purely technical standpoint.
In May 2004, a series of double-blind listening tests [5] (as reported on Slashdot [6]) suggested that Musepack and Ogg Vorbis (which was the 1.1 "aoTuV" fork at the time) were the two best available codecs for high-quality audio compression at bitrates around 128 kbit/s, beating MP3, AAC, WMA, and ATRAC. Listening tests of MPC:
A lossless audio coding format reduces the total data needed to represent a sound but can be de-coded to its original, uncompressed form. A lossy audio coding format additionally reduces the bit resolution of the sound on top of compression, which results in far less data at the cost of irretrievably lost information.
Lossless predictive audio compression (LPAC) is an improved lossless audio compression algorithm developed by Tilman Liebchen, Marcus Purat and Peter Noll at the Institute for Telecommunications, Technische Universität Berlin (TU Berlin), [1] to compress PCM audio in a lossless manner, in contrast to lossy compression algorithms.
Vorbis is a free and open-source software project headed by the Xiph.Org Foundation.The project produces an audio coding format and software reference encoder/decoder for lossy audio compression, libvorbis. [10]
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.