Lossy audio compression in the context of MP1


Lossy audio compression in the context of MP1

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⭐ Core Definition: Lossy audio compression

In information theory, data compression, source coding, or bit-rate reduction is the process of encoding information using fewer bits than the original representation. Any particular compression is either lossy or lossless. Lossless compression reduces bits by identifying and eliminating statistical redundancy. No information is lost in lossless compression. Lossy compression reduces bits by removing unnecessary or less important information. Typically, a device that performs data compression is referred to as an encoder, and one that performs the reversal of the process (decompression) as a decoder.

The process of reducing the size of a data file is often referred to as data compression. In the context of data transmission, it is called source coding: encoding is done at the source of the data before it is stored or transmitted. Source coding should not be confused with channel coding, for error detection and correction or line coding, the means for mapping data onto a signal.

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👉 Lossy audio compression in the context of MP1

MP1 (formally MPEG-1 Audio Layer I or MPEG-2 Audio Layer I) is a lossy audio codec and one of three audio formats included in the MPEG-1 standard. For files only containing MP1 audio, the file extension .mp1 is used.

It is a deliberately simplified version of MPEG-1 Audio Layer II, created for applications where lower compression efficiency could be tolerated in return for a less complex algorithm that could be executed with simpler hardware requirements. While supported by most media players, the codec is considered largely obsolete due to wider acceptance of the more complex MPEG-1 Audio Layer II (MP2) and MPEG-1 Audio Layer III (MP3) codecs.

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Lossy audio compression in the context of Opus (audio format)

Opus is a free and open source lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed for efficient low-latency encoding of both speech and general audio. Due to its lower latency relative to other standard codecs, Opus finds specific use cases in real-time interactive communication for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications.

Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed. Bitrate, audio bandwidth, complexity, and algorithm choice can be adjusted for each individual frame. Opus has low algorithmic delay (26.5 ms by default) ideal for use as part of a real-time communication link, networked music performances, and live lip sync; by trading off quality or bitrate, the delay can be further reduced down to 5 ms. Its delay thus is significantly lower compared to competing codecs, which require well over 100 ms. Opus remains competitive with these formats in terms of quality per bitrate.

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