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Free and open-source software portal; libavcodec is a free and open-source [4] library of codecs for encoding and decoding video and audio data. [5]libavcodec is an integral part of many open-source multimedia applications and frameworks.
G.729 is a royalty-free [1] narrow-band vocoder-based audio data compression algorithm using a frame length of 10 milliseconds. It is officially described as Coding of speech at 8 kbit/s using code-excited linear prediction speech coding (CS-ACELP), and was introduced in 1996. [2]
As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders do feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two early MP3 encoders set at about 128 kbit/s, [ 75 ] one scored 3.66 on a 1–5 scale, while the other scored only 2.22.
A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software. Some audio coding ...
Turing – A High Efficiency Video Coding (HEVC/H.265) encoder implemented by BBC Research. libaom – Reference implementation for the royalty free AV1 video coding format by AOMedia, inheriting technologies from VP9, Daala and Thor. Kvazaar – An academic open-source encoder based on the High Efficiency Video Coding (HEVC/H.265) standard.
Subband coding resides at the heart of the popular MP3 format (more properly known as MPEG-1 Audio Layer III), for example. Sub-band coding is used in the G.722 codec which uses sub-band adaptive differential pulse code modulation (SB-ADPCM) within a bit rate of 64 kbit/s. In the SB-ADPCM technique, the frequency band is split into two sub ...
G.711 is a narrowband audio codec originally designed for use in telephony that provides toll-quality audio at 64 kbit/s. It is an ITU-T standard (Recommendation) for audio encoding, titled Pulse code modulation (PCM) of voice frequencies released for use in 1972.
Code-excited linear prediction (CELP) is a linear predictive speech coding algorithm originally proposed by Manfred R. Schroeder and Bishnu S. Atal in 1985. At the time, it provided significantly better quality than existing low bit-rate algorithms, such as residual-excited linear prediction (RELP) and linear predictive coding (LPC) vocoders (e.g., FS-1015).