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HTTP Live Streaming (also known as HLS) is an HTTP-based adaptive bitrate streaming communications protocol developed by Apple Inc. and released in 2009. Support for the protocol is widespread in media players, web browsers, mobile devices, and streaming media servers.
Adaptive streaming overview Adaptive streaming in action. Adaptive bitrate streaming is a technique used in streaming multimedia over computer networks.. While in the past most video or audio streaming technologies utilized streaming protocols such as RTP with RTSP, today's adaptive streaming technologies are based almost exclusively on HTTP, [1] and are designed to work efficiently over large ...
FAAD2 – open-source decoder for Advanced Audio Coding. There is also FAAC, the same project's encoder, but it is proprietary (but still free of charge). libgsm – Lossy compression ; opencore-amr – Lossy compression (AMR and AMR-WB) liba52 – a free ATSC A/52 stream decoder (AC-3) libdca – a free DTS Coherent Acoustics decoder
Codec 2 is a low-bitrate speech audio codec (speech coding) that is patent free and open source. [1] Codec 2 compresses speech using sinusoidal coding, a method specialized for human speech. Bit rates of 3200 to 450 bit/s have been successfully created. Codec 2 was designed to be used for amateur radio and other high compression voice applications.
DASH is an adaptive bitrate streaming technology where a multimedia file is partitioned into one or more segments and delivered to a client using HTTP. [18] A media presentation description (MPD) describes segment information (timing, URL, media characteristics like video resolution and bit rates), and can be organized in different ways such as SegmentList, SegmentTemplate, SegmentBase and ...
A codec listening test is a scientific study designed to compare two or more lossy audio codecs, usually with respect to perceived fidelity or compression efficiency. Most tests take the form of a double-blind comparison. Commonly used methods are known as "ABX" or "ABC/HR" or "MUSHRA". There are various software packages available for ...
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.
This limitation was designed to simplify the negotiation of bitrate between the handset and the base station, thus vastly simplifying the implementation and testing. All other bitrates can still be used for other purposes in mobile phone networks, including multimedia messaging, streaming audio, etc.