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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.
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 .
Audio Unreal Media Server: Unreal Streaming Technologies 2003-09-15 15.0 (2023-01-03) 2023-01-03 $995 perpetual license proprietary: Audio/Video VLC media player: VideoLAN? 2.2.4 (2016-06-05) 2016-06-05 Free GPL v2: Audio/Video Windows Media Services: Microsoft: 1996-12-10 9.6 (2010-04-05) 2010-04-05 Free proprietary: Audio/Video Wowza ...
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 ...
streams music to DAAP clients like iTunes and Rhythmbox: GPL-2.0-or-later: Icecast: Yes Yes Yes a broadcast server, serves audio signals to clients over the HTTP protocol GPL-2.0-only: VLC media player: Yes Yes Yes Yes media and server programs for video and audio streaming VLC: GPL-2.0-or-later libVLC: LGPL-2.1-or-later
SBC, or low-complexity subband codec, is an audio subband codec specified by the Bluetooth Special Interest Group (SIG) for the Advanced Audio Distribution Profile (A2DP). [1] SBC is a digital audio encoder and decoder used to transfer data to Bluetooth audio output devices like headphones or loudspeakers. It can also be used on the Internet. [2]
Internet Low Bitrate Codec (iLBC) is a royalty-free narrowband speech audio coding format and an open-source reference implementation , developed by Global IP Solutions (GIPS) formerly Global IP Sound (acquired by Google Inc in 2011 [2]).
Possible bitrate and latency combinations compared with other audio formats. Opus supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s (or up to 256 kbit/s per channel for multi-channel tracks), frame sizes from 2.5 ms to 60 ms, and five sampling rates from 8 kHz (with 4 kHz bandwidth) to 48 kHz (with 20 kHz bandwidth, the human hearing range).