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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.
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]).
Internet Low Bit Rate Codec (iLBC, RFC 3951) – developed by Global IP Solutions/Google WebRTC; IETF Internet Draft. SILK (used by Skype) [22] CELT (developed by Xiph.Org Foundation) [23] libcelt; MPEG-4 Audio. MPEG-4 CELP; MPEG-4 HVXC; Skyphone MPLP; Inmarsat. INMARSAT-M IMBE; Inmarsat Mini-M AMBE; Meta MLow - used in Instagram, Messenger ...
MP3 LAME 3.99.5 VBR, -V 5 (~130 kbps, a well-known comparison but at higher bitrate) AAC FAAC v1.28 (Mid-low Anchor)-b 96; AAC FAAC v1.28 (Low Anchor)-q 30 (~52 kbps) Various 40 33 Opus: In results Opus is clear winner, Apple AAC is second, Ogg Vorbis and higher-bitrate LAME MP3 are statistically tied in joint third place. FAAC, known to be ...
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.
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]
G.722 is an ITU standard codec that provides 7 kHz wideband audio at data rates from 48, 56 and 64 kbit/s. This is useful for voice over IP applications, such as on a local area network where network bandwidth is readily available, and offers a significant improvement in speech quality over older narrowband codecs such as G.711, without an excessive increase in implementation complexity.
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.