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G.729 (06/12) Free [19] Free Various proprietary VoIP software FFmpeg (decoding only) voice recording: No Yes No Expired [19] No G.729.1: ITU-T 2006-05 G.729.1 Am.8 (03/13) Free [19] Free Various proprietary VoIP software voice recording, DECT telephony No Yes No No No GSM-FR: ETSI Special Mobile Group 1990-1994 (ETS 300 580-2) ETSI EN 300 961 ...
As such, the user normally doesn't have a raw AAC file, but instead has a .m4a audio file, which is a MPEG-4 Part 14 container containing AAC-encoded audio. The container also contains metadata such as title and other tags, and perhaps an index for fast seeking. [2] A notable exception is MP3 files, which are raw audio coding without a ...
In popular usage, MP3 often refers to files of sound or music recordings stored in the MP3 file format (.mp3) on consumer electronic devices. Originally defined in 1991 as the third audio format of the MPEG-1 standard, it was retained and further extended—defining additional bit rates and support for more audio channels —as the third audio ...
The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding.AMR is a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality [3] speech starting at 7.4 kbit/s.
G.722.1, another 7 kHz codec, operates at half the data rate while delivering comparable or better quality than G.722, but is a transform-based codec. G.722.1 Annex C is very similar to G.722.1, but provides twice the audio bandwidth, 14 kHz. And G.722.2, which operates on wideband speech and delivers very low bitrates, is an ACELP-based algorithm.
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.
G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723 , which described ADPCM for 24 and 40 kbit/s.
The speech encoder accepts 13 bit linear PCM at an 8 kHz sample rate. This can be direct from an analog-to-digital converter in a phone or computer, or converted from G.711 8-bit nonlinear A-law or μ-law PCM from the PSTN with a lookup table. In GSM, the encoded speech is passed to the channel encoder specified in GSM 05.03.