OPUS is an audio file that is encoded using the Ogg Opus format (lossy coding). It was created for online audio streaming, that is, in order to transmit sound through a global network. The format is being developed by the Internet Engineering Task Force (IETF) and Xiph.Org communities. Applies SILK codecs when used in Skype and CELT (from Xiph.Org), provides support for variable bitrate. Most often, the OPUS codec is used for video conferencing, game chats, VoIP telephony. OPUS is a free audio codec that has international standard status (IETF RFC 6716). Its main advantages are a low coding delay (from 2.5 to 60 ms) and its significant speed, an increased degree of compression of audio data with high-quality sound, as well as support for multi-channel audio (within 255 channels). In 2011, J. Skeglund of Google conducted two series of tests, during which OPUS coding and decoding were compared, taking into account the assessments of experts and ordinary listeners. Studies have shown that OPUS provides stereo music with the same quality as MP3 and better quality than G.719 64 kbps. OPUS offers great streaming capabilities with dynamic tweaking and very low latency. This is always high sound quality and excellent data compression. Full support for OPUS is provided by Mozilla applications. He is the key Skype audio codec.
3GA is a 3GPP audio file created in 1998. It was created by the 3rd Generation Partnership Project. It is used mainly on mobile devices in order to record, play and transmit audio data. 3GPP (3rd Generation Partnership Project) is a consortium that develops specifications for mobile telephony. The format resembles 3GP files, however, it includes only audio data. In most cases, 3GA files are used by mobile phones in order to record and transmit audio data. For example, this is how audio is recorded in Samsung Galaxy phones. The .3GA file extension can be changed to .3GP, which is widespread. Everything is supported by many programs. The 3GA format uses an adaptive coding audio codec with variable speed (AMR, AMR-NB, GSM-AMR). The development of this codec was done in order to compress encoded speech signals using adaptive modulation. In 1999, the 3GPP consortium adopted the Adaptive Variable Rate Coding (AMR) standard. Currently, it is used by GSM and UMTS communication systems. It is this format for speech recording that has found wide application on mobile devices. We note at the same time that it is not possible to play such files very often on certain devices.