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Speex is an audio compression codec specifically tuned for the reproduction of human speech and also a free software speech codec that may be used on voice over IP applications and podcasts. It is based on the code excited linear prediction speech coding algorithm. Its creators claim Speex to be free of any patent restrictions and it is licensed under the revised (3-clause) BSD license. It may be used with the Ogg container format or directly transmitted over UDP/RTP. It may also be used with the FLV container format.
G.723.1 is an audio codec for voice that compresses voice audio in 30 ms frames. An algorithmic look-ahead of 7.5 ms duration means that total algorithmic delay is 37.5 ms. Its official name is Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s. It is sometimes associated with a Truespeech trademark in coprocessors produced by DSP Group.
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.
The Adaptive Multi-Rateaudio 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 speech starting at 7.4 kbit/s.
Full Rate was the first digital speech coding standard used in the GSM digital mobile phone system. It uses linear predictive coding (LPC). The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample. The quality of the coded speech is quite poor by modern standards, but at the time of development it was a good compromise between computational complexity and quality, requiring only on the order of a million additions and multiplications per second. The codec is still widely used in networks around the world. Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate.
Adaptive Multi-Rate Wideband (AMR-WB) is a patented wideband speech audio coding standard developed based on Adaptive Multi-Rate encoding, using a similar methodology to algebraic code-excited linear prediction (ACELP). AMR-WB provides improved speech quality due to a wider speech bandwidth of 50–7000 Hz compared to narrowband speech coders which in general are optimized for POTS wireline quality of 300–3400 Hz. AMR-WB was developed by Nokia and VoiceAge and it was first specified by 3GPP.
G.729 is a royalty-free 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. The wide-band extension of G.729 is called G.729.1, which equals G.729 Annex J.
G.722 is an ITU-T standard 7 kHz wideband audio codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM (SB-ADPCM). The corresponding narrow-band codec based on the same technology is G.726.
G.722.1 is a licensed royalty-free ITU-T standard audio codec providing high quality, moderate bit rate wideband (50 Hz – 7 kHz audio bandwidth, 16 ksps audio coding. It is a partial implementation of Siren 7 audio coding format developed by PictureTel Corp.. Its official name is Low-complexity coding at 24 and 32 kbit/s for hands-free operation in systems with low frame loss. It uses a modified discrete cosine transform audio data compression algorithm.
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. G.726 also introduced a new 16 kbit/s rate. The four bit rates associated with G.726 are often referred to by the bit size of a sample, which are 2, 3, 4, and 5-bits respectively. The corresponding wide-band codec based on the same technology is G.722.
Selectable Mode Vocoder (SMV) is variable bitrate speech coding standard used in CDMA2000 networks. SMV provides multiple modes of operation that are selected based on input speech characteristics.
Variable-Rate Multimode Wideband (VMR-WB) is a source-controlled variable-rate multimode codec designed for robust encoding/decoding of wideband/narrowband speech. The operation of VMR-WB is controlled by speech signal characteristics and by traffic condition of the network. Depending on the traffic conditions and the desired quality of service (QoS), one of the 4 operational modes is used. All operating modes of the existing VMR-WB standard are fully compliant with cdma2000 rate-set II. VMR-WB modes 0, 1, and 2 are cdma2000 native modes with mode 0 providing the highest quality and mode 2 the lowest ADR. VMR-WB mode 3 is the AMR-WB interoperable mode operating at an ADR slightly higher than mode 0 and providing a quality equal or better than that of AMR-WB at 12.65 kbit/s when in an interoperable interconnection with AMR-WB at 12.65 kbit/s.
Packet loss concealment (PLC) is a technique to mask the effects of packet loss in voice over IP (VoIP) communications. When the voice signal is sent as VoIP packets on an IP network, the packets may travel different routes. A packet therefore might arrive very late, might be corrupted, or simply might not arrive at all. One example case of the last situation could be, when a packet is rejected by a server which has a full buffer and cannot accept any more data. Other cases include network congestion resulting in significant delay. In a VoIP connection, error-control techniques such as automatic repeat request (ARQ) are not feasible and the receiver should be able to cope with packet loss. Packet loss concealment is the inclusion in a design of methodologies for accounting for and compensating for the loss of voice packets.
internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS). It is suitable for VoIP applications and streaming audio. The encoded blocks have to be encapsulated in a suitable protocol for transport, e.g. RTP.
G.719 is an ITU-T standard audio coding format providing high quality, moderate bit rate wideband audio coding at low computational load. It was produced through a collaboration between Polycom and Ericsson.
G.718 is an ITU-T Recommendation embedded scalable speech and audio codec providing high quality narrowband speech over the lower bit rates and high quality wideband speech over the complete range of bit rates. In addition, G.718 is designed to be highly robust to frame erasures, thereby enhancing the speech quality when used in Internet Protocol (IP) transport applications on fixed, wireless and mobile networks. Despite its embedded nature, the codec also performs well with both narrowband and wideband generic audio signals. The codec has an embedded scalable structure, enabling maximum flexibility in the transport of voice packets through IP networks of today and in future media-aware networks. In addition, the embedded structure of G.718 will easily allow the codec to be extended to provide a superwideband and stereo capability through additional layers which are currently under development in ITU-T Study Group 16. The bitstream may be truncated at the decoder side or by any component of the communication system to instantaneously adjust the bit rate to the desired value without the need for out-of-band signalling. The encoder produces an embedded bitstream structured in five layers corresponding to the five available bit rates: 8, 12, 16, 24 & 32 kbit/s.
Constrained Energy Lapped Transform (CELT) is an open, royalty-free lossy audio compression format and a free software codec with especially low algorithmic delay for use in low-latency audio communication. The algorithms are openly documented and may be used free of software patent restrictions. Development of the format was maintained by the Xiph.Org Foundation and later coordinated by the Opus working group of the Internet Engineering Task Force (IETF).
SILK is an audio compression format and audio codec developed by Skype Limited, now a Microsoft subsidiary. It was developed for use in Skype, as a replacement for the SVOPC codec. Since licensing out, it has also been used by others. It has been extended to the Internet standard Opus codec.
Global IP Solutions was a United States-based corporation that developed real-time voice and video processing software for IP networks, before it was acquired by Google in May 2010. The company delivered embedded software that enabled real-time communications capabilities for video and voice over IP (VoIP). GIPS was perhaps best known for developing the narrowband iLBC and wideband iSAC speech codecs.
Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.
References
- 1 2 3 4 Duric, Alan; Andersen, Soren Vang (2004). "Real-time Transport Protocol (RTP) Payload Format for internet Low Bit Rate Codec (iLBC) Speech" . Retrieved 2011-06-23.
- ↑ Dana Blankenhorn (2010-05-18). "Why Google bought Global IP Solutions". ZDNet . Archived from the original on May 21, 2010. Retrieved 2011-06-23.
- ↑ "Global IP Solutions iLBC Freeware Public License" (PDF). 2008-10-30. Archived from the original (PDF) on October 30, 2008. Retrieved 2011-06-23.
- ↑ "iLBCfreeware". Archived from the original on December 19, 2008.
- 1 2 "iLBC Freeware". Archived from the original on 2011-07-05. Retrieved 2011-06-23.
- ↑ "Tuenti+WebRTC (Voip2day 2014)".
- ↑ Ooma * codes, 12 May 2015
- ↑ Roar Hagen, Speech Coders - a VoIP perspective , retrieved 2011-06-23
- ↑ Duric, Alan; Andersen, Steven C. (2002). "Internet Low Bit Rate Codec - draft-andersen-ilbc-00.txt" . Retrieved 2011-06-23.
- ↑ "former GIPS license" (PDF). Archived from the original (PDF) on October 30, 2008.