Vector sum excited linear prediction

Last updated

Vector sum excited linear prediction (VSELP) is a speech coding method used in several cellular standards. The VSELP algorithm is an analysis-by-synthesis coding technique and belongs to the class of speech coding algorithms known as CELP (Code Excited Linear Prediction). [1]

Variations of this codec have been used in several 2G cellular telephony standards, including IS-54, IS-136 (D-AMPS), GSM (Half Rate speech) and iDEN. It was also used in the first version of RealAudio for audio over the Internet. The IS-54 VSELP standard was published by the Telecommunications Industry Association in 1989.

D-AMPS (IS-54 and IS-136) VSELP specifies an encoding of each 20 ms of speech into 159-bit frames, thus achieving a raw data rate of 7.95 kbit/s. In an actual TDMA cell phone, the vocoder output is packaged with error correction and signaling information, resulting in an over-the-air data rate of 16.2 kbit/s. For internet audio, each 159-bit frame is stored in 20 bytes, leaving 1 bit unused. The resulting file thus has a data rate of exactly 8 kbit/s.

GSM half-rate VSELP (GSM 06.20) uses 20 ms frames with 112 bits per frame, giving a raw data rate of 5.6 kbit/s. The iDEN VSELP coder has three modes:

A major drawback of VSELP is its limited ability to encode non-speech sounds, so that it performs poorly when encoding speech in the presence of background noise. For this reason, use of VSELP has been gradually phased out in favor of newer codecs.

VSELP is also the proprietary digital vocoder used in early Motorola ASTRO products. It was phased out in favor of the newer IMBE method used for P25.

Related Research Articles

Speech coding is an application of data compression to digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream.

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.

<span class="mw-page-title-main">G.723.1</span> ITU-T Recommendation

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.

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.

Enhanced Full Rate or EFR or GSM-EFR or GSM 06.60 is a speech coding standard that was developed in order to improve the quality of GSM.

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.

Half Rate is a speech coding system for GSM, developed in the early 1990s.

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.

<span class="mw-page-title-main">G.729</span> ITU-T Recommendation

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.728 is an ITU-T standard for speech coding operating at 16 kbit/s. It is officially described as Coding of speech at 16 kbit/s using low-delay code excited linear prediction.

Mixed-excitation linear prediction (MELP) is a United States Department of Defense speech coding standard used mainly in military applications and satellite communications, secure voice, and secure radio devices. Its standardization and later development was led and supported by the NSA and NATO. The current "enhanced" version is known as MELPe.

Qualcomm code-excited linear prediction (QCELP), also known as Qualcomm PureVoice, is a speech codec developed in 1994 by Qualcomm to increase the speech quality of the IS-96A codec earlier used in CDMA networks. It was later replaced with EVRC since it provides better speech quality with fewer bits. The two versions, QCELP8 and QCELP13, operate at 8 and 13 kilobits per second (Kbit/s) respectively.

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.

Harmonic Vector Excitation Coding, abbreviated as HVXC is a speech coding algorithm specified in MPEG-4 Part 3 standard for very low bit rate speech coding. HVXC supports bit rates of 2 and 4 kbit/s in the fixed and variable bit rate mode and sampling frequency of 8 kHz. It also operates at lower bitrates, such as 1.2 - 1.7 kbit/s, using a variable bit rate technique. The total algorithmic delay for the encoder and decoder is 36 ms.

FS-1016 is a deprecated secure telephony speech encoding standard for Code-excited linear prediction (CELP) developed by the United States Department of Defense and finalized February 14, 1991.

Code-excited linear prediction (CELP) is a linear predictive speech coding algorithm originally proposed by Manfred R. Schroeder and Bishnu S. Atal in 1985. At the time, it provided significantly better quality than existing low bit-rate algorithms, such as residual-excited linear prediction (RELP) and linear predictive coding (LPC) vocoders. Along with its variants, such as algebraic CELP, relaxed CELP, low-delay CELP and vector sum excited linear prediction, it is currently the most widely used speech coding algorithm. It is also used in MPEG-4 Audio speech coding. CELP is commonly used as a generic term for a class of algorithms and not for a particular codec.

Internet Low Bitrate Codec (iLBC) is a royalty-free narrowband speech audio coding format and an open-source reference implementation (codec), developed by Global IP Solutions (GIPS) formerly Global IP Sound. It was formerly freeware with limitations on commercial use, but since 2011 it is available under a free software/open source license as a part of the open source WebRTC project. It is suitable for VoIP applications, streaming audio, archival and messaging. The algorithm is a version of block-independent linear predictive coding, with the choice of data frame lengths of 20 and 30 milliseconds. The encoded blocks have to be encapsulated in a suitable protocol for transport, usually the Real-time Transport Protocol (RTP).

Enhanced Variable Rate Codec B (EVRC-B) is a speech codec used by CDMA networks. EVRC-B is an enhancement to EVRC and compresses each 20 milliseconds of 8000 Hz, 16-bit sampled speech input into output frames of one of the four different sizes: Rate 1 - 171 bits, Rate 1/2 - 80 bits, Rate 1/4 - 40 bits, Rate 1/8 - 16 bits.

<span class="mw-page-title-main">G.718</span> ITU-T Recommendation

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.

References

This page is based on this Wikipedia article
Text is available under the CC BY-SA 4.0 license; additional terms may apply.
Images, videos and audio are available under their respective licenses.