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Multi-Band Excitation (MBE) is a series of proprietary speech coding standards developed by Digital Voice Systems, Inc. (DVSI).
In 1967 Osamu Fujimura (MIT) showed basic advantages of the multi-band representation of speech ("An Approximation to Voice Aperiodicity", IEEE 1968). This work gave a start to development of the "multi-band excitation" method of speech coding, that was patented in 1997 (now expired) by founders of DVSI as "Multi-Band Excitation" (MBE). All consequent improvements known as Improved Multi-Band Excitation (IMBE), Advanced Multiband Excitation (AMBE), AMBE+ and AMBE+2 are based on this MBE method.
AMBE is a codebook-based vocoder that operates at bitrates of between 2 and 9.6 kbit/s, and at a sampling rate of 8 kHz in 20-ms frames. The audio data is usually combined with up to 7 bit/s[ citation needed ] of forward error correction data, producing a total RF bandwidth of approximately 2,250 Hz (compared to 2,700–3,000 Hz for an analogue single sideband transmission). Lost frames can be masked by using the parameters of the previous frame to fill in the gap.
AMBE is used by the Inmarsat and Iridium satellite telephony systems and certain channels on XM Satellite Radio and is the speech coder for OpenSky Trunked radio systems.
AMBE is used in D-STAR amateur radio digital voice communications. It has met criticism from the amateur radio community because the nature of its patent [1] and licensing runs counter to the openness of amateur radio, as well as usage restriction for being "undisclosed digital code" under FCC rule 97.309(b) and similar national legislation. [2]
System Fusion, open specification from Yaesu, also uses AMBE codec with C4FM modulation.
The NXDN digital voice and data protocol uses the AMBE+2 codec. NXDN is implemented by Icom in the IDAS system and by Kenwood as NEXEDGE.
APCO Project 25 Phase 2 trunked radio systems also use the AMBE+2 codec, while older Phase 1 radios such as the Motorola XTL and XTS series use the earlier IMBE codec. Newer Phase 1 capable radios such as the APX series radios use the AMBE+2 codec, which is backwards compatible with Phase 1.
Digital Mobile Radio (DMR) and Motorola's MOTOTRBO use the AMBE+2 codec.
Use of the AMBE standard requires a license from Digital Voice Systems, Inc. While a licensing fee is due for most codecs, DVSI does not disclose software licensing terms. Anecdotal evidence [ citation needed ] suggests that licensing fee begin from between $100,000 to $1 million. For purposes of comparison, licensing fees for use of the MP3 standard started at $15,000. For small-scale use and prototyping, the only option is to purchase a dedicated hardware IC from DVSI. These ICs can be purchased for less than $100 in small quantities. [3]
DSP Innovations Inc. offers a software implementation of APCO P25 Phase 1 (Full-Rate) and Phase 2 (Half-Rate) codecs as well as DMR and dPMR codecs. A technology licence from DVSI is required. The patent for IMBE has expired.
Codec2 is an open source alternative which uses half of the bandwidth of AMBE to encode speech of similar quality, [ citation needed ] created by David Rowe and lobbied by Bruce Perens. Codec2 still continues to evolve, with additional "modes" being developed, refined and made available on a continuous basis. This has resulted in an open source codec that has progressively increased its robustness and performance – when subjected to some of the most challenging RF and acoustic environments. [4]
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.
A vocoder is a category of speech coding that analyzes and synthesizes the human voice signal for audio data compression, multiplexing, voice encryption or voice transformation.
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.
Project 25 is a suite of standards for interoperable digital two-way radio products. P25 was developed by public safety professionals in North America and has gained acceptance for public safety, security, public service, and commercial applications worldwide. P25 radios are a direct replacement for analog UHF radios, but add the ability to transfer data as well as voice, allowing for more natural implementations of encryption and text messaging. P25 radios are commonly implemented by dispatch organizations, such as police, fire, ambulance and emergency rescue service, using vehicle-mounted radios combined with repeaters and handheld walkie-talkie use.
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.
Extended Adaptive Multi-Rate – Wideband (AMR-WB+) is an audio codec that extends AMR-WB. It adds support for stereo signals and higher sampling rates. Another main improvement is the use of transform coding additionally to ACELP. This greatly improves the generic audio coding. Automatic switching between transform coding and ACELP provides both good speech and audio quality with moderate bit rates.
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.
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.
Secure voice is a term in cryptography for the encryption of voice communication over a range of communication types such as radio, telephone or IP.
The Enhanced Digital Access Communication System (EDACS) is a radio communications protocol and product family invented in the General Electric Corporation in the mid 1980s.
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.
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.
D-STAR is a digital voice and data protocol specification for amateur radio. The system was developed in the late 1990s by the Japan Amateur Radio League and uses minimum-shift keying in its packet-based standard. There are other digital modes that have been adapted for use by amateurs, but D-STAR was the first that was designed specifically for amateur radio.
NXDN stands for Next Generation Digital Narrowband, and is an open standard for public land mobile radio systems; that is, systems of two-way radios (transceivers) for bidirectional person-to-person voice communication. It was developed jointly by Icom Incorporated and Kenwood Corporation as an advanced digital system using FSK modulation that supports encrypted transmission and data as well as voice transmission. Like other land mobile systems, NXDN systems use the VHF and UHF frequency bands. It is also used as a niche mode in amateur radio.
Digital mobile radio (DMR) is a digital radio standard for voice and data transmission in non-public radio networks. It was created by the European Telecommunications Standards Institute (ETSI), and is designed to be low-cost and easy to use. DMR, along with P25 phase II and NXDN are the main competitor technologies in achieving 6.25 kHz equivalent bandwidth using the proprietary AMBE+2 vocoder. DMR and P25 II both use two-slot TDMA in a 12.5 kHz channel, while NXDN uses discrete 6.25 kHz channels using frequency division and TETRA uses a four-slot TDMA in a 25 kHz channel.
dPMR or digital private mobile radio, is a common air interface for digital mobile communications. dPMR is an open, non-proprietary standard that was developed by the European Telecommunications Standards Institute (ETSI) and published under the reference ETSI TS 102 658.
Codec 2 is a low-bitrate speech audio codec that is patent free and open source. 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.
M17 is a digital radio modulation mode developed by Wojciech Kaczmarski et al. M17 is primarily designed for voice communications on VHF amateur radio band and above. The project received a grant from the Amateur Radio Digital Communications in 2021 and 2022. The protocol has been integrated into several hardware and software projects - look here. In 2021, Kaczmarski received the ARRL Technical Innovation Award for developing an open-source digital radio communication protocol, leading to further advancements in amateur radio.
