The following are papers regarding AMBE® and IMBE™ technologies. For more information about DVSI's technology, existing systems or custom implementations using DVSI's speech compression products, please contact us today.
DVSI's Voice Compression technologies have been through extensive independent testing, winning eight of the last eight evaluations. Due to the proprietary nature of the test results, a limited number are presented for review.
by DVSI
As wireless communication becomes more prevalent so does the need to connect Land Mobile Radio (LMR) systems with commercial cellular LTE networks. Interconnecting these systems effectively is challenging because they do not share the same voice coding technology, known as a vocoder. Transcoding between the AMBE+2™ vocoder used in LMR networks and the Advanced Multi-Rate (AMR) vocoder used cellular LTE networks can lead to the loss of intelligibility, increase delay and interfere with communications security. By utilizing the AMBE+2™ vocoder in both LMR and LTE systems transcoding is not required and communication intelligibility is maintained.
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DVSI’s voice compression technologies have been through extensive independent test evaluations and emerged undefeated, winning eight of the last eight. As a result of winning these evaluations, DVSI’s speech compression technology, for which we provide numerous software and hardware solutions, was selected as the standard in most communication protocols worldwide.
Read Moreby Daniels provided courtesy of Daniels Electronics, Ltd.
This detailed Training Guide provides an Introduction to P25, Interface Standards, Practical Applications, Anatomy of the Common Air Interface and information on the AMBE+2™ vocoder, the P25 voice compression Standard.
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The need for increased utilization of available wireless communication spectrum has fueled the development of voice coding technology. From simple waveform coding techniques operating at 64 kbps, the advance of speech coding algorithms has produced communication quality systems at 2 kbps and below. In this paper, read about how channel degradations drive vocoder design, the coding gain vs. error persistence trade-off, how bit prioritization improves robustness and more.
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Independent test results show Digital Voice Systems, Inc.’s (DVSI) AMBE+™ Vocoder running at 4.8 kbps and 4.0 kbps outperforms both the ITU G.726A vocoder at 32 kbps and the ITU G.729 vocoder at 8 kbps. This voice coder builds upon the strengths of DVSI’s previous IMBE™ and AMBE™ Technology to provide toll quality speech with unsurpassed robustness to both acoustic noise and channel errors. DVSI’s AMBE+™ vocoder is targeted at applications requiring the very highest speech quality at rates as low as 3.0 kbps.
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MBE-based technology provides superior speech quality, while requiring substantially fewer MIPS and less memory than other speech coders. Read about how DVSI vocoder technologies have been designed for robustness in both background noise and channel errors, and selected for many international mobile communication standards, including standards in satellite communications, commercial aircraft telephony, and digital mobile radio. They are also widely used in many other applications such as in secure communications, voice storage, and desktop video conferencing.
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The rapid growth in personal communications has led to a critical need for increased capacity in voice communication networks. This need typically takes the form of a broad requirement to handle more simultaneous voice connections or messages in some constrained bandwidth or bit rate, without degrading power, weight, range or voice quality. The typical solution to this requirement is convert to a digital communication network and to then employ voice compression to reduce the amount of digital data (i.e. bits) which must be handled by this work.
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The incredible success of AMBE™ Voice Compression Hardware and Software is attributed to the simple fact that they utilize a fundamentally different technology than standard speech coders. This technology is the outgrowth of research that started at the Massachusetts Institute of Technology in the early 1980’s. The original goal of this work was to develop a robust speech model that would outperform the linear prediction speech model used in traditional speech coders. The outcome of this research was the introduction of the innovative Multi-Band Excitation (MBE) speech model.
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Explosive growth in the field of digital communication continues today. One of the principal technologies which has enabled this growth is voice coding, in which an analog speech signal from a microphone is digitally sampled via an A-to-D converter and then efficiently compressed into a digital bit stream for transmission or storage. A corresponding voice decoder receives this bit stream and decompresses it back into a series of digital speech samples suitable for playback through a D-to-A converter and a loudspeaker. Read more about the science behind it in the Voice Coding Overview.
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