http://cs.uef.fi/sipu/pub/SRE-review-Kinnunen-and-Li.pdf

An overview of text-independent speaker recognition: From features to supervectors

Security systems for protecting assets are described, including password-based security systems that can provide different levels of access responsive to entry of a primary or secondary password. In some versions, user-configurable security rules can provide customized responses to entry of primary or secondary passwords, including feigned or limited access, security alerts, etc. Passwords comprising overt and covert components can be used to provide enhanced security and improved user control over system response. Improved security systems involving transactions between multiple parties are also considered, with options for user-customized security rules including primary and secondary passwords, and reverse challenge and response methods. Systems for Limited Use Credentials are also disclosed to reduce the risk of identity theft.

Security Systems for Protecting an Asset

A shielded medical device implanted in a biological organism. The device has a magnetic shield, and the magnetic shield contains a layer of nanomagnetic material; such layer has a morphological density of at least 98 percent; and it is bonded to the medical device by means of an interlayer with a thickness of less than about 10 microns. The nanomagnetic material in such layer has a saturation magnetization of from about 1 to about 36,000 Gauss, a coercive force of from about 0.01 to about 5,000 Oersteds, a relative magnetic permeability of from about 1 to about 500,000, and an average particle size of less than about 100 nanometers.

Magnetically shielded assembly

During the Internet stock bubble, articles in the trade press frequently said that, in the near future, telephone traffic would be just another application running over the Internet. Such statements gloss over many engineering details that preclude voice from being just another Internet application. This paper deals with the technical aspects of implementing voice over Internet protocol (VoIP), without speculating on the timetable for convergence. First, the paper discusses the factors involved in making a high-quality VoIP call and the engineering tradeoffs that must be made between delay and the efficient use of bandwidth. After a discussion of codec selection and the delay budget, there is a discussion of various techniques to achieve network quality of service. Since call setup is very important, the paper next gives an overview of several VoIP call signaling protocols, including H.323, SIP, MGCP, and Megaco/H.248. There is a section on telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls.

/pdf/voice-over-internet-protocol-voip-35i0wqc1xn.pdf

Voice over Internet protocol (VoIP)

Efficient voice activity detection algorithms using long-term speech information

ITU-T recommendation G.729 Annex B : A silence compression scheme for use with G.729 optimized for V.70 digital simultaneous voice and data applications : Standardization and characterization of G.729

This article describes Annex B to ITU-T Recommendation G.729. Annex B defines a low-bit-rate silence compression scheme designed and optimized to work in conjunction with both the full version of G.729 and its low-complexity Annex A. To achieve good quality low-bit-rate silence compression, a robust frame-based voice activity detector module is essential to detect inactive voice frames, also called silence or background noise frames. For these detected inactive voice frames, a discontinuous transmission module measures the changes over time of the inactive voice signal characteristics and decides whether a new silence information descriptor frame should be sent to maintain the reproduction quality of the background noise at the receiving end. If such a frame is needed, the spectrum and energy parameters describing the perceptual characteristics of the background noise are efficiently coded and transmitted using 15 b/frame. At the receiving end, the comfort noise generation module regenerates the output background noise using transmitted updates or previously available parameters. The synthesized background noise is obtained by linear predictive filtering of a locally generated pseudo-white excitation signal of a controlled level. This method of coding the background noise enables the achievement of bit-rate savings for coded speech at average rates as low as 4 kb/s during normal speech conversation while maintaining reproduction quality.

ITU-T Recommendation G.729 Annex B: a silence compression scheme for use with G.729 optimized for V.70 digital simultaneous voice and data applications

A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

System of encoding and decoding speech signals

A method of encoding an input speech signal using a multi-rate encoder having a plurality of encoding rates is disclosed. A high-pass filter and then a perceptual weighting filter are applied to such signal to generate a first target signal. An adaptive codebook vector is identified from an adaptive codebook using the first target signal by filtering the vector to generate a filtered adaptive codebook vector. An adaptive codebook gain for the adaptive codebook vector is calculated and an error signal minimized. The adaptive codebook gain is adaptively reduced based on one encoding rate from the plurality of encoding rates to generate a reduced adaptive codebook gain. A second target signal based at least on the first target signal and the reduced adaptive codebook gain is generated. The input speech signal is converted into an encoded speech based on the second target signal.

Speech codec employing speech classification for noise compensation

A speech communication system and method that has an improved way of handling information lost during transmission from the encoder to the decoder. More specifically, the improved speech communication system more accurately recovers from losing information about a frame of speech such as line spectral frequencies (LSF's), pitch lag (or adaptive codebook excitation), fixed codebook excitation and/or gain information. After estimating lost parameters in a lost frame and synthesizing the speech, the improved system matches the energy of the synthesized speech to the energy of the previously received frame.

Huan-Yu Su

Papers

ITU-T recommendation G.729 Annex B : A silence compression scheme for use with G.729 optimized for V.70 digital simultaneous voice and data applications : Standardization and characterization of G.729

ITU-T Recommendation G.729 Annex B: a silence compression scheme for use with G.729 optimized for V.70 digital simultaneous voice and data applications

System of encoding and decoding speech signals

Speech codec employing speech classification for noise compensation

A speech communication system and method for handling lost frames