摘要:
In an automated exchange system, a separate virtual instrument is used in the matching process of the system. The virtual instrument guarantees that both a derivative instrument and its underlying instrument are traded together. The underlying instrument, i.e. the instrument in which derivative instruments are traded, is then preferably displayed together with the virtual instruments. The underlying instrument is presented with a price. The matching of the virtual instrument can take place in a matching module of the automated exchange system. After a trade in a virtual instrument is matched in the matching process of the system, the match is reported to a subsequent deal capture module where the corresponding different trades or deals of the virtual instrument are formed. The trades or deals formed in the deal capture module do not need to be matched because the number of instruments and the price can be deduced from the information relating to the virtual instrument.
摘要:
In an automated trading system for matching bids and offers entered into the system by a number of traders connected to the system, a server hosts a matching processor and an associated memory forming an orderbook of the system where both fixed-income instruments paying a coupon, referred to as bonds, and fixed-income instruments not paying a coupon (zero-coupon), referred to as stripped bonds, are traded. The system derives prices for bonds using information from stripped bonds.
摘要:
In an automated trading system for matching bids and offers entered into the system by a number of traders connected to the system, the system preferably comprises a server hosting a matching processor and an associated memory forming an orderbook of the system and wherein both fixed-income instruments paying a coupon and fixed-income instruments not paying a coupon (zero-coupon) are traded. The system is additionally designed to derive prices for bonds using information from stripped bonds.
摘要:
The present invention relates to a method and a user equipment arranged to communicate with at least a second user equipment in a VoIP service data transmission in a wireless communication system using a VoIP service, provided by an application server. The method comprises the steps of: receiving transmissions in form of a media stream from the at least one second user equipment; storing the data of the media stream; detecting whether an interruption of said transmissions having at least a minimum length has occurred during the VoIP service data transmission or expecting that an interruption of said transmissions having at least a minimum length will occur during the VoIP service data transmission, and if such an interruption is detected or expected, using a non-normative playout rate of the data at playout, thereby obtaining a more efficient interactivity in the user communication.
摘要:
The present invention relates to a method and a user equipment arranged to communicate with at least a second user equipment in a VoIP service data transmission in a wireless communication system using a VoIP service, provided by an application server. The method comprises the steps of: receiving transmissions in form of a media stream from the at least one second user equipment; storing the data of the media stream; detecting whether an interruption of said transmissions having at least a minimum length has occurred during the VoIP service data transmission or expecting that an interruption of said transmissions having at least a minimum length will occur during the VoIP service data transmission, and if such an interruption is detected or expected, using a non-normative playout rate of the data at playout, thereby obtaining a more efficient interactivity in the user communication.
摘要:
In a trading system providing an anonymous market, orders in the market are executed using cryptographic keys. Traders can view the orders and use their specific key to determine the particular rating of an order in accordance with the trader's own preferences. Since the same order information is sent to all traders, the bandwidth and processing requirements are kept at a minimum.
摘要:
A receiving terminal estimates a required jitter buffer depth for each received audio frame, by locating (61) the fastest previously received audio frame, calculating (62) an estimated required play-out delay from stored data associated with said fastest audio frame, and transforming (63) the estimated play-out delay into a required jitter buffer depth for accommodating the calculated play-out delay of the received audio frame. Further, this required jitter buffer depth is made available for jitter buffer management, e.g. to achieve a certain loss rate. Data associated with each received audio frame is stored to be used for estimating the required jitter buffer depth for consecutive audio frames.
摘要:
In automated exchange system, a single matching unit is supplemented with a calculation unit and a global memory accessible by both the calculation unit and the matching unit. Such a computer architecture will make it possible to perform some of the calculations related to the volume and/or prices of the baits needed in the matching to be performed in advance. The matching process is able to use the values resulting from the pre-calculation when needed, and since no or few calculations are done in one of the most critical parts of the system, i.e. the matching unit, the process of matching combination contracts can be performed at a much higher rate. Hereby the performance of the matching process will be significantly increased. The provision of one or several calculation units will make it possible to perform even very complex calculations can be performed since most calculations need not be performed in real time.
摘要:
A speech quality estimation technique that employs an arbitrary, speech quality estimation algorithm. The speech quality estimation technique analyzes a reference speech signal and a test speech signal, and based on this analysis, identifies the level of continuous delay variation, if any, and the location of and size of any intermittent delay variations along the test signal. The reference speech signal and/or the test speech signal are adjusted to account for continuous delay variation and intermittent delay variations, such that the reference speech signal and the test signal are similarly scaled with respect to the time domain. The reference speech signal and the test speech signal are then compared for the purpose of generating a speech quality estimation. The resulting speech quality estimation is then adjusted based on the level of continuous delay variation and any intermittent delay variations.