Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
2000-08-31
2004-08-03
Dorvil, Richemond (Department: 2654)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S230000
Reexamination Certificate
active
06772112
ABSTRACT:
TECHNICAL FIELD
The invention relates to relates generally to wireless communication networks and, more particularly, to a method for efficiently providing voice communications over wireless and/or cellular networks.
DESCRIPTION OF THE PRIOR ART
The widespread growing popularity of the Internet has encouraged wireless communication system developers to continually improve the data communication capabilities of their systems. In response to this need, various standards bodies have formulated and continue to formulate new third generation (3G) standards which support higher data rates. For example, standards organizations such as the European Telecommunications Standards Institute (ETSI), the Association of Radio Industries and Broadcasting (ARIB) and the Telecommunications Industry Association (TIA) are continually developing standards to support faster and more efficient wireless communications.
Similarly, the wireless communications industry is often developing and implementing new wireless transmission protocols which provide faster, more robust and more efficient data communications over air interfaces. For example, GSM continues to evolve. In another example, general packet radio service (GPRS) has been developed as a packet-switched upgrade for the well known time division multiple access (TDMA) system. In a further advancement in the art, enhanced GPRS (EGPRS) has also been developed.
Presently, GSM, GPRS and EGPRS physical layers have the following characteristics: a carrier that consists of two 200 kHz bandwidth segments of the allocated GSM spectrum, 45 MHz apart, one for the downlink and one for the uplink; time is divided into frames with a multiframe comprising 52 frames and spans 240 msec.; each frame consists of 8 time slots; one slot on one carrier is referred to as a GSM channel; there is a one-to-one correspondence between a slot (numbered j, j=0, . . . 7) on a downlink carrier at frequency (f) and an uplink slot (numbered j) on the corresponding uplink carrier (f+45 MHz); a transmission in a slot is referred to as a burst; and a block consists of a predefined set of four bursts on the same slot.
Radio access bearers are currently being designed in order to provide real time services in a next phase of EGPRS. However, recent approaches rely on using the existing burst based random access channels on the uplink and block based assignment channels on the downlink. Each block is interleaved and transmitted over 4 bursts (20 msec). However, investigation has shown systems based on 20 msec granularity require at least a 60 msec delay budget. Also, the investigation has shown transmission of assignments to multiple mobile stations within a single 20 msec message often is inefficient due to low packing and is incompatible with interference reduction techniques such as smart antennas and power control. As a result, block based assignment channels according to the recent approaches can result in excessive control overhead and excessive delays for statistical multiplexing of real time transfers (e.g. voice talkspurts). It is desirable to provide a better access and assignment system and method.
In order to efficiently use the high capacity of a wireless or a cellular data telecommunication system (e.g., GPRS or EGPRS), it is also desirable to provide voice and data multiplexing capability as well as statistical multiplexing of voice users. Currently these cellular data telecommunication systems are designed to provide primarily non-real time (delay insensitive) data services. Conversational speech and other real time interactive communications are delay sensitive and require the design of new control mechanisms to provide fast control channels to meet the critical delay requirements. Therefore, there is a need to redesign wireless data telecommunication systems to provide such control capabilities to make them suitable for multiplexing both non-real-time services and real-time services, such as conversational speech.
For cellular systems based on IS-136 and GSM standards existing before December 1999, chain interleaving was used in order to increase the interleaving depth without significantly increasing the delay. However, when chain interleaving according to those standards was used, half of the available bits over a 20 msec. transmission interval (e.g. first 4 GSM bursts) at the start and end of a voice talkspurt do not carry coded speech and therefore were wasted. Also, the first 20 msec. speech frame became available at the receiver only after 40 msec (if 8 burst interleaving was used as in the GSM full rate speech channels). The use of statistical multiplexing would improve traffic, but statistical multiplexing requires fast assignment of traffic channels at the beginning of each talkspurt. An additional problem is that the control channel procedures for access and assignment take time and introduce delays in the play out of speech at the receiver. These delays are in addition to the signal processing and transport delays that are inherent in wireless and cellular communications. It is desirable to keep one way and round-trip speech delays very short in order to provide high quality natural conversations.
SUMMARY OF THE INVENTION
Briefly stated in accordance with one aspect of the invention, the aforementioned problems are addressed and an advance in the art achieved by providing a wireless speech system using chain interleaving of speech frames where one or more speech frames are each replaced with a corresponding half speech frame that is encoded and interleaved over half the distance of the original speech frame.
In accordance with one aspect of the invention, the aforementioned problems are addressed and an advance in the art achieved by providing a wireless speech system using chain interleaving of speech frames where one or more speech frames are each replaced with a corresponding half speech frame that is encoded and interleaved over half the distance of the original speech frame. Each half speech frame is encoded and interleaved over the last half of the interleaving period of the full speech frame that the half speech frame replaced.
In accordance with one aspect of the invention, the aforementioned problems are addressed and an advance in the art achieved by providing a wireless speech system using chain interleaving of speech frames where one or more speech frames are each replaced with a corresponding half speech frame that is encoded and interleaved over half the distance of the original speech frame. Each half speech frame is encoded and interleaved over the last half of the interleaving period of the full speech frame that the half speech frame replaced. Further, the half speech frames correspond to the beginning of a talk spurt, and where nothing is transmitted during the first half of the interleaving period of the corresponding full speech frames.
In accordance with another aspect of the invention, the aforementioned problems are addressed and an advance in the art achieved by providing a system that uses a half speech block to reduce speech play-out delay to improve the quality of conversational speech. The half speech block results in half the number of coded bits as a full speech block after coding. This half speech block is encoded by a lower speech rate mode of an adaptive multi-rate vocoder.
REFERENCES:
patent: 5517492 (1996-05-01), Spear
patent: 5553190 (1996-09-01), Ohya et al.
patent: 5619496 (1997-04-01), Weir
patent: 5745492 (1998-04-01), Nakamura et al.
patent: 5778338 (1998-07-01), Jacobs et al.
patent: 5812968 (1998-09-01), Hassan et al.
patent: 6604070 (2003-08-01), Gao et al.
patent: WO 9619880 (1996-06-01), None
Dorvil Richemond
Lucent Technologies - Inc.
Storm Donald L.
LandOfFree
System and method to reduce speech delay and improve voice... does not yet have a rating. At this time, there are no reviews or comments for this patent.
If you have personal experience with System and method to reduce speech delay and improve voice..., we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and System and method to reduce speech delay and improve voice... will most certainly appreciate the feedback.
Profile ID: LFUS-PAI-O-3320255