Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission
Reexamination Certificate
1999-12-21
2002-09-24
McFadden, Susan (Department: 2654)
Data processing: speech signal processing, linguistics, language
Speech signal processing
For storage or transmission
C704S500000, C709S231000
Reexamination Certificate
active
06456967
ABSTRACT:
CLAIM OF PRIORITY
This application makes reference to, incorporates the same herein, and claims all benefits accruing under 35 U.S.C. Section 119 from an application for METHOD FOR VARYING FRAME STRUCTURE IN LOCAL AREA NETWORK filed in the Korean Industrial Property Office on Dec. 23, 1998 and there duly assigned Ser. No.
98-57869.
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates generally to a voice data processing system, and in particular, to a method for transmitting voice data in real time regardless of the traffic generated in a transmission line.
2. Description of the Related Art
In general, an effective and high-speed data transmission is very important in a communication network. With regard to the bandwidth for data transmission, the voice data does not require a wide bandwidth because the voice data can be compressed from 64 Kbps to 16 Kbps by digital encoding (or compressing). However, for video data transmission, a data transfer rate of about 1.5-6 Mbps is required, and the video compression technique as set forth by MPEG (Moving Picture Experts Group) is used to compress the video data while maintaining high quality. In addition, for data communication an additional bandwidth of over 10 Mbps is required.
However, for effective and high-speed data transmission, the network should be able to satisfy the communication quality required by the respective transmission media as well as the wide bandwidth requirement. The communication quality is referred to as a quality of service (QoS), which depends on the media and applications. For example, in an internet phone service, the quality of service depends on the ability to transmit voice data in real time from the transmission side to the receiving side with a little delay as possible, and the ability to retrieve the voice data at the receiving side from the transmission side with as little jitter as possible.
Accordingly, many efforts have been made to improve the QoS in the data transmission system, especially to minimize the jitter problem. With regard to the operation of the data transmission system to improve the transmission of the voice data, a controlled transmission environment is created to enhance the transmission through a protocol prior to the transmission of the compressed voice data. The controlled transmission environment is set by determining the type of packets to be transmitted and the type of the transmission method. For example, when the transmission line exhibits a good transmission quality, a transmission packet is assembled by adding a plurality of data cells to a given header prior to transmission. On the other hand, the transmission packet is assembled with a fewer number of data cells to a given header when poor transmission quality exists. The purpose of such implementation is to improve transmission efficiency by attaching a plurality of data cells to one header rather than attaching one data cell to one header to a data cell. As the number of bits required for the header in assembling one frame affects the transmission, adding more data cells to a given cell improves the transmission efficiency. Such data frame assembling method is set forth in ITU-T (International Telecommunication Union-Telecommunication standardization sector) Recommendation H.245.
However, when using the above method, the data is transmitted with a fixed frame (RTP multi-frame) determined by the initial setup process regardless of the change in the traffic condition and remains unchanged until the call is ended, thereby causing the following problems:
(1) when receiving the voice data through a LAN (Local Area Network) having an irregular bandwidth, the voice data is not received at regular intervals, thus the low delay jitter requirement cannot be satisfied;
(2) the voice data is reproduced intermittently, thus deteriorating the quality of a call; and,
(3) the writing of DSP (Digital Signal Processor) exhibits a long waiting time, causing an increase in the processing delay time and reducing the utilization efficiency of the system.
SUMMARY OF THE INVENTION
It is, therefore, an object of the present invention to provide a method for selectively controlling the number of packets constituting a voice data frame according to the traffic condition, thereby securing the real-time transmission of the voice data.
It is another object of the present invention to provide a method for controlling the number of packets constituting a voice data frame in response to the transmission delay of the voice data, thereby increasing the transmission efficiency.
To achieve the above objects, there is provided a method for assembling a voice data frame in a voice data processing system. The method comprising the steps of: upon the receipt of a voice data frame, storing a stamp time of the voice data frame and the number of packets constituting the voice data frame; removing an RTP header included in the voice data frame and storing the removed RTP header in a receiving buffer; calculating an anticipated delay time in response to the current packet number in view of a previously received frame; calculating an error time based on the difference between the actual delay time and the anticipated delay time; increasing the number of packets constituting the transmission frame when the error time is greater than a threshold value; decreasing the number of packets constituting the transmission frame when the error time is less than the threshold value; updating the DSP component data with the newly determined packet number; setting a valid flag for the transmission buffer when the updated number of packets are written; and reading the updated number of packets from the transmission buffer to assemble the next voice data frame.
The anticipated delay time is determined by multiplying the stored reference processing time by the packet number. The actual delay time is determined according to the difference between the stored time stamp and the time stamp of the previously received frame.
REFERENCES:
patent: 6157636 (2000-12-01), Voit et al.
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patent: 6292481 (2001-09-01), Voit et al.
patent: 6295292 (2001-09-01), Voit et al.
patent: 6304567 (2001-10-01), Rosenberg
patent: 6366959 (2002-02-01), Sidhu et al.
Cha Steve
Klauber & Jackson
McFadden Susan
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