Code converter and a system using same

Details Code converter and a system using same Code converter and a system using same

2000-02-18

2004-03-02

Tran, Cong Van (Department: 2683)

Telecommunications

Receiver or analog modulated signal frequency converter

Frequency conversion between signal source and receiver

C455S022000, C455S236100, C375S219000, C375S220000, C370S503000

Reexamination Certificate

active

06701139

ABSTRACT:

FIELD OF THE INVENTION
The present invention relates to telecommunication systems in general, and in particular to the transmission of compressed signals in telecommunication systems.
BACKGROUND OF THE INVENTION
Telecommunication has moved in the recent years towards digital networks carrying voice, facsimile and other signals. One known way in the art to increase the efficiency of such networks is to transmit the signals in a compressed form, thus using the available bandwidth for simultaneous transmission of more information.
U.S. Pat. No. RE 35,740 discloses a system for carrying voice, facsimile and non-facsimile voice band data. This system includes various compressing mechanisms and offers an improved detection of facsimile signals. Still, when speech, facsimile and other signals are subjected to several compression/decompression cycles, their quality degrades substantially, and at times, the signals may be received at a quality and delay that are unacceptable.
Some methods were suggested in the past to overcome this handicap. Cox et al., in International Conference on Communications, Vol. 1, June 1998, pp. 90-95, suggest a method to decrease the distortion and delay introduced while using low rate speech coding. It was suggested there that the receiving module, the compressor, searches the less significant bits of the input 64 Kb/s mu-law coded stream for the synchronization pattern. If it detects the pattern, it temporarily squelches the compressed signal and enables the de-compressor to begin padding rather than decoding. When the de-compressor is padding the low-rate frame, a new synchronization pattern is inserted into one of the non-used bits. When the far end compressor detects this new synchronization pattern it disables the squelch on its compressor and begins stripping the padded bits. However, when the compressor at the receiving end does not find the “robbed” less significant bits synchronization pattern in the 64 Kb/s stream, it will not proceed to the pad and strip state, and the de-compressor will insert the robbed bit synchronization pattern into the 64 Kb/s stream, transmitted to the subscriber. One way of increasing the rate of the low rate bit stream suggested in this publication is, by inserting the low rate signal into the least significant bits of the mu-law word of the 64 Kb/s stream. The advantage of using such a method is that if by any change the system treats the signal as mu-low coded speech, only low level noise signal would be noticed by the human listener.
Our co-pending application, U.S. Ser. No. 09/465,456 filed Dec. 17, 1999, describes a digital telecommunication station adapted to receive different types of signals, detecting and identifying their type and allows handling the incoming signals in an end-to-end mode of operation. By this mode, the signals can be transmitted through a number of compressing/decompressing devices, thus retaining them in their compressed form and only decompress them into their digital de-compressed output signals only at the most downstream compressing/decompressing device.
Still, there are other cases where signals are transmitted via a number of different types of compressing/decompressing devices located along the transmission path. The common way used in the art to handle these cases, is, if the next leg along the transmission's path does not have the capability of decompressing this type of compressed transmission, then the signal is decompressed and transmitted via this next leg in its non-compressed form. Once the transmission is received at the receiving end of the this next leg, the non-compressed transmission may be compressed again (provided of course that there is at least one further decompressing means operative along the transmission path) by using the algorithm prevailing in that part of the transmission path. Naturally, this type of operation has a significant impact on the quality of the signals.
As an example of a process suffering from this drawback, let us consider a voice call between two cellular telephone users. In this example, each user is connected to a different cellular network each using a different type of compressing algorithm. When the voice call is transmitted from the first transmitting mobile station, the signals are transformed into a digital transmission representing the voice signals, and compressed by say the full rate GSM algorithm (about 13.8 kbit/s). Other processes in turn change this digital information into a radio signal.
After being detected by a base station antenna, the signals are processed and the digital signals representing the voice, which are delivered to a speech transcoder are recovered therefrom. By this example, the two end networks operate each at a different compression rate, thus, the encoded transmission received from the transmitting end will be decompressed by the transcoder to a rate of 64 Kbit/s, and be routed through the Mobile services Switching Center (“MSC”) and various links and switches towards the receiving end network. There, the transmission will be compressed again in accordance with the operating coding algorithm used in that second network, say the half rate GSM algorithm, and be transmitted to the receiving mobile station by the reverse process of the one described above. In reality, such a process may be more complicated than that described above. For example, the transmission path may further comprise PSTN links that include one or more pairs of compressing/decompressing devices. In this latter case, the transmission will be subjected to one or more additional compressing/decompressing cycles. The end result of such a process, is a very poor call quality with highly distorted signals. In order to retain a good quality of service, it would be highly beneficial to minimize the number of these compressing/decompressing cycles, as each of them deteriorates the signals quality.
A similar problem exists in the solution suggested in U.S. Pat. No. 4,890,282. This patent discloses a network wherein the mode of the speech compression in a channel carrying speech information is determined by the ports through which it is transmitted. Once the link across which data will be transmitted, is determined, the mode of data compression is determined. By the embodiments described in U.S. Pat. No. 4,890,282, the compression may be conducted either by using the digital speech interpolation (DSI) algorithm, or alternatively the AD-PCM algorithm. Once a neighboring node to which the call is to be forwarded informs the sending node that it does not have decompression resources needed to process the call, the sending node decompresses the signal, and transmit it in its decompressed form (at the rate of 64 kbit/s). As would be appreciated such solution does not solve the problem at hand, being how to reduce the number of compression/decompression cycles along the transmission path.
The present invention is therefore directed to overcome such problems, and to provide a way of forwarding the transmission along a transmission path, essentially in a compressed form.
SUMMARY OF THE INVENTION
It is therefore an object of the present invention to provide a telecommunication converter for converting signals received coded at a first code, into signals coded at a second code.
It is another object of the present invention to provide a novel digital communication system having a good quality of the signals transmitted.
Yet another object of the present invention is to provide a method for converting signals coded according to a first coding algorithm, into signals coded by another one.
Further object and features of the invention will become apparent to those skilled in the art, from the following description and the accompanying drawings.
In accordance with the present invention there is provided a converter operative in a digital telecommunication system to receive signals coded at a first compression mode and convert them into a second compression mode, which conversion leads to substantially less impairments in the decompressed form of the signals conve

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