TECHNICAL FIELD

The present invention relates to method and device for enhancing speech properties in a mobile device.

BACKGROUND

Adaptive Multirate Wide Band (AMR-WB) is a speech-compression algorithm. It offers substantially better voice quality (even in noisy environment) because of doubled throughput, without extra radio and transmission bandwidth requirements.

It is standardized in 3GPP Rel-5 and applicable in 3GPP mobile circuit switched systems (GSM, WCDMA) as well as packet switched systems (IMS Telephony, VoIP). AMR-WB comprises nine coding rates including the first three rates 6.60, 8.85 and 12.65kbps, which make up the mandatory multi-rate configuration.

The ongoing evolution of wireless communication systems and mobile phones has given rise to a variety of compelling mobile applications (music player, camera, game console) and services (mobile internet, mobile TV, and so on). Likewise, many services have evolved significantly in order to satisfy user demands. In contrast, from a user

perspective, voice telephony has not changed noticeably since mobile telephony was still very young. Notwithstanding, voice service has continued to evolve. Significant milestones include the introduction of the enhanced full-rate codec (EFR) and, later, the Adaptive Multirate (AMR) voice codec, which increased voice quality and boosted channel error robustness and capacity. The narrowband AMR (AMR-NB) codec, which supports the bandwidth of traditional telephony, is now widely deployed in GSM/EDGE and UMTS systems. It is also the codec of choice for the forthcoming multimedia telephony service for IMS (MTSI) standard from 3GPP.

The new wideband AMR (AMR-WB) codec, whose voice frequency band is twice that of AMR-NB, enables telephony services with true, natural voice quality, clearly outperforming other existing mass-market telephony services, including those used for wire-line telephony.

However, there excites a problem when, for example a caller changes between cells. When an AMR-WB call is transferred into an AMR-NB call there is an audible degradation in voice sound quality.

The principle for bandwidth extension presently used is illustrated in Fig. 1. An incoming AMR NB call 5 to the device is processed to generate a high frequency element in an non-linear element 6 and then filtered using a multi-tap FIR (Finite Impulse Response) filter 7 for overtone shaping, which is added 8 to the incoming AMR-NB call to produce a call with fixed bandwidth and bandwidth extension. The result 9 is a call with extension added fixed bandwidth. SUMMARY

One objective of the present invention is to overcome the audio degeneration as mentioned earlier.

Existing technologies for bandwidth extension uses a fix set of filter parameters to extend bandwidth. The proposed method of the invention utilizes the ongoing call to extract optimum filter parameters.

The advantage of the proposed method gives, amongst others, a better, more natural optimized bandwidth extension for the callers involved, and hence a less perceived degradation when a call is transferred from AMR WB to AMR NB.

Caller optimized bandwidth extension filters according to the present invention are of higher audible quality than standard filters with fixed parameters, and may be optimized to fit every voice fair.

At least for these reasons a method of improving quality of a voice transmission, the method comprising: extracting filter coefficient parameters with respect to a voice signal in a first speech transmission rate, and using the extracted filter coefficient parameters in a second transmission rate, the second transmission rate being equal or lower than the first transmission rate. The first transmission rate uses Adaptive Multirate Wide Band (AMR- WB) or Adaptive Multirate Full Rate (AMR-FR).The second transmission rate uses Adaptive Multirate Narrow Band (AMR-NB) or Adaptive Multirate Half Rate (AMR-HR).

The method may further comprise steps of: filtering the transmission in the first transmission rate filtered and extracting a signal in the second transmission rate, providing the extracted signal in the second transmission rate to a nonlinear element for bandwidth extension, providing and original transmission and output from the filter to a comparator, providing output of the comparator, which is a difference between the original transmission signal and output of the filter with added bandwidth extension to a LMS calculator, providing output of the LMS calculator to a filter coefficient adapter, in which the coefficients in the bandwidth extension filter is adapted to optimize the LMS value, and providing the output from the filter coefficient adapter to the filter. The filter may be a FIR filter. In one embodiment filter coefficients are stored for different voices with respect to incoming unique identity and/or voice recognition when available first transmission rate available.

The invention also relates to an arrangement for enhancing quality of a voice transmission in a communication device, the arrangement comprising: a first portion for extracting filter coefficient parameters with respect to a speech signal in a first transmission rate, and a second portion for using the extracted filter coefficient parameters as a reference value in a second transmission rate, the second transmission rate being equal or lower than the first transmission rate. The arrangement may comprise a fixed filter, a nonlinear element, a Multi-tap FIR filter, a FIR filter coefficient adapter, a comparator and an arrangement for optimizing filter coefficients to minimize differences between original and created signals.

The invention also relates to a mobile communication device comprising a housing, a display, a keypad, a microphone, an ear-piece, an antenna, a radio interface circuitry, a codec circuitry, and a controller and a memory, wherein the controller is configured to extract filter coefficient parameters with respect to a voice signal in a first transmission rate, and use the extracted filter coefficient parameters as a reference value in a second transmission rate, the second transmission rate being equal or lower than the first transmission rate. The invention also relates to a computer program comprising program code means for improving quality of a voice transmission when run on a computer. The computer program comprises: code for extracting filter coefficient parameters with respect to a voice signal in a first speech transmission rate, and a code for using the extracted filter coefficient parameters in a second transmission rate, the second transmission rate being equal or lower than the first transmission rate.

The invention also relates to a computer product comprising program code means stored on a computer readable medium, when the program product is run on a computer, for performing improvement of quality of a voice transmission when run on a computer. The computer program comprises: code for extracting filter coefficient parameters with respect to a voice signal in a first speech transmission rate, and a code for using the extracted filter coefficient parameters in a second transmission rate, the second transmission rate being equal or lower than the first transmission rate.

BRIEF DESCRIPTION OF THE DRAWINGS

The invention is described in a non-limiting way with respect to a number of exemplary embodiments, in which: Fig. 1 is a diagram of principle for bandwidth extension according to prior art,

Fig. 2 is a schematic view of a communication network implementing a device according to the present invention,

Fig. 3 is a schematic diagram of an arrangement for bandwidth extension according to present invention,

Fig. 4 is a schematic view of an electronic device that can be used in conjunction with the implementation of various embodiments of the present invention;

Fig. 5 is a schematic representation of the circuitry which may be included in the

electronic device of Fig. 4; and

Fig. 6 is a schematic flow diagram illustrating the main steps of the method of the

invention. DETAILED DESCRIPTION

Fig. 2 illustrates one example of an application of the present invention. A mobile device 10, such as a cell phone used by a caller moves within a first cell 30 covered with a base station 31 towards a second cell 40 covered with a base station 41. The voice

transmission 32 uses AMR WB and when the caller changes between cells 30 and 40 and thus between the base stations 31 and 41 , the AMR WB call may be transferred into an AMR NB call, e.g. due to signal strength, handover procedure, etc., and an audible degradation in voice sound quality may be experienced as the quality of AMR WB is usually superior to AMR NB.

According to the present invention, to make this degradation less experienced and annoying, the bandwidth extension filtering techniques are used.

During an ongoing call on high quality (AMR WB/FR), it is possible to adjust filtering properties towards an ongoing reference, to create filters (maximally) adapted to the involved callers. Fig. 3 illustrates schematically one exemplary embodiment of an adaptive bandwidth extension optimization arrangement according to the present invention, when a reference call is available.

The arrangement 100 comprises AMR NB fixed filter 110, a non-linear element 115, a Multi-tap FIR filter 120, FIR filter coefficient adapter 130, a comparator 140 and a Least Means Squared (LMS) calculator 150. As LMS algorithms are used in adaptive filters to find the filter coefficients that relate to producing the least mean squares of the error signal, difference between the desired and the actual signal, other types of filter/calculators may be used, for example but not exclusively Normalized least mean squares filter (NLMS), Recursive least squares ( RLS), Wiener filter, Multi-delay block frequency domain adaptive filter (MDF). The ongoing AMR WB call signal 101 in the device is filtered in the AMR NB fixed filter 10 such that an AMR NB call signal is extracted and provided to a nonlinear element 1 15, which creates a signal with high frequency and wideband content out of the low frequency (narrowband) input signal and bandwidth extraction. The result is provided to the Multi-tap FIR filter 120 for tuning. The FIR filter fine tunes the extended frequency content to sound as natural as possible. Thus, filter coefficients are optimized to minimize differences between original and created signals.

The output of the FIR filter 120 is provided to a comparator 140, which compares the fine- tuned output from the filter 120 to the original AMR-WB call signal 101.

The output 141 of the comparator 140, which is the difference between the original AMR- WB call signal and the AMR-NB with added bandwidth extension from the FIR filter 120 is compared in the LMS calculator 150, e.g. using a LMS algorithm or similar. The output 151 of the LMS calculator 150 is provided to the FIR filter coefficient adapter 130, in which the coefficients in the bandwidth extension FIR filter is adapted to optimize the LMS value. The outputs 151 , FIR filter coefficients, from the FIR filter coefficient adapter 130 are provided to the FIR filter 120. The parameters are compared (e.g. in LMS calculator) and parameters resulting optimal values are stored.

The FIR filter may be design using one or several of, for example: Parks-McClellan, Windowing, or Direct Calculation. Of course other methods suitable for the invention may be used. Other filters with same functionality may be used to substitute FIR filter.

Thus, the invention suggests, extracting filter parameters for the received voice call during an AMR WB (high quality) call which is assumed to have better quality. These are then stored during the call session and used for bandwidth extension if the call is routed over to a channel with a lower bandwidth (AMR NB) is used.

Thus, a "default filter" can be used when, for example, a user puts a call for the first time in an AMR NB connection and there are no "out-filtered" optimized filter coefficients. According to one embodiment of the invention, the filter coefficients may be stored for different callers, e.g. with respect to incoming phone number and/or voice recognition or any other unique identity, etc. to be used for AMR NB calls when available.

Figs. 4 and 5 show one representative mobile device 10 within which the present invention may be implemented. It should be understood, however, that the present invention is not intended to be limited to one particular type of electronic device. The mobile device 10 of Figs. 3 and 4 includes a housing 1 1 , a display 12, e.g. in the form of a liquid crystal display, a keypad 13, a microphone 14 and an ear-piece 15, an antenna 16, radio interface circuitry 17, codec circuitry 18, a controller 19 and a memory 20. Individual circuits and elements are all of a type well known in the art, for example in the Sony Ericsson Mobile Communications range of mobile telephones.

In summary and as an general example of the present invention, as illustrated in flow diagram of Fig. 6, the method of improving quality of the voice transmission comprises: extracting filter coefficient parameters 601 with respect to a voice signal in a first speech transmission rate, using 609 the extracted filter coefficient parameters in a second transmission rate, filtering 602 the transmission in the first transmission rate and extracting a signal in the second transmission rate, providing the extracted signal in the second transmission rate to a non-linear element for bandwidth extension 603, fine-tuning 604 output from said non-linear element in a filter, providing the original transmission and output from said filter to a comparator and comparing 606, providing output of the comparator, which is a difference between the original transmission signal and output of said filter with added bandwidth extension to a LMS calculator 606, providing the output of the LMS calculator to a filter coefficient adapter, in which the coefficients in the bandwidth extension filter is adapted 607 to optimize the LMS value, and providing the output from the filter coefficient adapter to the filter 608.

The invention may be implemented in the controller and Codec parts of the device. The invention may as well be implemented in systems using AMR FR (Full Rate) and AMR HR (Half Rate).

The various embodiments of the present invention described herein is described in the general context of method steps or processes, which may be implemented in one embodiment by a computer program product, embodied in a computer-readable medium, including computer-executable instructions, such as program code, executed by computers in networked environments. A computer-readable medium may include removable and non-removable storage devices including, but not limited to, Read Only Memory (ROM), Random Access Memory (RAM), compact discs (CDs), digital versatile discs (DVD), etc. Generally, program modules may include routines, programs, objects, components, data structures, etc. that perform particular tasks or implement particular abstract data types. Computer-executable instructions, associated data structures, and program modules represent examples of program code for executing steps of the methods disclosed herein. The particular sequence of such executable instructions or associated data structures represents examples of corresponding acts for implementing the functions described in such steps or processes.

Software and web implementations of various embodiments of the present invention can be accomplished with standard programming techniques with rule-based logic and other logic to accomplish various database searching steps or processes, correlation steps or processes, comparison steps or processes and decision steps or processes. It should be noted that the words "component" and "module," as used herein and in the following claims, is intended to encompass implementations using one or more lines of software code, and/or hardware implementations, and/or equipment for receiving manual inputs.

The foregoing description of embodiments of the present invention, have been presented for purposes of illustration and description. The foregoing description is not intended to be exhaustive or to limit embodiments of the present invention to the precise form disclosed, and modifications and variations are possible in light of the above teachings or may be acquired from practice of various embodiments of the present invention. The

embodiments discussed herein were chosen and described in order to explain the principles and the nature of various embodiments of the present invention and its practical application to enable one skilled in the art to utilize the present invention in various embodiments and with various modifications as are suited to the particular use

contemplated. The features of the embodiments described herein may be combined in all possible combinations of methods, apparatus, modules, systems, and computer program products.