CN1917039B - Voice quality enhancement method and system - Google Patents

Abstract

The invention provides a voice quality enhancement method and a system, and the core of the method is as follows: the control end determines that voice quality enhancement processing needs to be carried out on the voice stream, the control end loads control information of the voice quality enhancement processing needing to be carried out in a control packet and transmits the control information to an execution end of the voice quality enhancement processing, and the execution end carries out corresponding voice quality enhancement processing on the received voice stream according to the control information loaded in the control packet. The control end in the invention controls the voice quality enhancement execution end to carry out voice quality enhancement processing such as acoustic echo cancellation and the like through the control packet, so that the method of the invention can be applied to a 2G/3G network with a separated framework, thereby realizing the purpose of improving the voice quality.

Description

Method and system for enhancing voice quality

technical field

The invention relates to the technical field of network communication, in particular to a voice quality enhancement method and system.

Background technique

During the user's call, the user's voice in the handset of the mobile phone or landline will spread into the room, and will also reach the microphone through the inside of the phone, so that the speaker will hear his own voice coming from the other party's handset.

This phenomenon is not felt by the user when the line delay is relatively small. When the line delay is relatively large, the user will hear his own voice coming from the receiver of the other party, especially in the hands-free mode. This problem is more obvious.

When the channel delay in mobile communication is relatively large, an acoustic echo cancellation technology, that is, an AEC (Acoustic Echo Cancellation) technology is usually used to improve voice communication quality.

AEC technology is a voice quality enhancement method. In addition to AEC technology, there are the following voice quality enhancement methods:

NR (Background Noise Suppression): NR can eliminate the interference noise of the background environment during the call, so that the other party can hear the call clearly.

ALE (Automatic Local Earpiece Gain Control): The ALE function can automatically adjust the gain of the user's hearing sound according to the noise intensity of the user's environment, so that the user can hear the call content clearly in a noisy environment.

ALG (Automatic Gain Control): The ALG function can automatically adjust the volume of the user's voice. When the voice of the call becomes small, it can automatically adjust the gain so that the other party will not hear the voice too low or too high.

VEI (Voice Enhanced Identification): Can make speech easy to understand.

DD (Data Service Detection): It can detect the in-band data service in the voice signal and transparently transmit the in-band data service.

EEC (Electrical Echo Cancellation): It can offset the electrical echo generated by the analog telephone 2-wire to 4-wire, usually with different tail lengths.

The voice quality enhancement function mentioned above has working directions such as two-way, internal, external, etc. during use, that is, the voice quality enhancement function can work in two directions, can work internally, and can work externally.

At present, the methods for enhancing voice quality by using the above-mentioned voice quality enhancement functions such as acoustic echo cancellation technology, NR, and ALE are as follows:

Integrate voice quality enhancement functions such as AEC, NR, ALE, ALG, VEI, DD, EEC into the codec, such as integrating voice quality enhancement functions such as AEC, NR, ALE, ALG, VEI, DD, EEC into BSS ( base station subsystem) codec, and integrate voice quality enhancement functions such as AEC, NR, ALE, ALG, VEI, DD, EEC into the codec on the MSC (Mobile Switch) to eliminate mobile users to the core Acoustic echo in the direction of the net to improve voice quality.

At the same time, the AEC function integrated in the codec can also identify the FAX (fax) service and data service, realize automatic opening and closing of the AEC function, etc.

Although the above speech quality enhancement method can obviously improve the speech quality, the implementation of this method will reduce the density and processing capacity of the codec, and it is not suitable for use in the 2G/3G separate architecture.

Contents of the invention

The object of the present invention is to provide a voice quality enhancement method and system, the control terminal controls the executive terminal to enhance the voice quality through the control packet, so that the method of the present invention can be applied in the separation framework of 2G/3G, thereby realizing the improvement Voice quality purposes.

In order to achieve the above object, a method for enhancing voice quality provided by the present invention includes:

a. The control terminal determines that it is necessary to perform voice quality enhancement processing on the voice stream;

b. The control terminal carries the control information of the voice quality enhancement process to be carried out in a control packet and transmits it to the execution terminal of the voice quality enhancement process;

c. The executing end performs corresponding voice quality enhancement processing on the received voice stream according to the control information carried in the control packet.

The control end includes: a call server.

The execution end of the voice quality enhancement processing includes: a media gateway in a 2G/3G network or a media gateway in an NGN network based on a split architecture mode.

Described step a specifically comprises:

The control end determines that voice quality enhancement processing needs to be performed on the voice stream of the calling side/called side according to the office route information and/or user subscription information of the calling side/called side.

The control packet in the step b is: a control packet based on H.248.

The control information of the voice quality enhancement processing required in step b includes: turning on/off acoustic echo cancellation, and/or electrical echo suppression depth, and/or turning on/off background noise suppression, and/or turning on/off automatic Gain control, and/or automatic local earpiece gain control on/off, and/or speech enhancement recognition on/off, and/or data traffic detection on/off.

The control information that requires speech quality enhancement in step b also includes: the direction of acoustic echo cancellation, and/or the direction of automatic gain control, and/or the direction of automatic local earpiece gain control, and/or the direction of speech enhancement recognition , and/or the direction of data service detection.

When the electrical echo suppression depth carried in the control information is a predetermined value, the execution end of the speech quality enhancement process independently determines the electrical echo suppression depth according to needs.

Described step c specifically comprises:

The execution end allocates corresponding voice quality enhancement resources according to the control information carried by the control packet, and performs voice quality enhancement processing on the externally transmitted G.711-based voice stream.

Described step c specifically comprises:

The execution end allocates corresponding voice quality enhancement resources and transcoder resources according to the control information carried by the control packet;

The code converter resource converts the externally transmitted voice stream into a G.711 voice stream, and the voice quality enhancement resource performs corresponding voice quality enhancement processing on the G.711 voice stream.

The present invention also provides a voice quality enhancement system, including:

Control terminal: when it is determined that voice quality enhancement processing needs to be performed on the voice stream, the control information of the voice quality enhancement processing that needs to be performed is carried in the control packet and transmitted to the execution end of voice quality enhancement processing;

Executing end: perform corresponding voice quality enhancement processing on the voice stream according to the control information carried in the control packet it receives.

The control terminal includes:

Determination module: according to the office route information and/or user subscription information of the calling side/called side, it is determined that voice quality enhancement processing needs to be performed on the voice stream of the calling side/called side, and notifies the control information delivery module;

The control information delivery module: according to the notification of the determination module, the control information of the voice quality enhancement processing that needs to be performed is carried in the control packet and transmitted to the execution end.

The execution end includes:

Resource allocation module: according to the control information carried in the control packet transmitted by the control terminal, allocate corresponding voice quality enhancement resources to the execution module;

Execution module: perform voice quality enhancement processing on the G.711 voice stream according to the resources allocated by the resource allocation module.

The execution terminal also includes:

Conversion module: convert the externally transmitted voice stream into a G.711-based voice stream, and transmit it to the execution module.

The control terminal is: call server;

The execution end is: a media gateway in a 2G/3G network, or a media gateway in an NGN network.

It can be seen from the description of the above technical solution that the control terminal of the present invention controls the execution terminal to enhance the voice quality through the control packet, so that the method of the present invention can be applied in the 2G/3G network of the separated architecture; through the execution of the voice quality enhancement process Terminals such as media gateways etc., carry out voice quality enhancement processing on voice streams, avoiding the impact on the density and processing capacity of the codec; the control terminal of the present invention carries out one or more voice quality enhancement functions in the control packet Control the information, and set its working direction, so that the execution end such as the media gateway can complete the voice quality enhancement processing through simple command execution, which simplifies the voice quality enhancement operation of the media gateway; thus the technical solution provided by the invention realizes the improvement of voice quality. purpose of quality.

Description of drawings

Fig. 1 is a schematic diagram of a voice quality enhancement method in a 2G network of the present invention;

Fig. 2 is a schematic diagram of a voice quality enhancement method in a 3G network of the present invention;

Fig. 3 is a schematic diagram of the voice quality enhancement method in the intercommunication between the 3G network and the PSTN network of the present invention.

Detailed ways

The core of the present invention is: the control terminal determines that the voice quality enhancement needs to be performed on the voice stream, the control terminal carries the control information of the voice quality enhancement that needs to be performed in the control packet and transmits it to the execution terminal of the voice quality enhancement processing, and the execution The terminal performs voice quality enhancement processing on the received voice stream according to the control information carried in the control packet.

The technical solution provided by the present invention will be further described below based on the core idea of the present invention.

In the 2G/3G network under the split architecture mode, the control end of the present invention may be a call server, and the execution end may be a media gateway. In an NGN network, the execution end of voice quality enhancement processing may be a media gateway.

The call server can determine whether to perform voice quality enhancement processing on the voice stream according to the office route information of the calling user, the called user, and user subscription information. If the call server determines that the voice stream needs to be voice quality enhanced, it also needs to Part of the voice quality enhancement function provides direction indication to indicate whether it is one-way voice quality enhancement processing or two-way voice quality enhancement processing, and the call server transmits the control information of voice quality enhancement processing to the media gateway through the control packet based on H.248.

The control information of the voice quality enhancement process carried in the H.248-based control packet can include one or more of the following voice quality enhancement functions: enable or disable the AEC processing function, the depth of electrical echo suppression, and enable or disable the background noise Inhibit, turn on or off the automatic gain control, turn on or off the automatic local earpiece gain control, turn on or off the speech enhancement recognition, turn on or off the data traffic detection, etc.

In addition, when the voice quality enhancement control information of the control packet includes enabling automatic gain control, enabling automatic local receiver gain control, enabling speech enhancement recognition, or enabling data service detection, the voice quality enhancement function should also be indicated in the control information The effective direction of the device, that is, the direction of the automatic gain control, the direction of the automatic local earpiece gain control, the direction of the speech enhancement recognition, and the direction of the data service detection.

The depth of electrical echo suppression in the control information can be: 16ms, 32ms, 48ms, 64ms, 96ms or 128ms, and the call server can according to the call characteristics of the calling side/called side, such as the office route information of the calling side/called side, the user Subscription information, etc. to determine the depth of echo suppression for the voice stream of the calling side/called side.

The control package enhance ec package (enhanced echo suppression package) of the present invention based on H.248 can be defined as following form:

PropertyID: aec(0x0001)

Type: integer

Possible values: 0 means off, 1 means on. The default is off.

Defined in: LocalControlDescriptor

Characteristics: read/write

Property ID: aec_direction (0x0002)

Type: integer

Possible values: 0 means incoming, 1 means outgoing, 2 means both, and the default value is inward.

Description: the direction of aec

Defined in: LocalControlDescriptor

Characteristics: read/write

Property ID: eec_depth (0x0003)

Type: integer

Possible values: 0 means that the media gateway chooses the depth of the electrical echo suppression function according to actual needs, 1 means 16ms EEC, 2 means 32ms EEC, 3 means 48ms EEC, 4 means 64ms EEC, 5 means 96msEEC, 6 means 128msEEC, the default value is 0;

Defined in: LocalControlDescriptor

Characteristics: read/write

PropertyID: nr(0x0004)

Description: Background noise suppression

Type: integer

Possible values: 0 (off) means off, 1 (on) means on. The default is off.

Defined in: LocalControlDescriptor

Characteristics: read/write

PropertyID: nr_dir (0x0005)

Description: The direction of action of nr

Type: Enumeration

Possible values: incoming (0), outgoing (1), both (2). Default: both

Defined in: LocalControlDescriptor

Characteristics: read/write

PropertyID: ale(0x0006)

Description: Automatic local earpiece gain control

Type: integer

Possible values: 0 (off) means off, 1 (on) means on, the default value is off.

Defined in: LocalControlDescriptor

Characteristics ：read/write

PropertyID: ale_dir (0x0007)

Description: The direction of action of ale

Type: Enumeration

Possible values: 0 means incoming, 1 means outgoing, 2 means both, and the default value is both.

Defined in: LocalControlDescriptor

Characteristics: read/write

PropertyID: alg(0x0008)

Description: automatic gain control

Type: Enumeration

Possible values: 0 means incoming, 1 means outgoing, 2 means both, and the default value is both.

Defined in: LocalControlDescriptor

Characteristics: read/write

Property ID: alg_dir (0x0009)

Description: The direction of action of alg

Type: Enumeration

Possible values: 0 means incoming, 1 means outgoing, 2 means both, and the default value is both.

Defined in: LocalControlDescriptor

Characteristics: read/write

PropertyID: vei(0x000A)

Description: speech enhancement recognition

Type: integer

Possible values: 0 (off) means off, 1 (on) means on, the default value is off.

Defined in: LocalControlDescriptor

Characteristics: read/write

PropertyID: vei_dir (0x000B)

Description: The direction of action of vei

Type: Enumeration

Possible values.: 0 means incoming, 1 means outgoing, 2 means both, and the default value is both.

Defined in: LocalControlDeseriptor

Characteristics: read/write

Property ID: dd (0x000C)

Description: Data service detection

Type: integer

Possible values: 0 (off) means off, 1 (on) means on. The default is off

Defined in: LocalControlDescriptor

Characteristics: read/write

Property ID: dd_dir (0x000D)

Description: The direction of action of dd

Type: Enumeration

Possible values: 0 means incoming, 1 means outgoing, 2 means both, and the default value is both.

Defined in: LocalControlDescriptor

Characteristics: read/write

The present invention increases the type of voice enhancement by extending the H.248 protocol interface, and the opening and closing of the common electrical EC (EEC) is processed through the TDM Circuit Package (time division multiplexing circuit package) defined by the original H.248.1. The tail length of the electrical EC is extended in the ec package, that is, the depth of the electrical echo suppression. When the electrical EC is enabled with the TDM Circuit Package, the eec_depth parameter defined by the enhanceec package can be used to indicate the tail length.

Between the softswitch and the media gateway, the H.248-based control packet defined above can be used to realize the transmission of voice enhancement processing control information between the call server and the media gateway, and the media gateway allocates the voice stream according to the control information of the call server. Corresponding voice quality enhancement resources, through which the media gateway performs corresponding voice quality enhancement processing on the voice stream, so as to achieve the purpose of improving the voice quality without affecting the density and processing capability of the codec.

The technical solution of the present invention will be described in detail below in conjunction with the accompanying drawings.

In the 2G network under the split architecture mode, the implementation method of the voice quality enhancement of the present invention is shown in Fig. 1 .

In Fig. 1, both MS1 and MS2 are mobile users, BSC1 is the base station controller corresponding to MS1, and BSC2 is the base station controller corresponding to MS2. If MS1 is set as the calling user and MS2 is the called user, then BSC1 is the calling side base station controller, and BSC2 is the called side base station controller.

Between MS1 and BSC1, between MS2 and BSC2 is the air interface part, and transmits voice data through the wireless channel; the media gateway is respectively connected to BSC1 and BSC2 through the E1 interface board, and TDM (time division multiplexing) is adopted between BSC1 and BSC2 To transmit the voice flow, while the BSC and the media gateway are connected through E1, which is used to transmit the voice data flow between the BSC and the MSC.

Because the voice stream transmitted by BSC1 and BSC2 to the media gateway is the voice stream of the voice coding algorithm of G.711, and the current EC chip supports the voice stream of the voice coding algorithm of G.711, therefore, the media of the present invention The gateway can directly allocate VQE1 (Voice Quality Enhancement) and VQE2 resources to the calling side and the called side according to the control information carried in the H.248-based control packet transmitted by the call server, so as to control the E1 interface board in the media gateway. Perform voice quality enhancement processing on the voice service data flow, wherein, the VQE1 resource is used to perform voice quality enhancement processing on the voice stream of MS1, and the VQE2 resource is used to perform voice quality enhancement processing on the voice stream of MS2. The purpose of voice quality on the called side.

In the R4 network of the 3G network in the separated architecture mode, since the EC chip is required to support compressed voice, or to convert the compressed voice into a G.711-based voice stream before performing voice quality enhancement processing, it is recommended that the The voice quality enhancement processing function may not be supported in the R4 networking.

In the R99 networking of the 3G network under the separation framework mode, the implementation method of the voice quality enhancement of the present invention is shown in FIG. 2 .

In Figure 2, RNC is the radio network controller in the 3G network, RNC1 is set to be the radio network controller corresponding to the calling side wireless user terminal, RNC2 is the radio network controller corresponding to the called side wireless user terminal, and the wireless user Voice data is transmitted between the terminal and the RNC through the wireless interface. The broadband interface board between the RNC and the media gateway can use the ATM broadband interface board.

Since the voice streams transmitted by RNC1 and RNC2 to the media gateway do not use the G.711 voice coding algorithm, the media gateway of the present invention can adopt dual TC according to the control information carried in the H.248-based control packet transmitted from the call server. The method of double AEC carries out voice quality enhancement processing to the voice flow from the calling side to the called side, and from the called side to the calling side respectively, that is, the media gateway of the present invention increases the number of calls to the calling side from the IU interface of the RNC to the TDM direction. , The voice quality enhancement processing function of the voice stream on the called side. The specific method is: the media gateway directly allocates TC1, VQE1, TC2, and VQE2 resources to the calling side and the called side, so as to decode and convert the voice service data stream carried on the broadband interface board in the media gateway and to enhance the voice quality. , the TC1 resource is used to decode the calling-side voice stream sent by RNC1 to the media gateway into a voice stream based on the G.711 voice coding algorithm, and then, the VQE1 resource performs voice quality enhancement processing on the voice stream decoded by TC1, and TC2 The resource is used to decode the voice stream of the called side sent by RNC2 to the media gateway into a voice stream based on the G.711 voice coding algorithm, and then, the VQE2 resource performs voice quality enhancement processing on the voice stream decoded by TC2, thereby achieving The purpose of improving the voice quality of the calling side and the called side.

When the R99 network of the 3G network and the PSTN (Public Switched Telephone Network) in the separation framework mode intercommunicate, the voice quality enhancement method of the present invention is shown in Figure 3.

In Figure 3, RNC1 is the radio network controller in the 3G network. RNC1 is set to be the radio network controller corresponding to the wireless user terminal on the calling side. Voice data is transmitted between the wireless user terminal and RNC1 through the wireless interface, and the PSTN user is the Call the side user.

The broadband interface board between the RNC and the media gateway can use the ATM broadband interface board.

Because the speech flow that RNC1 transmits to media gateway does not adopt the speech coding algorithm of G.711, and there is no acoustic echo on the PSTN side, therefore, media gateway of the present invention can carry the weight in the control packet based on H.248 that the call server transmits. The control information of the system adopts TC+VQE mode, performs voice quality enhancement processing other than EEC on the voice stream from the calling side to the called side, and performs ordinary EEC processing on the voice stream from the called side to the calling side. The specific method is as follows: The media gateway directly allocates TC1 and VQE1 resources to the calling side and the called side, and sets the working switch and working direction of each voice enhancement technology in the VQE1 resource. To decode and convert the voice service data stream carried on the broadband interface board in the media gateway, and enhance the voice quality, among them, TC1 resources are used to decode the calling side voice stream sent by RNC1 to the media gateway into G.711-based voice coding Algorithm voice stream, then, the VQE1 resource performs voice quality enhancement processing on the voice stream decoded by TC1 according to the working parameter settings, so as to achieve the purpose of improving the voice quality of the calling side and the called side.

The voice quality enhancement system provided by the invention includes: a control terminal and an execution terminal. The control end may be a call server, and the execution end may be a media gateway, such as a media gateway in a 2G/3G network or a media gateway in an NGN network.

The control terminal is mainly used to carry the control information of the required voice quality enhancement process in the control packet and transmit it to the execution terminal of the voice quality enhancement process when it is determined that voice quality enhancement processing needs to be performed on the voice stream.

The functions of the control terminal are realized by the determination module and the control information distribution module.

The determining module is mainly used to determine the need to perform voice quality enhancement processing on the voice stream of the calling side/called side according to the office route information and/or user subscription information of the calling side/called side, and notify the control information delivery module. The control information delivery module is mainly used to carry the control information of the voice quality enhancement process required to be carried in the control packet and transmit it to the execution end according to the notification of the determination module. The content of the control information of the voice quality enhancement processing and the message structure of the control packet are as described in the above method.

The execution end is mainly used to perform corresponding voice quality enhancement processing on the voice stream according to the control information carried in the control packet it receives. The function of execution end is realized by resource allocation module, execution module and conversion module.

The resource allocation module is mainly used to allocate corresponding voice quality enhancement resources for the execution module according to the control information carried in the control packet transmitted from the control terminal, such as allocating TC1 and VQE1 resources for the calling side and the called side, as described above. in the description.

The execution module is mainly used to enhance the voice quality of the G.711-based voice stream transmitted from the outside or the G.711-based voice stream transmitted by the conversion module according to the resources allocated by the resource allocation module. The specific process is as described in the above method.

The conversion module is mainly used to convert the externally transmitted voice stream into a G.711-based voice stream and transmit it to the execution module.

While the invention has been described by way of example, those skilled in the art will appreciate that there are many variations and changes to the invention without departing from the spirit of the invention, and it is intended that such variations and changes be covered by the appended claims.

Claims (11)

1. A voice quality enhancement method, characterized in that, comprising: a. The control terminal determines that it is necessary to perform voice quality enhancement processing on the voice stream of the calling side and/or the called side according to the office route information and/or user subscription information of the calling side/called side; b. The control terminal carries the control information that needs to perform voice quality enhancement processing on the voice stream of the calling side and/or the called side in a control packet and transmits it to the execution terminal of the voice quality enhancement processing, and the control packet is based on H.248 control packet; c. The executing end performs corresponding voice quality enhancement processing on the received voice stream according to the control information carried in the control packet. 2. A method for enhancing voice quality according to claim 1, wherein the control terminal is a call server. 3. A kind of voice quality enhancement method as claimed in claim 1, is characterized in that, the execution end of described voice quality enhancement processing is: the media gateway in the 2G/3G network based on separation architecture mode or the media gateway in the NGN network gateway. 4. A kind of speech quality enhancement method as claimed in claim 1, 2 or 3, is characterized in that, the control information of the speech quality enhancement processing that needs to carry out in the described step b comprises: opening/closing acoustic echo cancellation, and and/or electrical echo suppression depth, and/or background noise suppression on/off, and/or automatic gain control on/off, and/or automatic local earpiece gain control on/off, and/or speech enhancement recognition on/off, and /or turn on/off data traffic detection. 5. A method for enhancing voice quality as claimed in claim 4, wherein the control information required to enhance voice quality in the step b further includes: the direction of acoustic echo cancellation and/or the direction of automatic gain control , and/or the direction of automatic local earpiece gain control, and/or the direction of speech enhancement recognition, and/or the direction of data service detection. 6. a kind of voice quality enhancing method as claimed in claim 4, is characterized in that: When the electrical echo suppression depth carried in the control information is a predetermined value, the execution end of the speech quality enhancement process independently determines the electrical echo suppression depth according to needs. 7. A kind of voice quality enhancement method as claimed in claim 1, 2 or 3, is characterized in that, described step c specifically comprises: The execution end allocates corresponding voice quality enhancement resources according to the control information carried by the control packet, and performs voice quality enhancement processing on the externally transmitted G.711-based voice stream; or The execution end allocates corresponding voice quality enhancement resources and transcoder resources according to the control information carried by the control packet; The code converter resource converts the externally transmitted voice stream into a G.711 voice stream, and the voice quality enhancement resource performs corresponding voice quality enhancement processing on the G.711 voice stream. 8. A voice quality enhancement system, characterized in that, comprising: Control terminal: when it is determined that the voice stream needs to be processed with voice quality enhancement, the control information of the voice quality enhancement process that needs to be performed is carried in the control packet and transmitted to the execution end of the voice quality enhancement process. The control packet is based on the H. 248 control packets; Executing end: perform corresponding voice quality enhancement processing on the voice stream according to the control information carried in the control packet it receives; The control terminal includes: Determination module: according to the office route information and/or user subscription information of the calling side/called side, it is determined that voice quality enhancement processing needs to be performed on the voice stream of the calling side/called side, and notifies the control information delivery module; Control information distribution module: According to the notification of the determination module, the control information that needs to be processed to enhance the voice quality of the voice stream of the calling side/called side is carried in the control packet and transmitted to the execution end. 9. A kind of speech quality enhancement system as claimed in claim 8, is characterized in that, described execution end comprises: Resource allocation module: according to the control information carried in the control packet transmitted by the control terminal, allocate corresponding voice quality enhancement resources to the execution module; Execution module: perform voice quality enhancement processing on the G.711 voice stream according to the resources allocated by the resource allocation module. 10. A kind of voice quality enhancement system as claimed in claim 9, is characterized in that, described execution end also comprises: Conversion module: convert the externally transmitted voice stream into a G.711-based voice stream, and transmit it to the execution module. 11. A kind of speech quality enhancement system as claimed in claim 8, 9 or 10, is characterized in that: The control terminal is: call server; The execution end is: a media gateway in a 2G/3G network or a media gateway in an NGN network.

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