JP2000022623A - Lsi for mobile communication terminal - Google Patents

Abstract

PROBLEM TO BE SOLVED: To reduce the hardware scale of a one-chip LSI for communication for moving object by providing a memory and an audio management control program, which is loaded on the memory, for transferring audio encoded data between a channel codec and a sound codec. SOLUTION: When A/D and D/A conversions are completed between the analog sounds of a microphone 41 and a speaker 42 and PCM data handled by a sound codec 1113, an interruption request signal IRQ is asserted to an interruption controller 112 and an audio manager 1111 is activated as interruption processing. The audio manager 1111 performs PCM data transfer between an A/D and D/A: 14 for sound and the sound codec 1113 and ADPCM data transfer between a CHC 13 and the sound codec 1113. Therefore, the hardware scale of a PHS(R) one-chip base band LSI: 1 can be suppressed just for two of channel codec LSI: 13 and MPU: 11M.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a channel codec for performing layer 1 processing of mobile communication, a voice codec for performing voice coding / decoding processing of mobile communication, layer 2 and 3 processing of mobile communication, An MPU for performing a system control process for a man-machine or the like;
I. a mobile communication terminal LSI comprising: a memory; and a voice management control program mounted on the memory and transferring voice encoded data between a channel codec and a voice codec. .

[0002]

2. Description of the Related Art A conventional mobile communication LSI includes a channel codec LSI for performing layer 1 processing of mobile communication,
Voice LS for voice coding / decoding of mobile communication
I and an MPU which performs layer 2 and 3 processing of mobile communication and system control processing such as man-machine processing.
It consists of.

On the other hand, at present, demands for lower cost and lower power consumption of mobile communication LSIs have become more severe, and as one solution to this problem, the above-mentioned conventional mobile communication LSI has a three-chip configuration. It is conceivable to make the product into one chip.

[0004]

However, in the case where the above-mentioned conventional three-chip mobile communication LSI having a three-chip configuration is integrated into a single chip without changing the conventional hardware-dependent system configuration, the following problems arise. There is.

[0005] The first problem is that if the conventional system configuration is directly integrated into one chip, the hardware scale becomes large, and it is not possible to reduce the cost and power consumption.

A second problem is that if the current system configuration depending on hardware is integrated into one chip as it is, it is difficult to change or add functions for responding to market changes.
There is a problem that a lot of time and money are required for development.

A first object of the present invention is to reduce the hardware scale of a one-chip LSI for mobile communication, thereby realizing low cost and low power consumption of the one-chip LSI for mobile communication. is there.

A second object of the present invention is to make the system configuration of a one-chip LSI for mobile communication dependent on software, thereby facilitating the change or addition of functions for responding to changes in the market. The goal is to achieve shortening.

[0009]

As means for achieving the above object, there are provided a channel codec for performing layer 1 processing of mobile communication, a voice codec for performing voice encoding / decoding processing of mobile communication, and a mobile codec. MPU that performs communication layer 2 and 3 processing and system control processing such as man-machine
And a voice management control program mounted on the memory and transferring voice encoded data between a channel codec and a voice codec.

[0010]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS A mobile communication terminal using a one-chip baseband LSI for a mobile communication terminal according to an embodiment of the present invention will be described with reference to FIGS.

FIG. 1 shows a system configuration of a mobile communication terminal using a one-chip baseband LSI for a mobile communication terminal according to an embodiment of the present invention.

The functional blocks of the mobile communication terminal include a baseband block b for performing baseband data processing,
RF block 2 for high-frequency processing and man-machine interface block 4 for man-machine interface
And consisting of The RF block 2 includes an antenna 21
And an RF 22 for controlling transmission and reception of radio waves. The man-machine interface block 4 has a microphone 41 and a speaker 42 as audio input / output devices, and a liquid crystal 43 and keys 44 as input / output devices for operation and status.

The baseband block 2 includes a one-chip baseband LSI 1 for mobile communication terminals that performs layer 1 processing of mobile communication, and layers 2 and 3 of mobile communication.
A system controller 3 that performs processing and system control processing such as man-machine processing. The system controller 3 is a memory that stores a program that operates on the MPU 11M of the one-chip microcomputer 11 in the one-chip baseband LSI 1 for mobile communication terminals.

Further, the one-chip baseband LSI 1 for mobile communication terminals includes a one-chip microcomputer 11 and a modem 12 for converting between a digital signal and a signal on a telephone line.
A channel codec 13 for performing layer 1 processing of mobile communication, a voice codec 223 for performing voice coding / decoding processing of mobile communication, and analog voice of the microphone 41 / speaker 42 and PCM data handled by the voice codec. A / D, D / A for audio that performs A / D conversion and D / A conversion between
A14.

Further, the one-chip microcomputer 11 has an M
It has a PU 11M, a memory 111, and an interrupt controller 112. Note that the MPU 1
Voice management control 1111 realized by 1M program
And an audio codec 1113.

FIG. 2 shows an operation sequence of voice processing by a voice management control program in a one-chip baseband LSI for a mobile communication terminal according to a first embodiment of the present invention.

First, audio A / D, D / A14
Means that access to PCM data handled by the audio codec 1113 becomes possible, that is, A / D conversion and D / A conversion between analog audio of the microphone 41 / speaker 42 and PCM data handled by the audio codec 1113 are performed. When the processing is completed, an interrupt request signal IRQ is asserted to the interrupt controller 112.

Second, the interrupt controller 112
When the assertion of the interrupt request signal IRQ is detected, the voice management control 1111 is activated as an interrupt process.

Third, the voice management control 1111 controls the transfer of PCM data between the A / D / D / A 14 for voice and the voice codec 1113, and the transfer of ADPCM data between the CHC 13 and the voice codec 1113. And do.

The access to the PCM data in the audio A / D and D / A 14 in the third sequence causes the interrupt request signal IRQ of the audio A / D and D / A 14 to be transmitted.
Is negated and returns to the first sequence.

According to this embodiment, in the baseband LSI for the mobile communication terminal, the voice management control for transferring the voice data and the voice codec are performed by the one-chip microcomputer in the baseband LSI for the mobile communication terminal. It can be realized by software using hardware resources.

That is, a channel codec LSI which is a conventional baseband LSI set for mobile communication terminals
, And the hardware scale of the one-chip baseband LSI for mobile communication terminals having the speech codec LSI and the MPU can be reduced to two for the channel codec LSI and the MPU.

Therefore, the hardware scale of the one-chip baseband LSI for the mobile communication terminal can be reduced, so that the price and power consumption of the one-chip baseband LSI for the mobile communication terminal can be reduced.

Also, since the system configuration of the one-chip baseband LSI for mobile communication terminals can be made dependent on software, it is easy to change or add functions for responding to market changes, and shorten the development period. realizable.

A PHS terminal using a one-chip baseband LSI for PHS according to a second embodiment of the present invention will be described with reference to FIGS. 3 to 9 and Table 1.

FIG. 3 shows a second embodiment of the present invention.
1 shows a system configuration of a PHS slave unit using a one-chip baseband LSI for S. One-chip baseband L for PHS to reduce the price and power consumption of PHS slaves
SI1, RF2, F which is a memory for user program
ROM 31, SRAM 32, SRAM 33, MMI /
F, microphone 41, speaker 42, LCD 43, ke
With a simple configuration including y44, a PHS slave device system can be realized.

For this reason, the one-chip baseband LSI 1 for PHS includes a one-chip microcomputer 11, a modem 12 for performing modulation / demodulation, a channel codec accelerator 13 for TDMA communication control, and a voice for an analog interface with a microphone / speaker. A / D, D / A14 is 1
Built in chip.

The middleware 111 using a one-chip microcomputer includes a voice manager 1111 for managing and controlling various types of voice processing, and a channel codec driver 1112 for controlling TDMA communication using a channel codec accelerator.

FIG. 4 shows a PH according to a second embodiment of the present invention.
3 shows a software hierarchical structure of a PHS slave using a one-chip baseband LSI for S. FIG.

The user software 3s includes an application 3s1, a communication protocol layer 2 and 3 function 3s2, a frame driver 3s3 for matching with the layer 1, and the like.
User software can be developed in a high-level language by allocating software that operates at a low speed, such as satisfying the response time of the s order.

On the other hand, one chip baseband LS for PHS
A / D and D for audio are stored in middleware 111 inside I1.
A response time of about several tens of us is required for performing hardware control, such as a voice manager 1111 for managing and controlling various types of voice processing using / A or the like, a channel codec driver 1112 for performing TDMA communication control using a channel codec accelerator, and the like. It incorporates Layer 1 of the PHS communication protocol that requires high-speed operation.

Here, the voice manager 1111 includes user software 3s, a channel codec driver 1112,
ADPCM encoding / decoding between ADPCM data transmitted / received by the channel codec driver 1112 and PCM data input / output by the audio A / D / D / A 14, which is located between the audio A / D and D / A 14. It controls the conversion process and various audio application processes.

FIG. 5 shows a second embodiment of the present invention.
The processing timing of the audio data in the call of the PHS slave using the one-chip baseband LSI for S is shown.

The radio access system in the PHS divides 5 ms per frame into eight slots, and uses TDMA-TDD (Ti
me Division Multiple Access−Time Division Duplexi
ng) method.

On the other hand, the processing timing of voice data in a call using the one-chip baseband LSI 1 for PHS is as follows.

The receiving system is controlled by a channel codec driver 1112 which starts in synchronization with the slot of Air.
After deciphering the ADPCM data for the 5 ms section received in the x1 slot, the deciphered ADPCM is performed by the voice manager 1111 that starts up at a period of 8 kHz.
By decoding the data and transmitting the decoded PCM data to the audio D / A 1111, the received audio for the 5 ms section is output.

On the other hand, the transmission system is a voice manager 1111
Then, after encoding the PCM data received from the audio A / D 14 for the 5 ms section, the channel codec driver 1112 performs concealment control of the ADPCM data for the 5 ms section, and then transmits the Tx1 slot.

[0038]

[Table 1]

Table 1 shows the PH of the second embodiment of the present invention.
2 shows a list of functions of audio processing of a one-chip baseband LSI for S.

Basic software that always operates during a call includes P
ADPCM CODEC that performs ADPCM encoding / decoding, which is the HS audio encoding method, and handles various audio processing.
There is a digital filter that performs frequency conversion between PCM of kHz sampling and PCM of 32 kHz sampling handled by audio A / D and D / A. In addition, in terms of audio level adjustment, there are a transmission / reception audio attenuator, a side tone, a transmission / reception audio mute, and an echo suppressor for a case where the base station does not support an echo canceller.

On the other hand, the application software operated by the user as necessary includes the following in relation to recording and reproduction:
13 kbps audio compression recording and playback with a low bit rate,
In the case of ADPCM recording / reproducing of 32 kbps and high tone quality, there are tone generation and tone detection.

FIG. 6 shows a second embodiment of the present invention.
4 shows an audio processing block diagram of a one-chip baseband LSI for S.

As shown in the figure, between the channel codec driver 1112 and the audio A / D and D / A 14,
The various audio processes shown in Table 1 are arranged.

The execution procedure of the various voice processes in the voice manager is as follows according to the voice processing flow shown in FIG.
Various audio processes such as ADPCM created in a subroutine format are called in a subroutine.

Since the transmission / reception voice attenuator, side tone, and transmission / reception voice mute processing are small in processing amount, they are incorporated in the voice manager.

FIG. 7 shows a second embodiment of the present invention.
4 shows a block diagram of a voice manager of a one-chip baseband LSI for S. FIG.

As shown in the figure, the voice manager 111
Reference numeral 1 denotes an AD which is located between the user software 3s, the channel codec driver 1112, and the audio A / D and D / A 14, and which transmits and receives between the channel codec driver 1112.
It controls ADPCM CODEC processing between PCM data and PCM data input / output between the audio A / D and D / A 14 and audio processing of various application software. Note that the channel codec driver 1112,
Operation instructions to the audio A / D, D / A 14, and various application software are performed by the user software 3s.

The interface with the user software 3s is as follows.
It accepts an instruction to start / stop the voice manager 1112. In order to accept an instruction to start / stop the voice manager 1111 from the user software 3s, a function-based voice manager start module 11112 / stop module 11113 is provided.

The interface with the audio A / D and D / A 14 is for transmitting and receiving PCM data between the audio A / D and D / A 14. Data transfer between the audio A / D and D / A 14 is performed by the serial I / O 1 in the one-chip microcomputer 11.
12 and decided to do so. Furthermore, serial I / O
The data transfer between the PCM buffer 112 and the PCM buffer 111
3 was performed. Also, transfer of PCM data between audio A / D and D / A 14 performed at a period of 32 kHz,
Since the audio processing 111111 performed at an 8 kHz cycle is performed in parallel, a PCM buffer 1111 having a double buffer configuration is used.
12 were provided.

The interface with the channel codec driver 1112 transfers ADPCM data and receives an instruction to mute received ADPCM data and an instruction to minimize audio delay. AD between channel codec drivers 1112 performed at 5 ms cycle
PCM data transfer and audio processing 1 with 8kHz cycle
To perform the operations 11111 in parallel, an ADPCM buffer 111113 having a double buffer configuration is provided.

When a bit error is detected in the received ADPCM data, the ADPCM having the bit error transferred from the channel codec driver 1112 is used to mute the audio output in the corresponding section.
A mute control flag MUTE 111111 is provided for receiving a data mute instruction. Further, as will be described later in detail, in the voice processing during a call, the voice manager uses a channel codec driver and an ADPCM COD.
ADPCM CODEC processing and ADPCM processing are performed to minimize and adjust the delay of ADPCM data processed by the DEC.
An audio delay adjustment flag DLYADJ111114 for receiving a buffer transfer start instruction is provided.

As shown in FIG. 7, the module configuration of the voice manager 1111 is composed of three modules: a start module 11112 and a stop module 11113, which are user software interfaces, and an interrupt module 11111 which performs voice processing at a frequency of 8 kHz. did.

Further, since the interrupt module 11111 for performing the audio processing 111111 is activated at a cycle of 8 kHz, the serial I / O 112 in the one-chip microcomputer 11
DMA controller 113, interrupt controller 11
4 was used. Specifically, audio A / D,
D / A14 32kHz sampling PCM data 4
DMA transfer is performed through the serial I / O 112, and an interrupt is generated at the end of the DMA transfer. The frequency conversion between the sampling frequency of 8 and 32 kHz is performed by a digital filter.

The activation module 11112 activates the interruption module 1111 of the voice manager 1111 and has a function (subroutine) format that can be called from the user software 3s.

Specifically, the interrupt controller 114
, A DMA transfer end interrupt is permitted, the DMA controller 113 sets and starts four DMA transfers between the serial I / O 112 and the PCM buffer 111112, and the A / D and D / D for audio are sent to the serial I / O 112. A
14 for serial data transfer. Also, various buffers and variables in the audio manager 1111 and ADP
It also initializes internal variables for various audio processing such as CM CODEC.

The stop module 11113 stops the interrupt module 11111 of the voice manager 1111 and has a function (subroutine) format that can be called from the user software 3s. Specifically, the DMA controller 113 disables the DMA transfer end interrupt for the interrupt controller 114, and the serial I / O 11
2. DMA transfer between the PCM buffers 111112 is stopped, and A / D and D / A
The transfer of serial data between 112 is stopped.

The interrupt module 11111 is of an interrupt routine type, and includes a PCM buffer 1111.
A / D for audio through the I / O 112 and the serial I / O 112,
It is activated every time a DMA transfer interrupt of four times of PCM data between the D / A 14 is interrupted. Specifically, as the audio A / D and D / A interface processing, the PCM data is stored and downloaded to the serial I / O 112 side buffer of the PCM buffer 111112 having the double buffer configuration, and then the DMA controller 113 The DMA transfer for the four times of transmission and reception between the serial I / O 112 and the PCM buffer 111112 is reset and restarted.

As the audio processing 111111, various audio processing such as an ADPCM codec created in a subroutine format are subroutine-called in accordance with the audio processing flow shown in FIG. Here, when encoding or decoding of the ADPCM data for the 1111 and 5 ms sections is completed, the voice manager exchanges the ADPCM data for the 5 ms section between the channel codec drivers 1112.
The ADPCM data is stored or downloaded into the channel codec driver 1112-side buffer of the M buffer 111113.

FIG. 8 shows a second embodiment of the present invention.
The control timing of the audio A / D and D / A interfaces and the interrupt module activation of the audio manager of the one-chip baseband LSI for S is shown.

In the following description, the serial I / O1
12 strobe signals STS, reception register full flag RDRF, reception register SIRDR, reception shift register SIRSR, transmission register empty flag TDR
E, transmission register SITDR, transmission shift register SI
The TSR is cited by abbreviation. Also, the serial I / O11
The illustration of the transfer clock 2 is omitted.

From the user software 3s to the activation module 11
When the 112 is called, the activation module 11112 stores the mute data in the SITDR and PCM buffer 111112, and then stores the mute data in the DMA controller 113.
And setting and activation of the serial I / O 112.

Activated serial I / O 112 and DM
The A controller 113 operates as follows.

In the receiving operation from the audio D / A 14, when the STS is asserted, the serial I / O 112 starts serial reception of the PCM data in the audio A / D 14 to the SIRSR via the SRxD and the PCM data. When all 16 bits are received, the contents of SIRSR are
After transferring to RDR, assert RDRF. Also,
Upon receiving the assertion of the RDRF, the DMA controller 113 performs a DMA transfer from the SIRDR to the PCM buffer 111112. Further, the serial I / O 112
Negates RDRF when SITDR DMA transfer is performed.

On the other hand, in the transmission operation to the audio D / A 14, upon receiving the assertion of STS, the serial I / O 112 transfers the contents of SITDR to SITSR,
Assert TDRE and SI via STxD
The serial transmission of the PCM data in the TSR to the audio D / A 14 is started. Also, the DMA controller 113
When the TDRE is asserted, the PCM buffer 111
DMA transfer from 112 to SITDR is performed. further,
When the SITDR DMA transfer is performed, the serial I / O 112 negates the TDRE.

Here, the DMA timing in the transmission / reception operation corresponds to the STS assertion,
Although the transfer is executed immediately, the DMA transfer on the receiving operation side is delayed by a transfer clock for 16 bits of PCM data.
Therefore, the interrupt module 11111 is activated upon completion of the four DMA transfers on the receiving operation side.

FIG. 9 shows a PH according to a second embodiment of the present invention.
The control timing of the channel codec driver interface during a call of the voice manager of the one-chip baseband LSI for S is shown.

Transmission / reception A in the voice manager 1111
The transfer control of the DPCM data is as follows.

In air, the AD of 5 ms section per frame
PCM is transmitted and received in Tx1 and Rx1 slots. On the other hand, the channel codec driver 1112
In the Rx2 slot, the received ADPCM data is de-concealed and the voice manager 111 of the de-concealed ADPCM data is activated.
1, and in the Rx4 slot, transfer of ADPCM data transmitted from the voice manager 1111 and confidentiality control are performed. On the other hand, in the voice manager 1111, the channel codec driver 1112
Transmission / reception ADPCM data transfer processing and transmission / reception AD
ADPCM encoding / decoding processing and various application signal processing for PCM data are performed.

That is, in the voice manager 1111,
Channel codec driver 1112 performed at a period of 5 ms
Burst transfer processing of ADPCM data between 8 kHz
ADPCM for transmission / reception ADPCM data performed periodically
It is necessary to execute two processes of encoding / decoding process and various application signal processes in parallel. For this reason, the audio manager 1111 has an ADPCM buffer for transferring transmission / reception audio data between the channel codec drivers 1112 in a double buffer configuration.
We decided to perform the following data transfer.

In the receiving system, a channel codec driver 1112 activated in synchronization with the Air slot
Is ADPC for 5ms section received in Rx1 slot
ADP for 5ms section after de-concealing M data
The CM data is stored in the reception ADPCM buffer 1RAB1. On the other hand, the voice manager interrupt module 11111 activated at a period of 8 kHz receives the ADPCM data stored in the reception ADPCM buffer 2RAB2 and receives the ADPCM data.
Using the PCM buffer 2 pointer RAB2P, samples are taken out one sample at a time, ADPCM decoding processing and various application processing are performed, and PCM data is transmitted to the audio D / A 14 to output received 5 ms section audio. The voice manager interrupt module 11
111 determines that the audio processing for the 5 ms section stored in the reception ADPCM buffer 2RAB2, that is, for all of the 40 samples, has been completed by the pointer B2P for the reception ADPCM buffer 2 and the next 5 ms stored in the reception ADPCM buffer 1RAB1. The ADPCM data for the section is received by the reception ADPCM buffer 2 RAB.
Transfer to 2.

On the other hand, the voice manager interrupt module 11111 of the transmission system performs various application processing and AD processing on the PCM data received from the voice A / D 14.
Performs PCM encoding and sends ADPCM data
ADPCM buffer 2 transmitted to PCM buffer 2 TAB2
Each sample is stored using the pointer TAB2P. Also, the voice manager interrupt module 1111
1 is an ADPC to the transmission ADPCM buffer 2 TAB2
When it is determined by the transmission ADPCM buffer 2 pointer TAB2P that the storage of the M data has been completed for the 5 ms section, that is, for all 40 samples, the ADPCM data for the 5 ms section in the transmission ADPCM buffer 2TAB2 is transferred to the transmission ADPCM buffer 1TAB1. . On the other hand, the channel codec driver 1112
After performing concealment control of the ADPCM data for the 5 ms section stored in the PCM buffer 1TAB1, the transmission is performed in the Tx1 slot.

The control for minimizing the audio delay in the audio manager is as follows.

The voice delay is the time required for the other party to hear the voice uttered by one party, and is desirably small.

Therefore, the channel codec driver 1112 and the voice manager interrupt module 1111
As a control for minimizing the audio delay between the two, the audio manager interrupt module 11111 performs audio processing on the de-concealed received ADPCM data immediately after the de-encryption processing of the channel codec driver 1112, while the secrecy of the channel codec driver 1112 is performed. ADPCM data obtained by audio processing immediately before the control processing is transferred to the channel codec driver 1112.

Here, the channel codec driver 11
12. Reception / transmission ADPCM for controlling voice delay minimization between voice manager interrupt modules 11111
Since the buffer transfer timing can be grasped only by the channel codec driver 1112, the audio delay adjustment flag is received from the channel codec driver 1112 to receive an instruction to access the reception / transmission ADPCM buffer and to start / stop the ADPCM encoding / decoding process. DLYADJ flag 111115 is provided.

Specifically, in the voice manager activation module 11112 called from the user software 3s,
Pointer RAB2P for receive / transmit ADPCM buffer 2
/ TAB2P is initialized to 4 and 3, respectively, and DL
Reception / transmission ADPC by YADJ flag 111115
M buffer 2 RAB2 / TAB2 access and ADP
In a state where the CM encoding / decoding process is started / stopped, the voice manager interrupt module 11111 is started. On the other hand, the activation setting of the DLYADJ flag 111115 is determined by the R code of the channel codec driver 1112.
It was decided to perform it at the time of confidentiality control performed by x4.

As a result, the voice manager interrupt module 11111
Immediately after the concealment release processing of No. 12, the pointer RAB2P for the reception ADPCM buffer 2 becomes 0, while immediately before the concealment control processing of the channel codec driver 1112, the pointer TAB2P for the transmission ADPCM buffer 2 becomes 0. That is, in the reception / transmission, the delay of the ADPCM data transfer between the channel codec driver 1112 and the voice manager interrupt module 11111 can be reduced.

As shown, a margin is actually provided. This is because multiple occurrences of higher-order interrupts and jitter of interrupts are taken into account.

The automatic reception mute control in the voice manager is as follows.

The automatic reception mute control means that the reception ADPC
For example, when a bit error is detected in the M data, it is necessary to mute the audio output in the section corresponding to this, and this control is performed.

Since the ON / OFF instruction of the automatic reception mute control can be grasped only by the channel codec driver 1112, the channel codec driver 111
2, a MUTE flag 111114 is provided as a mute control flag to accept an instruction to mute the received ADPCM data.

More specifically, the voice manager interrupt module 11111 controls the reception ADPCM buffer 1RAB.
The MUTE flag 111114 is determined at the time of data copy processing from 1 to the reception ADPCM buffer 2RAB2, and if mute is on, the reception ADPCM buffer 2RAB2
Stores mute data. On the other hand, in the case of mute off, the reception ADPM buffer 1
The data is copied to the CM buffer 2RAB2.

According to the present embodiment, the layer 1 of the mobile communication
Mobile communication including a channel codec for performing processing, a voice codec for performing voice coding / decoding processing for mobile communication, and an MPU for performing system control processing such as layer 2 and 3 processing of mobile communication and a man-machine. In a terminal LSI, a voice management control function for managing and controlling voice data transfer between a channel codec and a voice codec and between voice A / D and D / A and a voice codec;
The ADPCM codec, tone generator, tone detector, attenuator, side tone, filter, echo suppressor, and voice compression, which are HS voice functions, can be implemented by a program, and can be mounted in the memory in the LSI.

Therefore, since the hardware scale of the one-chip LSI for PHS can be reduced, the price and power consumption of the one-chip LSI for PHS can be reduced.

Further, since the system configuration of the PHS one-chip LSI can be made dependent on software, it is easy to change or add functions for responding to market changes.
The development period can be shortened.

[0086]

According to the present invention, layer 1 of mobile communication is provided.
Mobile communication including a channel codec for performing processing, a voice codec for performing voice coding / decoding processing for mobile communication, and an MPU for performing system control processing such as layer 2 and 3 processing of mobile communication and a man-machine. In a terminal LSI, a voice management control function for managing and controlling voice data transfer between a channel codec and a voice codec can be realized by a program and mounted on a memory in the LSI.

Therefore, the hardware scale of the mobile communication terminal LSI can be reduced, so that the cost and power consumption of the mobile communication terminal LSI can be reduced.

Further, since the system configuration of the mobile communication terminal LSI can be made dependent on software, it is easy to change or add functions for responding to market changes.
The development period can be shortened.

[Brief description of the drawings]

FIG. 1 is a system configuration diagram of a mobile communication terminal using a one-chip baseband LSI for a mobile communication terminal according to an embodiment of the present invention.

FIG. 2 is a diagram showing an operation sequence of voice processing by a voice management control program in a one-chip baseband LSI for a mobile communication terminal according to a first embodiment of the present invention.

FIG. 3 is a system configuration diagram of a PHS slave device using a one-chip baseband LSI for PHS according to a second embodiment of the present invention.

FIG. 4 is a diagram showing a software hierarchical structure of a PHS slave device using a one-chip baseband LSI for PHS according to a second embodiment of the present invention.

FIG. 5 is a timing chart of processing voice data in a call of a PHS slave unit using a one-chip baseband LSI for PHS according to a second embodiment of the present invention.

FIG. 6 shows an audio processing block diagram of a one-chip baseband LSI for PHS according to a second embodiment of the present invention.

FIG. 7 is a block diagram of a voice manager of a one-chip baseband LSI for PHS according to a second embodiment of the present invention.

FIG. 8 shows an A / D for voice of a voice manager of a one-chip baseband LSI for PHS according to a second embodiment of the present invention;
The control timing of the D / A interface and the activation of the interrupt module is shown.

FIG. 9 shows control timing of a channel codec driver interface during a call of a voice manager of a one-chip baseband LSI for PHS according to an embodiment of the present invention.

[Explanation of symbols]

DESCRIPTION OF SYMBOLS 1 ... One chip baseband LSI for PHS, 11 ... One chip microcomputer, 14 ... Audio A / D, D / A, 111 ...
Middleware, 112: Controller, 113: DMA
Controller 114 interrupt controller 111
DESCRIPTION OF SYMBOLS 1 ... Voice manager, 1112 ... Channel codec driver, 3s ... User software, 11111 ... Interrupt module, 11112 ... Activation module, 11113 ...
Stop module, 111111 ... voice processing, 11111
2 PCM buffer, 111113 ADPCM buffer, 111114 Mute flag, 111115 Audio delay adjustment flag.

Claims (8)

[Claims] 1. A channel codec for performing layer 1 processing of mobile communication, a voice codec for performing voice coding / decoding processing of mobile communication, layer 2 and 3 processing of mobile communication, and system control of man-machine and the like. An MPU for processing;
A mobile communication terminal LSI having a memory;
And a voice management control program mounted on the memory and transferring voice encoded data between a channel codec and a voice codec. 2. A channel codec for performing layer 1 processing of mobile communication, a voice codec for performing voice encoding / decoding processing of mobile communication, layer 2 and 3 processing of mobile communication, and system control such as man-machine. An MPU for processing;
A mobile communication terminal LSI having a memory;
A moving body mounted on the memory and having a voice management control program for transferring voice coded data between a channel codec and a voice codec; and an interrupt controller for activating the voice management control program. LSI for communication terminals. 3. A channel codec for performing layer 1 processing of mobile communication, a voice codec for performing voice coding / decoding processing of mobile communication, layer 2 and 3 processing of mobile communication, and system control of a man-machine. An MPU for processing;
A mobile communication terminal LSI having a memory;
A voice management control program mounted on the memory, and for transferring voice encoded data between a channel codec and a voice codec; and a voice management control program when receiving a voice activation request from voice activation request control means. And an audio activation request control means for generating an audio activation request as an activation request of the audio management control program to the interrupt controller when the PCM data handled by the audio codec is accessible. Characteristic LSI for mobile communication terminals. 4. A channel codec for performing layer 1 processing of mobile communication, a voice codec for performing voice encoding / decoding processing of mobile communication, layer 2 and 3 processing of mobile communication, and system control of man-machine and the like. An MPU for processing;
A mobile communication terminal LSI having a memory;
Mounted in the memory, and transfer voice encoded data between a channel codec and a voice codec, and,
PCM between audio A / D, D / A and audio codec
A voice management control program for transferring data, and a voice A
/ D, an interrupt controller that starts a voice management control program when a voice activation request is received from D / A, analog voice of a microphone / speaker of a mobile communication terminal, and PCM data handled by a voice codec of a mobile communication. When the A / D conversion or D / A conversion between the PC and the PCM data is enabled by the completion of the A / D conversion or the D / A conversion, the voice management control program is transmitted to the interrupt controller. When an audio activation request is generated as an activation request and the audio management control program accesses the PCM data, the audio A / D, D for ending the audio activation request is terminated.
/ A for mobile communication terminals, comprising:
SI. 5. The mobile communication terminal LSI according to claim 1, wherein the voice management control program manages at least one of initialization control, activation control, and stop control of the voice codec. An LSI for mobile communication characterized by controlling. 6. The mobile communication terminal LSI according to claim 1, wherein the voice function other than the voice codec of the mobile communication includes a tone generator, a tone detector, a voice detector, and an attenuator. , Side tone, filter, echo canceller, echo suppressor, synthesizer, voice recognition, voice synthesis, voice compression, replacement of received voice data at the time of a reception error, the voice management control program includes a voice codec and A mobile communication system for transferring PCM with other voice functions and managing and controlling at least one of initialization control, activation control, and stop control of voice functions other than the voice codec. LSI. 7. The mobile communication terminal LSI according to claim 1, wherein a voice codec, a tone generator, a tone detector, and a voice function, which are voice functions of the mobile communication, are provided.
Voice detector, attenuator, side tone, filter,
Echo canceller, echo suppressor, synthesizer,
An LSI for mobile communication, wherein at least one of voice recognition, voice synthesis, voice compression, and replacement of received voice data at the time of a reception error is realized by a program, and is mounted on the memory. 8. The mobile communication terminal LSI according to claim 1, wherein the audio codec is AD.
An LSI for mobile communication, which is a PCM codec.