JP5235309B2 - Audio playback apparatus and audio playback method - Google Patents

Description

  The present invention relates to an audio reproducing apparatus and an audio reproducing method for reproducing encoded audio data encoded using an audio codec.

  In a wireless terminal device such as PHS (Personal Handy phone System), an analog signal based on a user's voice is once converted into a digital signal and transmitted to another user's wireless terminal device, where it is converted back to an analog signal and reproduced. Is done. However, simply converting an analog signal into a digital signal increases the amount of information, and all audio data cannot be transmitted over a communication path in which the amount of information that can be transmitted is limited. Therefore, an audio codec that compresses the information amount of the audio data is necessary so that it can be transmitted through this communication path.

  The audio codec is composed of encoding (decoding) and decoding (decoding) of audio digital data, and audio data encoded by the audio encoding device is decoded and reproduced by the audio reproducing device. Examples of the voice codec used in the above-described wireless terminal device such as PHS include ADPCM (adaptive differential PCM) coding technology. Such ADPCM is a technique that encodes a difference of audio data by utilizing the property that analog data such as audio continuously changes.

The operation by ADPCM will be described. As shown in FIG. 9, in the speech encoding apparatus 10, the difference d _n between the speech data y _n to be transmitted and the previous speech data y ′ _n−1 is _expressed by a quantization width Δ _n . The quantized code is assigned, and the encoded encoded audio data c _n is transmitted to the audio reproduction device 12. Then, the audio reproduction device 12 decodes the encoded audio data c _n using the quantization width Δ _n , adds the decoded difference d ′ _n to the previous value y ′ _n−1, and adds the audio data y 'Get _n . In ADPCM, so that it can be reproduced in amplitude is small analog data has a characteristic capable of changing the resolution of the quantization in accordance with the amplitude variation, depending on the encoded audio data c _n, as shown in FIG. 9, the analog at an amplitude range of the data is large, increases the quantization width delta _n, is at a small to reduce the quantization width delta _n. By using such ADPCM, it is possible to substantially restore the audio data y _n of the speech coding apparatus 10 from the compressed small amount of information.

In the wireless terminal device, for example, 32 kbps ADPCM (ITU-T recommendation G.726) is adopted in the QPSK modulation method, and here, windowing is performed using a window function in order to perform reproduction without hearing discomfort. It is possible to generate audio data that maintains the continuity between the current frame data arranged on the time axis and the previous frame data. In such 32 kbps ADPCM voice communication, if an error occurs, the encoded voice data before decoding is replaced with zero data (0x0 or 0xF) to forcibly silence the sound and only when noise is generated. Noise can be suppressed by reducing the pressure (for example, Patent Document 1).
JP 2002-229593 A

  In the wireless terminal apparatus, from the viewpoint of effective use of the radio channel, parallel use of 16 kbps ADPCM in the BPSK modulation system is being studied in addition to the above-described 32 kbps ADPCM in the QPSK modulation system. Although 16 kbps ADPCM transmits less information per unit time than 32 kbps ADPCM, it has an advantage that communication can be performed relatively stably even when the radio wave intensity is weak.

  However, when an error occurs in such 16 kbps ADPCM, there is no zero data in the standard in 16 kbps ADPCM, and therefore, it is not possible to generate a silence state by replacing the encoded voice data with zero data as in 32 kbps ADPCM. Further, in 16 kbps ADPCM, abnormal noise occurs in the audio data even after the error is recovered due to the characteristics of the audio codec. Therefore, the voice reproduction technology of 32 kbps ADPCM cannot be simply applied to 16 kbps ADPCM.

  The present invention has been made in view of the above problems at the time of audio data reproduction in 16 kbps ADPCM, and the object of the present invention is to reproduce unpleasant (uncomfortable) audio at the time of error even when 16 kbps ADPCM is used. It is an object of the present invention to provide a new and improved audio reproduction device and audio reproduction method that can surely prevent the above.

In order to solve the above problems, according to an aspect of the present invention, a data receiving unit that receives encoded audio data encoded using an audio codec, and an error in the received encoded audio data are detected. An error detection unit; a data storage unit that stores encoded audio data in a reproduction buffer when the error detection unit does not detect an error; a decoding unit that decodes encoded audio data stored in the reproduction buffer; Audio reproduction device comprising an attenuation unit capable of attenuating audio data decoded by the unit and an audio reproduction unit for reproducing audio data, wherein the data storage unit detects an error by the error detection unit The previous value of the encoded audio data stored in the reproduction buffer is maintained, and the attenuation unit detects an error by the error detection unit and When the value is maintained, it decoded by the decoding unit, the audio data based on the previous value, and wherein the attenuating at a fixed maximum attenuation rate, the sound reproducing apparatus is provided. The audio codec may be 16 kbps ADPCM (Adaptive Differential PCM).

The ADPCM is a technique for encoding a difference between current audio data and previous audio data. Therefore, when an error is detected and the encoded audio data cannot be used, it is estimated that the audio data is continuously changing, and the previous value is maintained without rewriting the encoded audio data. Further, when an error is detected and the previous value of the encoded audio data is maintained in the reproduction buffer, the audio data based on the previous value decoded by the decoding unit is attenuated.

Damping portion is provided in front from the sound reproducing unit, a predetermined time from the recovery error, but it may also be attenuated audio data.

  Since the audio codec encodes the difference from the previous value of the audio data, it takes time to recover the original normal data once the audio data becomes an incorrect value. In particular, in the 16 kbps ADPCM, since the difference information amount is small compared to the 32 kbps ADPCM, it takes a long time to recover the audio data. With such a configuration in which the audio data is attenuated for a predetermined time, even if abnormal noise occurs in the recovery stage of the audio data, the audio is not affected, and an unpleasant audio reproduction can be prevented.

  In addition, the count starts after the error is recovered for the predetermined time, and the count value is reset at each error, so even if multiple errors occur consecutively at intervals, the last error A predetermined time is secured after the recovery of the noise, and the noise on the characteristics of the voice codec can be surely suppressed by the attenuation unit.

  The attenuation unit may gradually decrease the attenuation rate so that the attenuation rate becomes zero a predetermined time after the error is recovered.

  Here, after the error is recovered, the maximum attenuation rate is maintained before the predetermined time, and thereafter, the attenuation rate is gradually decreased to gradually increase the sound pressure (volume) of the audio data, and then attenuated at the predetermined time. The attenuation rate is forcibly shifted so that the rate becomes 0 (zero). In the 16 kbps ADPCM, there is no effect of recovering the gain with an appropriate slope unlike the window function of 32 kbps ADPCM, and thus the attenuation rate is intentionally adjusted in this way. With the configuration in which the attenuation rate is gradually reduced, the reproduction state can be smoothly recovered without a sense of incongruity.

In order to solve the above problems, according to another aspect of the present invention, a data receiving step for receiving encoded audio data encoded using an audio codec, and detecting an error in the received encoded audio data An error detecting step, a data storing step for storing the encoded audio data in the reproduction buffer when no error is detected, a decoding step for decoding the encoded audio data in the reproduction buffer, and the audio decoded by the decoding step An audio reproduction method including an attenuation step capable of attenuating data and an audio reproduction step for reproducing audio data. In the data storage step, when an error is detected in the error detection step, the data is stored in a reproduction buffer. The previous value of the encoded audio data is maintained, and in the attenuation step, the error detection step Error is detected, when the previous value of the encoded voice data is maintained in the reproduction buffer, decoded by the decoding step, the audio data based on the previous value, that attenuates at a fixed maximum attenuation rate A featured audio playback method is provided.

  The components corresponding to the technical idea of the above-described sound reproduction device and the description thereof can be applied to the sound reproduction method.

  As described above, according to the present invention, it is possible to reliably prevent annoying sound reproduction when an error occurs, particularly when the error is for a long period of time.

  Exemplary embodiments of the present invention will be described below in detail with reference to the accompanying drawings. In addition, in this specification and drawing, about the component which has the substantially same function structure, duplication description is abbreviate | omitted by attaching | subjecting the same code | symbol.

  For example, 32 kbps ADPCM is used as a voice codec for digital data in communication between a wireless terminal device such as a PHS terminal or a mobile phone and a base station. In addition to the 32 kbps ADPCM, a parallel use of 16 kbps ADPCM is being studied from the viewpoint of effective use of the wireless line. Due to the characteristics of such ADPCM, abnormal noise may occur when an error occurs, and unpleasant sound may be heard.

(Wireless communication system 100)
In the present embodiment, there is provided an audio reproduction device capable of reliably preventing reproduction of unpleasant (unpleasant) audio at the time of an error even in such a 16 kbps ADPCM. In the present embodiment, the wireless terminal device 110 is used as an audio playback device. Hereinafter, an outline of the wireless communication system 100 that performs wireless communication by the wireless terminal device (sound reproduction device) 110 of the present embodiment will be described, and then each component will be described in detail.

  FIG. 1 is a block diagram illustrating a schematic configuration of a wireless communication system 100. The wireless communication system 100 includes a wireless terminal device 110, a base station 120, a communication network 130 such as the Internet or a dedicated line, a relay server 140, and another wireless terminal device 150.

  In the wireless communication system 100, when attempting to make a call to another wireless terminal device 150 using the wireless terminal device 110, the user operates his / her wireless terminal device 110 to establish a base station in the wireless communicable area. Establish wireless communication with the station 120, and communicate via the communication network 130, the relay server (including the base station control device and the exchange) 140, and the base station 120 in the wireless communication area of the other wireless terminal device 150. A voice call is performed with another wireless terminal device 150 of the other party.

(Wireless terminal apparatus 110)
FIG. 2 is a functional block diagram illustrating a hardware configuration of the wireless terminal device 110. The wireless terminal device 110 includes various electronic devices such as a laptop personal computer and a PDA (Personal Digital Assistant) in addition to the above-described mobile phone and PHS, and includes a terminal control unit 210 and a terminal memory 212 as constituent elements. A display unit 214, an operation unit 216, a voice input / output unit 218, and a terminal radio unit 220.

  The terminal control unit 210 manages and controls the entire wireless terminal device 110 by a semiconductor integrated circuit including a central processing unit (CPU). The terminal control unit 210 naturally performs a call function and a mail distribution function using the wireless terminal device 110 using the program in the terminal memory 212, but an error detection unit 254, data, which will be described later with reference to FIG. It also functions as a storage unit 256 and a decoding unit 260.

The terminal memory 212 is composed of ROM, RAM, E ² PROM, nonvolatile RAM, flash memory, HDD (Hard Disk Drive), etc., and stores programs processed by the terminal control unit 210, audio data, and the like. It can also function as a reception buffer 252 and a reproduction buffer 258 described later.

  The display unit 214 is composed of a color or monochrome display, and is stored in the terminal memory 212 or provided from an application relay server (not shown) via the communication network 130, and is a GUI (Web GUI or application GUI). Graphical User Interface) can be displayed.

  The operation unit 216 includes switches such as a keyboard, a cross key, and a joystick, and receives a user operation input.

  The voice input / output unit 218 includes a microphone and a speaker, converts the user's voice input during a call into voice data, and converts the voice data of the other party into voice. Further, a ring tone, an operation sound by the operation unit 216, an alarm sound, and the like can be output. It also functions as an audio playback unit 264 described later.

  The terminal radio unit 220 performs radio communication with the base station 120 in the mobile phone network. It also functions as a data receiving unit 250 described later.

  The hardware of the wireless terminal device 110 has been described above with reference to FIG. 2. The functions and operations performed on such hardware will be described in detail below.

  FIG. 3 is a functional block diagram showing a schematic function of the wireless terminal device 110. The wireless terminal device 110 includes a data reception unit 250, a reception buffer 252, an error detection unit 254, a data storage unit 256, a reproduction buffer 258, a decoding unit 260, an attenuation unit 262, and an audio reproduction unit 264. Consists of including.

  The data receiving unit 250 receives encoded audio data encoded using an audio codec, specifically, 16 kbps ADPCM, and stores it in a reception buffer.

  The reception buffer 252 temporarily holds the encoded audio data received by the data receiving unit 250.

  The error detection unit 254 detects an error in the encoded audio data received by the data reception unit 250. The encoded voice data is managed in a so-called frame in which packets of a predetermined length are combined every 5 msec. As errors detected in the present embodiment, whether or not the CI information indicating the frame type indicates voice data (TCH), CRC (Cyclic Redundancy Check), the presence or absence of a synchronization code (UW), etc. Is used. In addition to the errors listed here, various error detection methods that can determine the validity of the frame can be applied.

  The data storage unit 256 stores the encoded audio data of the reception buffer 252 in the reproduction buffer 258 when the error detection unit 254 detects no error. At this time, the encoded sound data can be subjected to sound quality improvement processing and stored in the reproduction buffer 258.

  The data storage unit 256 may maintain the previous value of the encoded audio data stored in the reproduction buffer 258 when the error detection unit 254 detects an error.

  The ADPCM is a technique for encoding a difference between current audio data and previous audio data. Therefore, when an error is detected and the encoded audio data cannot be used, it is estimated that the audio data is continuously changing, and the previous value is maintained without rewriting the encoded audio data. Under such a configuration, if the error is recovered before the predetermined number of times is reached, the audio data can be recovered quickly and smoothly.

  Further, the data storage unit 256 may store, in the reproduction buffer 258, encoded audio data that becomes silence or arbitrary audio when decoded, when the error detection unit 254 detects an error continuously for a predetermined number of times or more. The encoded voice data that becomes silent when decoded can be obtained, for example, by back-calculating (encoding) difference data for transition from already decoded voice data to silent voice data.

  As described above, when the error detection unit 254 detects an error, the data storage unit 256 maintains the previous value of the encoded audio data stored in the reproduction buffer 258 so that the audio data can be recovered quickly and smoothly. Prepare. However, if the error continues, there is a high possibility that the actual voice data and the estimated voice data are separated. Therefore, if the previous value is maintained, the voice data will expand in one direction, which will be harmful. There is also. Therefore, if the error is detected continuously for a predetermined number of times or more, the encoded audio data is stored in the reproduction buffer 258 so that the audio data becomes silent or arbitrary audio, thereby reliably preventing annoying audio reproduction. be able to.

  The reproduction buffer 258 temporarily holds the encoded audio data stored by the data storage unit 256.

  The decoding unit 260 decodes the encoded audio data stored in the reproduction buffer 258 into analog audio data.

  The attenuation unit 262 can attenuate the audio data decoded by the decoding unit 260. In the present embodiment, in particular, the audio data is attenuated for a predetermined time during the error and after the error is recovered. The attenuation unit 262 can be configured by software in the terminal control unit 210 or can be configured as hardware.

  Since the audio codec encodes the difference from the previous value of the audio data, it takes time to recover the original normal data once the audio data becomes an incorrect value. In particular, in the 16 kbps ADPCM, since the difference information amount is small compared to the 32 kbps ADPCM, it takes a long time to recover the audio data. With such a configuration in which the audio data is attenuated for a predetermined time, even if abnormal noise occurs in the recovery stage of the audio data, the audio is not affected, and an unpleasant audio reproduction can be prevented.

  In addition, the count starts after the error is recovered for the predetermined time, and the count value is reset at each error, so even if multiple errors occur consecutively at intervals, the last error A predetermined time is secured after the recovery of the noise, and the noise on the characteristics of the voice codec can be surely suppressed by the attenuation unit.

  The attenuation unit 262 may gradually decrease the attenuation rate so that the attenuation rate becomes 0 (zero) a predetermined time after the error is recovered.

  Here, after the error is recovered, the maximum attenuation rate is maintained before the predetermined time, and thereafter, the attenuation rate is gradually decreased to gradually increase the sound pressure of the audio data. The attenuation rate is changed so as to be 0 (zero). In the 16 kbps ADPCM, there is no effect of recovering the gain with an appropriate slope unlike the window function of 32 kbps ADPCM, and thus the attenuation rate is intentionally adjusted in this way. With the configuration in which the attenuation rate is gradually reduced, the reproduction state can be smoothly recovered without a sense of incongruity.

  The audio reproduction unit 264 reproduces audio data that has passed through the attenuation unit 262.

  With the configuration as described above, the wireless terminal device 110 can reproduce the encoded audio data encoded using the audio codec while avoiding annoying audio at the time of an error.

(Audio playback method)
Next, an audio reproduction method for reproducing encoded audio data using the wireless terminal device 110 as the audio reproduction device described above will be described.

  FIG. 4 is a flowchart showing the overall flow of the audio reproducing method. The wireless terminal device 110 initializes the normal counter value and the abnormal counter value to 0 for subsequent processing, and maintains the voice data standby state. Then, in response to a reception interrupt by wireless communication, the encoded speech data is received (data reception step: S300), and the received encoded speech data is stored in the reception buffer 252 (S302).

  Then, the error detection unit 254 determines the validity of the encoded audio data stored in the reception buffer 252. Here, the received encoded audio data packet is concatenated in units of frames, and an error is detected for each frame (error detection step). Specifically, it is determined whether or not the CI information indicating the frame type indicates voice data (TCH) (S304), and if the frame is voice data, the presence or absence of a bit error is determined by CRC ( S306). Here, it can also be determined whether or not the frame format, sampling frequency, transmission rate, etc. show reasonable numerical values. If no bit error is detected even in the CRC, it is finally determined whether or not there is a synchronization code UW that is represented by a specific bit string and indicates the position of the encoded audio data (S308).

  If no error is detected by such an error detection step, the received encoded audio data is regarded as normal and normal reception processing (S310) is performed. If an error is detected in any of the error detection steps, the encoded voice data is regarded as illegal data, and an abnormal reception process (S312) is performed. Then, the audio data stored in the reproduction buffer by each reception process is attenuated by the attenuation unit 262 and reproduced by the audio reproduction unit (S314). Hereinafter, the normal reception process (S310) and the abnormal reception process (S312) will be described in detail.

(Normal reception processing S310)
FIG. 5 is a flowchart showing a specific flow of normal reception processing S310. In the normal reception process S310, first, the data storage unit 256 transfers the encoded audio data regarded as normal data from the reception buffer 252 to the reproduction buffer 258 (S330), and sets the abnormal counter value used in the abnormal reception process S312. It is reset to 0 (S332). Then, the decoding unit 260 decodes the encoded audio data stored in the reproduction buffer 258 (S334).

  In addition, the attenuation unit 262 adjusts the attenuation rate of the audio data in order not to convert abnormal noise in the audio data recovery stage into audio. This adjustment maintains a maximum attenuation rate (in this case, -15 dB) for a sufficient time until the audio data is stabilized (until the normal counter value reaches the silence threshold), and thereafter gradually reduces the attenuation rate to cause an error. This is executed by changing the attenuation rate so that the attenuation rate becomes 0 after a predetermined time from the recovery of (when the normal counter value reaches the attenuation threshold).

  Therefore, the attenuation unit 262 compares the normal counter value with the silence threshold (here, “6”) (S336), and sets the attenuation rate to the maximum −15 dB if the normal counter value does not reach the silence threshold ( In step S338, the normal counter value is incremented (S340). If the silence threshold is reached, it is compared with the attenuation threshold (here, “10”) (S342). If the attenuation value is not reached, the difference between the attenuation value and the normal counter value is multiplied by −3 dB. The rate is set (S344), and the normal counter value is incremented (S346). By doing so, it is possible to adjust the attenuation rate to recover by -3 dB. When the normal counter value reaches the attenuation threshold, thereafter, 0 dB is set as the attenuation rate until an error occurs (S348), and the sound is not suppressed.

  Through such normal reception processing S310, it is possible to reproduce the received encoded audio data as audio and to suppress abnormal noise that still occurs even when the error is recovered.

(Abnormal reception process S312)
FIG. 6 is a flowchart showing a specific flow of the abnormal reception process S312. In the abnormal reception process S312, in order to quickly and smoothly recover the audio data when the error is recovered, the data storage unit 256, when the error detection unit 254 detects an error, stores the encoding stored in the reproduction buffer 258. When the previous value of the audio data is maintained and the error detection unit 254 detects errors continuously for a predetermined number of times (holding threshold) or more, the encoded audio data that becomes silence or arbitrary audio when decoded is stored in the reproduction buffer 258. .

  Therefore, in the abnormal reception process S312, first, the abnormal counter value is compared with the holding threshold (here, “4”) (S360). If the abnormal counter value does not reach the holding threshold, it remains in the reproduction buffer 258. The previous encoded audio data is maintained as it is (S362), and the abnormal counter value is incremented (S364). If the holding threshold is reached, silence data that becomes silence when decoded is generated and stored in the reproduction buffer 258 (S366).

  Here, the configuration is described in which the encoded audio data is maintained by not overwriting the value of the reproduction buffer 258 until the abnormal counter value reaches the holding threshold. However, the present invention is not limited to this, and the previous value is stored in another storage unit. It is also possible to adopt a configuration in which it is held in the memory and referred to each time.

  When the encoded audio data in the reproduction buffer 258 is determined, the decoding unit 260 decodes the encoded audio data stored in the reproduction buffer 258 (S368), and the attenuation rate is fixed to −15 dB in order to suppress audio reproduction ( S370). Then, the normal counter value used in the normal reception process S310 is reset to 0 (S372).

  Through such an abnormal reception process S312, the voice at the time of error is suppressed and the voice data is recovered quickly and smoothly when the error is recovered.

  Next, transition of data and attenuation rate in the reproduction buffer 258 by the above-described audio reproduction method will be described. Here, since the reception interruption of the above-described audio reproduction method is performed every 5 msec, the sampling time is 5 msec.

  FIG. 7 is an explanatory diagram showing the transition of data in the reproduction buffer 258 at the time of an error. On the left side of FIG. 7, encoded audio data received by the data receiving unit 250 and stored in the reception buffer 252 is shown. Then, the error judgment of the error detection unit 254 (“◯” when normal, “×” when abnormal) is shown, and the contents of the reproduction buffer 258 based on the judgment are shown on the right side of FIG.

  Referring to FIG. 7, when an error is detected, the previously sampled encoded audio data is maintained. For example, when the sampling number is 3, “reception data 2” used in the sampling number 2 is maintained. Further, even if the sampling number is 5, the “reception data 4” used in the sampling number 4 is maintained, and “reception data 4” is maintained while errors continue.

  When an error is continuously detected a predetermined number of times (four times), the reproduction buffer 258 has silence data for 16 kbps ADPCM that is decoded until the error is no longer detected, that is, until the error is recovered. Is forcibly written.

  Since the error is recovered at the sampling number 12, the transfer from the reception buffer 252 to the reproduction buffer 258, which is normal processing, is performed.

  FIG. 8 is an explanatory diagram showing the transition of the attenuation rate at the time of error. In FIG. 8, the vertical axis represents the attenuation rate of the attenuation unit 262, and the horizontal axis represents the number of samplings (time). In addition, “◯” and “×” shown on the horizontal axis indicate error determination results (“◯” when normal, “×” when abnormal). For example, when an error is detected in the third sampling, the abnormal reception process S312 is started, the attenuation rate becomes -15 dB, and the voice data is narrowed down so that the voice is cut off.

  Although the error is recovered by the fourth sampling and the normal counter value is counted 410, the normal counter value is reset because the error is detected again by the fifth sampling. Then, counting is again performed 412 from the 12th sampling in which the error is recovered. Here, until the normal counter value reaches the silence threshold, the attenuation rate is maintained at -15 dB.

  Then, when the normal counter value reaches the silence threshold during the 18th sampling without detecting an error again, the attenuation rate is decreased for each sampling, the sound pressure is increased little by little, and finally the actual audio data indicates Playback with sound pressure. The period up to the silence threshold can be arbitrarily set according to the implementation status, but it is preferable to select a time that is long enough to stabilize the audio data and short enough for the user not to feel uncomfortable.

  As mentioned above, although preferred embodiment of this invention was described referring an accompanying drawing, it cannot be overemphasized that this invention is not limited to the example which concerns. It will be apparent to those skilled in the art that various changes and modifications can be made within the scope of the claims, and these are naturally within the technical scope of the present invention. Understood.

  Note that the steps in the transmission quality compensation method of the present specification do not necessarily have to be processed in time series in the order described in the flowchart, but are performed in parallel or individually (for example, parallel processing or object processing). May be included.

  The present invention can be used for an audio reproducing apparatus and an audio reproducing method for reproducing encoded audio data encoded using an audio codec.

1 is a block diagram showing a schematic configuration of a wireless communication system. It is the functional block diagram which showed the hardware constitutions of the radio | wireless terminal apparatus. It is the functional block diagram which showed the schematic function of the radio | wireless terminal apparatus. It is the flowchart which showed the whole flow of the audio | voice reproduction | regeneration method. It is the flowchart which showed the specific flow of the normal reception process. It is the flowchart which showed the specific flow of the abnormal reception process. It is explanatory drawing which showed transition of the data in the reproduction | regeneration buffer at the time of an error. It is explanatory drawing which showed transition of the attenuation factor at the time of an error. It is a block diagram for demonstrating an audio codec.

Explanation of symbols

110 Wireless terminal device (voice playback device)
220 Terminal Radio Unit 250 Data Receiving Unit 254 Error Detection Unit 256 Data Storage Unit 258 Playback Buffer 260 Decoding Unit 262 Attenuation Unit 264 Audio Playback Unit

Claims (4)

A data receiving unit that receives encoded audio data encoded using an audio codec, an error detection unit that detects an error in the received encoded audio data, and the error detection unit did not detect an error A data storage unit for storing the encoded audio data in the reproduction buffer, a decoding unit for decoding the encoded audio data stored in the reproduction buffer, and attenuating the audio data decoded by the decoding unit. An audio reproduction device comprising a possible attenuation unit and an audio reproduction unit for reproducing the audio data,
When the error detection unit detects an error, the data storage unit maintains the previous value of the encoded audio data stored in the reproduction buffer,
The attenuation unit detects audio data based on the previous value decoded by the decoding unit when an error is detected by the error detection unit and the previous value of the encoded audio data is maintained in the reproduction buffer. Attenuate at a fixed maximum attenuation rate ,
The sound reproducing apparatus according to claim 1, wherein the attenuation unit gradually reduces the attenuation rate so that the attenuation rate becomes zero after a predetermined time after the error is recovered .   2. The audio reproduction device according to claim 1, wherein the attenuation unit is provided before the audio reproduction unit, and attenuates the audio data for a predetermined time after the error is recovered. 3. The audio reproducing apparatus according to claim 1, wherein the audio codec is 16 kbps ADPCM (adaptive differential PCM). A data receiving step for receiving encoded audio data encoded using an audio codec;
An error detection step of detecting an error in the received encoded audio data;
A data storage step of storing the encoded audio data in a reproduction buffer when the error is not detected;
A decoding step of decoding the encoded audio data of the reproduction buffer;
An attenuation step capable of attenuating the audio data decoded by the decoding step;
An audio reproduction step of reproducing the audio data;
An audio playback method including:
In the data storage step, when an error is detected in the error detection step, the previous value of the encoded audio data stored in the reproduction buffer is maintained,
In the attenuation step, when an error is detected by the error detection step and the previous value of the encoded audio data is maintained in the reproduction buffer, the audio data based on the previous value decoded by the decoding step is Attenuate at a fixed maximum attenuation rate ,
In the attenuation step, the attenuation rate is gradually reduced so that the attenuation rate becomes zero a predetermined time after the error is recovered .

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