JP2007140063A - Device for sound recording and reproducing - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a device for sound recording and reproducing device which can optimize the sound recording quality according to an object of sound recording, is easy to use and is free of misoperations. <P>SOLUTION: The sound recording and reproducing device comprises a sound signal level detection unit 1, which compares an input sound signal with a specified criterion to detect a sound signal not lower than the criterion; a sound-recorded object decision unit 2 which decides the object of sound recording, based on the frequency of input of the sound signal that is not lower than the acceptance criterion detected by the sound signal level detection unit 1; a sampling frequency selection unit 3 which selects a sampling frequency, based on the decision result of the sound-recorded object decision unit 2 so as to optimize the sound recording quality; and an encoding unit 4 which encodes the input sound signal into sound data of a bit rate, corresponding to the sampling frequency selected by the sampling frequency selection unit 3. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to an audio recording / reproducing apparatus having a recording target discrimination function.

  After converting an analog audio signal into digital audio data, the audio data is recorded in a flash memory as a rewritable recording medium, and the audio data recorded in the flash memory is converted into analog and played back. A portable audio recording / reproducing apparatus (hereinafter referred to as an IC recorder) has been put into practical use since the late 1990s. Furthermore, in recent years, in addition to the above functions, as a music player that can reproduce audio files in WMA (Windows (registered trademark) Media Audio) format and MP3 (MPEG-1 audio layer3) format with high sound quality. IC recorders have also been commercialized.

  Such an IC recorder can selectively execute various operations such as recording, reproduction, fast forward, and fast reverse by operating a predetermined operation unit. For example, during a recording operation, audio data converted into a digital signal is recorded in an audio data recording area provided in advance in a flash memory, and index information (audio data recording) about the audio data is recorded in the index information recording area. Various information regarding audio data such as address and recording date / time) is recorded.

  Here, the audio data recorded in the flash memory constitutes an audio file together with the index information. An audio file is assigned a file number (file name) and a folder symbol at the time of recording so as to facilitate subsequent file search and reproduction. In addition, a plurality of folders such as A, B, and C folders are often provided so that audio files that continue to increase every recording can be efficiently managed. Further, when selecting a folder, the folder mark is pop-displayed on a display device such as an LCD provided in the IC recorder body so that the operator can easily recognize the folder being selected from among the plurality of folders provided. The device is made.

  In this way, the IC recorder has evolved sufficiently in function, performance, and operability, but it can be recorded according to the recording target, as represented by dictation and conference (conversations, lectures, etc.). The technology for optimizing quality still has to rely on the operator's own experience and intuition.

As a technique for solving such a problem, for example, in a recording apparatus proposed in Patent Document 1, switching selection of recording time and recording quality corresponding to a recording target is performed. Specifically, according to the recording time or sound quality designated by the operator, a combination of sampling frequency and compression rate that can be recorded with the highest sound quality is selected for each recording target.
JP-A-8-185671

  Here, in the case of the proposal of the above-mentioned patent document 1, it is necessary to designate the recording time and the sound quality each time by the operator's own situation judgment before starting the recording. Therefore, there is a possibility that the designated operation will not be in time when recording a song. In addition, inexperienced operators who have little experience in recording often cannot imagine the relationship between the recording target and the expected recording quality (recording time, sound quality, etc.), and in this sense, the operator himself requires complicated operations. Is not realistic.

  The present invention has been made in view of the above circumstances, and an object of the present invention is to provide an audio recording / reproducing apparatus that can optimize recording quality according to a recording target and that is easy to use and has no erroneous operation.

  In order to achieve the above object, an audio recording / reproducing apparatus according to the first aspect of the present invention includes an audio signal level detecting means for comparing the input audio signal with a predetermined criterion value within a predetermined time. Recording target determination means for determining a recording target based on the number of times of input of an audio signal equal to or higher than the determination reference value detected by the voice signal level detection means, and the recording quality is optimal based on the determination result by the recording target determination means Sampling frequency selection means for selecting the sampling frequency so as to be, and encoding means for encoding the input audio signal into the audio data of the bit rate corresponding to the sampling frequency selected by the sampling frequency selection means It is characterized by comprising.

  According to the present invention, it is possible to provide an audio recording / reproducing apparatus that can optimize recording quality according to a recording target and that is easy to use and has no erroneous operation.

Hereinafter, embodiments of the present invention will be described with reference to the drawings.
FIG. 1 is a block diagram showing the main configuration of an audio recording / reproducing apparatus according to an embodiment of the present invention. As shown in FIG. 1, the audio recording / playback apparatus according to the present embodiment includes an audio signal level detection unit 1, a recording target determination unit 2, a sampling frequency selection unit 3, and an encoding unit 4. .

  The audio signal level detection unit 1 compares the audio signal input through a microphone (not shown) and amplified by a microphone amplifier (not shown) with a predetermined determination reference value, and as a result of this comparison, the audio signal level detection unit 1 is equal to or greater than the predetermined determination reference value. Is output to the recording target determination unit 2. The recording target determination unit 2 counts the number of input audio signals that are equal to or greater than the determination reference value input from the audio signal level detection unit 1 within a predetermined time, and determines the recording target based on the number of times. Then, the determination result is output to the sampling frequency selection unit 3. The sampling frequency selection unit 3 selects a sampling frequency based on the determination result of the recording target by the recording target determination unit 2 so as to optimize the recording quality, and supplies the selected sampling frequency to the encoding unit 4. The encoding unit 4 encodes the audio signal into audio data having a bit rate corresponding to the sampling frequency supplied from the sampling frequency selection unit 3.

  FIG. 2 is a block diagram showing an internal configuration of an IC recorder as an example of an audio recording / reproducing apparatus according to an embodiment of the present invention. The IC recorder shown in FIG. 2 includes a microphone (MIC) 20 that converts sound into an electrical signal. At the time of voice recording, the microphone (MIC) 20 converts the voice into an electrical signal (analog voice signal), and outputs the voice signal obtained by the conversion to the microphone amplifier (AMP) 21 at the subsequent stage. The microphone amplifier (AMP) 21 amplifies the audio signal input via the microphone (MIC) 20 and outputs the amplified audio signal to the low-pass filter (LPF) 22. The low-pass filter (LPF) 22 cuts an unnecessary frequency band from the input audio signal and outputs the cut signal to the A / D converter (ADC) 23. The A / D converter (ADC) 23 converts the input audio signal into a digital signal, and outputs the digital audio signal obtained thereby to the digital signal processing unit (DSP) 24. During recording, the digital signal processing unit (DSP) 24 reads the digital audio signal obtained by the A / D converter (ADC) 23 in units of frames and performs predetermined encoding under the control of the system control unit 25. Encode (compress) audio data in the format. The encoded audio data is temporarily stored in a buffer memory (not shown) provided in the system control unit 25.

  On the other hand, the digital signal processing unit (DSP) 24 reads the encoded audio data in units of frames from a buffer memory (not shown) of the system control unit 25 and decodes it under the control of the system control unit 25 during reproduction of the audio. (Expand) and output to the D / A converter (DAC) 26. The D / A converter (DAC) 26 converts the encoded audio data into an analog signal and outputs the analog signal to the low-pass filter (LPF) 27. The low-pass filter (LPF) 27 cuts an unnecessary frequency band from the input audio signal and outputs it to the power amplifier (AMP) 28. The power amplifier (AMP) 28 amplifies the input audio signal and outputs it to the speaker (SP) 29. The speaker (SP) 29 outputs the input audio signal as audio.

  The system control unit 25 includes, for example, a CPU, and a recording unit (memory) 30, an operation unit 31, and a display unit 32 are connected in addition to the digital signal processing unit (DSP) 24.

  The recording unit (memory) 30 is configured by a nonvolatile semiconductor memory such as a flash memory, for example. During recording, audio data encoded by the digital signal processing unit (DSP) 24 is recorded in the recording unit (memory) 30 through a buffer memory (not shown) under the control of the system control unit 25. . At this time, index information related to the audio data is also recorded.

  The operation unit 31 includes a recording switch (REC), a playback switch (PLAY), a stop switch (STOP), a fast forward switch (FF), a fast reverse switch (REW), a menu switch (MENU), an erase switch (ERASE), and a hold switch. Various operation members such as (HOLD) are provided for each function. In response to such operation of the operation unit 31, the system control unit 25 executes various operation processes.

  When a predetermined process is started in response to any operation included in the operation unit 31, the display unit 32 displays an operation mode of the IC recorder or a subsequent operation state in the operation mode. For example, when the recording switch (REC) is pressed by the operator, the display unit 32 displays the elapsed recording time, the remaining recording time, the file number, and the like. In addition, when the menu switch (MENU) is pressed by the operator, the display unit 32 selects the microphone sensitivity (high / low), the recording mode (automatic / standard / long), and the alarm selection (on). Display related to the function selection of this IC recorder such as / off). Further, when the system control unit 25 has a clock function, the display unit 32 also displays a calendar of the current date and time.

  Next, the main operation of the IC recorder having the configuration shown in FIG. 2 will be described with reference to FIG. FIG. 3 is a flowchart showing the main operation of the IC recorder of FIG.

  When the IC recorder is powered on by operating a power switch (not shown) or the like, as shown in FIG. 3, the system control unit 25 performs predetermined initial settings (S1). After the end of the initial setting, the system control unit 25 starts counting a timer (not shown) (S2). This timer is a timer for measuring the time for the IC recorder to enter the standby mode (low current consumption mode) from the normal operation mode after a predetermined time has elapsed.

  After the timer start in S2, the system control unit 25 performs a switch detection process shown in S3 to S9. That is, the system controller 25 determines whether the recording switch (REC) is turned on (S3), whether the playback switch is turned on (S4), whether the fast-forward switch (FF) is turned on (S5), Whether the rewind switch (REW) is turned on (S6), whether the stop switch (STOP) is turned on (S7), whether the menu switch (MENU) is turned on (S8), and the erase switch ( It is sequentially determined whether or not (ERASE) is turned on (S9).

  If it is determined in S3 to S9 that all the switches are off, the process proceeds to S10, and the system control unit 25 determines whether or not the time counted by the timer has exceeded a predetermined time (S10). If it is determined in S10 that the time counted by the timer does not exceed the predetermined time, the process returns to S3, and the switch detection process in S3 to S9 is continued. On the other hand, if it is determined in S10 that the time counted by the timer exceeds the predetermined time, S10 is branched to S11, and the system control unit 25 performs a standby mode subroutine process (S11). When the standby mode is entered, the IC recorder operates at a low current. Specifically, the microphone (MIC) 20, microphone amplifier (AMP) 21, low-pass filter (LPF) 22, A / D converter (ADC) 23, digital signal processor (DSP) 24, D / A in FIG. The power supply to the converter (DAC) 26, the low-pass filter (LPF) 27, the power amplifier (AMP) 28, the recording unit (memory) 30, and the display unit 32 is cut off, or each block of FIG. By outputting a non-selection signal from the system control unit 25 to a chip enable terminal provided in the IC, a low current consumption state is obtained. At this time, the CPU of the system control unit 25 itself switches to the low-speed clock with the least current consumption by switching the operation clock to the low current consumption state. In some cases, the operation clock may be switched from a main clock (for example, 16.384 MHz) to a sub clock (for example, 32.768 kHz), and then the main clock may be completely stopped until a switch input is detected.

  Here, if any switch operation is performed by the user during the standby mode, the process returns to S2 and the processing of the main operation is resumed.

  If it is determined in S3 that the recording switch is turned on, S3 is branched to S12, and the system control unit 25 executes a recording processing subroutine (S12). This recording process will be described later. If it is determined in S4 that the regeneration switch is turned on, S4 is branched to S13, and the system control unit 25 executes a subroutine for the regeneration process (S13). If it is determined in step S5 that the fast forward switch is turned on, step S5 is branched to step S14, and the system control unit 25 executes a fast forward processing subroutine (S14). If it is determined in S6 that the fast reverse switch is turned on, S6 is branched to S15, and the system control unit 25 executes a subroutine of the fast reverse process (S15). If it is determined in S7 that the stop switch is turned on, S7 is branched to S16, and the system control unit 25 executes a subroutine for stop processing (S16). If it is determined in S8 that the menu switch is turned on, S8 is branched to S17, and the system control unit 25 executes a menu change processing subroutine (S17). If it is determined in S9 that the erasure switch is turned on, S9 is branched to S18, and the system control unit 25 executes a subroutine for erasure processing (S18). After executing any of these subroutines, the system control unit 25 restarts the timer (S19) and returns to S3.

  Here, each of the reproduction process of S13, the fast-forward process of S14, the fast-rewind process of S15, the stop process of S16, the menu change process of S17, and the erase process of S18 can be performed using known technical means. In addition, since it is not directly related to the contents of the present embodiment, detailed description thereof is omitted here.

  Next, the recording process of the audio recording / reproducing apparatus of this embodiment will be further described. FIG. 4 is a diagram showing a detailed configuration related to the recording process among the configurations of FIG.

  The audio signal (VIN) output from the microphone (MIC) 20 in FIG. 2 corresponds to the amplifier circuit (the microphone amplifier (AMP) 21) configured by the operational amplifier 40, capacitors C1 and C2, and resistors R1 and R2. ) 21. In the amplifier circuit 21 of FIG. 4, the input audio signal (VIN) is amplified approximately R2 / R1 times, and the audio signal (VLO) obtained thereby is output to the low-pass filter (LPF) 22.

  Here, a negative feedback circuit is formed between the input and output of the amplifier circuit 21 so that the amplitude of the audio signal (VLO) obtained by amplification does not exceed the input dynamic range of the A / D converter 23 described later. An auto gain controller (AGC) 42 is added. The auto gain controller (AGC) 42 automatically suppresses excessive input of the audio signal to the A / D converter 23.

  The low-pass filter (LPF) 22 is an antialiasing filter that operates at a cut-off frequency that is half or less of the sampling frequency, for example, approximately 0.45 times the sampling frequency. The audio signal (VLO) output from the amplifier circuit 21 passes through a low-pass filter (LPF) 22, so that an audio signal in a high frequency band unnecessary for quantization is attenuated and removed. The audio signal in the frequency band lower than the cut-off frequency that has passed through the low-pass filter (LPF) 22 is quantized every sampling period (which is the reciprocal of the sampling frequency) in the A / D converter (ADC) 23 in the subsequent stage. (A / D conversion). The number of quantization bits at this time is at least 14 bits. The digital audio signal obtained by the A / D converter (ADC) 23 is output to the digital signal processing unit (DSP) 24 and the magnitude comparator 45.

  By the way, in the low pass filter (LPF) 22, the A / D converter (ADC) 23, and the digital signal processing unit (DSP) 24, either one of fs for high bit rate or fs for low bit rate is used as the sampling frequency. Only waves are supplied. That is, the sampling frequency corresponding to the bit rate is alternatively selected from the low frequency filter (LPF) 22 and the A / D converter (ADC) according to the fs selection signal input from the CPU 47 constituting the system control unit 25 to the selector 46. 23 and a digital signal processing unit (DSP) 24. That is, the CPU 47 and the selector 46 constitute the sampling frequency selection unit 3 in FIG. Here, it is assumed that the high bit rate fs has a higher frequency than the low bit rate fs.

  In the digital signal processing unit (DSP) 24 corresponding to the encoding unit 4 in FIG. 1, the digital audio signal is converted according to the combination of the audio data compression rate specified in advance by the CPU 47 and the sampling frequency selected by the selector 46. It is encoded into audio data of a predetermined bit rate.

  Of course, the sampling frequency that can be selected by the selector 46 is not limited to two waves, and may be selected from, for example, three or more fixed frequencies. Further, an arbitrary frequency may be supplied using a PLL instead of the selector 46. By increasing the number of sampling frequencies that can be supplied in this way, it is possible to cope with a wide selection of bit rates.

  Further, in the magnitude comparator 45 corresponding to the audio signal level detection unit 1 of FIG. 1, the digital audio signal output from the A / D converter (ADC) 23 is a predetermined upper threshold (+ VTH) and lower threshold (−VTH). A comparison is made at each sampling period. The detection result of the magnitude comparator 45 is output to the CPU 47 as an audio signal level detection signal.

  Specifically, it is symmetric with respect to the operation reference voltage (AVREF) of the operational amplifier 40 within the dynamic range (VCC> ADIN (MAX), ADIN (MIN)> GND) of the A / D converter 23 shown in FIG. The size of the audio signal (VLO) of the portion that vibrates is compared. In this size comparison, when the sound input to the microphone (MIC) 20 in FIG. 2 is loud, the sound signal (VLO) vibrates with an amplitude exceeding the upper threshold (+ VTH) and the lower threshold (−VTH). The magnitude comparator 45 outputs audio signal level detection signals indicating that ADIN (MAX)> VLO ≧ + VTH and −VTH ≧ VLO> ADIN (MIN). Conversely, when the sound input to the microphone (MIC) 20 of FIG. 2 is relatively small, the sound signal (VLO) vibrates with an amplitude within the range of the upper threshold value (+ VTH) and the lower threshold value (−VTH). The magnitude comparator 45 outputs an audio signal level detection signal indicating that + VTH> VLO> −VTH.

  Hereinafter, the recording process will be described with reference to FIG. FIG. 6 is a flowchart showing the recording process. The processing in FIG. 6 is performed by the CPU 47 in the system control unit 25. Here, the following description will be continued assuming that the system control unit 25 performs the processing.

  When the recording process starts, the system control unit 25 first sets the microphone sensitivity (high / low), the recording mode (automatic / standard / long), the file number, the recording start address in the audio data recording area for recording the audio data, and the like. Are recorded in the index information recording area (S30). Here, “automatic” is selected as the recording mode unless the operator intentionally selects “standard” or “long”.

  Next, the system control unit 25 clears the LBR flag to 0 (S31), and further clears the RJF flag to 0 (S32). Here, the LBR flag is a sign flag that is set when switching from a high bit rate fs to a low bit rate fs is performed in a recording target determination process described later. The RJF flag is a sign flag that is set when the recording target determination process is completed.

  After S32, the system control unit 25 outputs the fs selection signal so that the high bit rate fs is selected as the sampling frequency (S33). That is, when the recording mode is “automatic”, the high bit rate fs is always selected at the start of the recording operation. In the present embodiment, the recording mode “standard” corresponds to the high bit rate fs, and the “long” case corresponds to the low bit rate fs.

  After the above processing, the analog audio signal output from the microphone (MIC) 20 is quantized (A / D conversion) for each sampling period (S34). Thereafter, the system control unit 25 determines the state of the RJF flag (S35). If the RJF flag = 0 in the determination of S35, the process proceeds to the recording target determination sequence after S41. In the recording target determination sequence, first, the level of the quantized audio signal is detected every sampling period by the magnitude comparator 45 of FIG. 4 (S41). Here, the level detection of the audio signal in S41 may be performed for a very short predetermined time from the start of recording, for example, about 30 seconds from the start of recording. Next, the system control unit 25 determines whether or not the predetermined time has elapsed since the start of recording (S42). If it is determined at S42 that the predetermined time has not elapsed, the process returns to S34.

  On the other hand, when the predetermined time has passed in the determination of S42, S42 is branched to S43, and the system control unit 25 selects the recording target based on the audio signal level detection signal collected within the predetermined time of S42. Determine (S43). More specifically, the audio having a level equal to or higher than a predetermined level (for example, ADIN (MAX)> VLO ≧ + VTH and −VTH ≧ VLO> ADIN (MIN)) among the audio signal level detection signals collected within the predetermined time of S42. The recording target is determined by the number of detections of the signal level detection signal.

  Here, it is only necessary to roughly divide recording targets into “conference recording” and “dictation recording”. In general, in the case of “conference recording”, since the sound perceived by the microphone (MIC) 20 is weak and small, the number of detections of the audio signal level detection signal above a predetermined level is reduced. On the other hand, in the case of “dictation recording”, since the sound perceived by the microphone (MIC) 20 is strong and loud, the number of detections of the audio signal level detection signal equal to or higher than a predetermined level is larger than that in “conference recording”. Therefore, in S43, in order to simplify the explanation, when the number of detections of the audio signal level detection signal equal to or higher than the predetermined level is equal to or higher than the predetermined number, the recording target is “oral recording”; Determines that the recording target is “conference recording”.

  Note that the number of detections of the audio signal level detection signal of the predetermined level or higher collected within the predetermined time of S42 is compared with the average value of the number of detections of the audio signal level detection signal of the predetermined level or higher actually measured for each recording target. The recording target may be determined by the above. In this case, the system control unit 25 detects the average number of detection times closest to the number of detections of the audio signal level detection signal of a predetermined level or higher collected within a predetermined time, and the recording corresponding to this detection number average value. Determine the target as the current recording target. Here, the average number of detection times of the audio signal level detection signal of a predetermined level or more actually measured for each recording target may be obtained by actual measurement such as a field test, and a program (not shown) provided in the CPU 47 in the system control unit 25. It can be stored in memory as a default value.

  Furthermore, if you want to improve the accuracy of judgment of the recording target, set multiple thresholds for the detection level of the audio signal, or perform statistical arithmetic processing on the detection interval and the number of detections of the audio signal, and correlate with the model recording target. The recording target may be inferred from the gender.

  If it is determined in S43 that the number of detections of the audio signal level detection signal equal to or higher than the predetermined level is not equal to or higher than the predetermined number, S43 is branched to S44, and the system control unit 25 determines that the recording target is “conference recording”. judge. In the case of “conference recording”, since the sound perceived by the microphone (MIC) 20 is weak and small, a high bit rate is used as a sampling frequency so that the atmosphere in the recording environment and the superposition of voices of multiple speakers can be recorded clearly. An fs selection signal is output so that fs for use is selected (S44). Thereafter, the process proceeds to S47. On the other hand, if it is determined in S43 that the number of detections of the audio signal level detection signal equal to or higher than the predetermined level is equal to or higher than the predetermined number, S43 is branched to S45, and the system control unit 25 determines that the recording target is “dictation recording”. It is determined that In the case of “dictation recording”, since the sound perceived by the microphone (MIC) 20 is strong and loud, an fs selection signal is output so that the low bit rate fs is selected as the sampling frequency (S45). When the low bit rate fs is selected, the LBR flag is set to 1 (S46).

  After the above processing, the system control unit 25 sets the RJF flag to 1 (S47) and returns to S34.

  On the other hand, if it is determined in S35 that the RJF flag is 1, S35 is branched to S36, and encoding is performed by the digital signal processing unit (DSP) 24, and the audio data obtained thereby is recorded in the recording unit ( (Memory) 30 (S36). Here, encoding itself is continuously performed in units of frames until a predetermined number of frames of audio data is reached. That is, when the audio data stored in the buffer memory (not shown) of the CPU 47 reaches a predetermined number of frames, the predetermined number of frames of audio data are recorded in the audio data recording area of the recording unit (memory) 30. For example, when audio data obtained by encoding an audio signal of 14 bits / frame up to 4 bits / frame is written to the recording unit (memory) 30 in units of 512 bytes, the predetermined number of frames is 1024.

  Next, the system control unit 25 determines whether or not the stop switch is turned on (S37). In the determination of S37, the processes of S34 to S36 are repeated until the stop switch is turned on. On the other hand, if it is determined in S37 that the stop switch is turned on, S37 is branched to S38, and the system control unit 25 determines the state of the LBR flag (S38). In the determination of S38, when the LBR flag = 0, the system control unit 25 executes the recording end process as it is (S40) and ends the recording.

  On the other hand, in the determination at S38, when the LBR flag = 1, the audio data from the start of recording at the high bit rate fs to the elapse of the predetermined time in S42 is the audio data recorded after the elapse of the predetermined time. Is converted into low bit rate fs audio data equivalent to (S39). Thereafter, the system control unit 25 executes a recording end process (S40) and ends the recording. Here, for example, when the sampling frequency is fs = 16 kHz for high bit rate and fs = 8 kHz for low bit rate, if there is a margin in the memory capacity that can be used as a work area, a predetermined time elapses from the start of recording. The quantized digital audio signal may be temporarily stored in a buffer memory provided in the CPU 47 in FIG. 4 or a recording unit (memory) 30 in FIG. For example, if the predetermined time is 30 seconds, the memory capacity required for writing a 14-bit digital audio signal for 30 seconds is slightly (14 bits × 16 kHz × 30 seconds) ÷ 8 bits = 840 kbytes. If the capacity of the recording unit (memory) 30 is 1 Gbyte, the consumption of 840 kbytes is 0.1% or less, and recording of audio data (that is, recording time) is not significantly compressed.

  The conversion to low bit rate audio data is performed, for example, by sequentially reading the digital audio signals temporarily written in the work area of the recording unit (memory) 30 from the head address, and the digital signal processing unit (DSP) 24 reducing the audio data. After encoding corresponding to bit rate fs = 8 kHz, the converted audio data may be sequentially re-recorded in the audio data recording area of the corresponding file number in the recording unit (memory) 30. Of course, if the processing capability of the digital signal processing unit (DSP) 24 is high, the audio data to be converted is read directly from the audio data recording area of the recording unit (memory) 30 and the digital signal processing unit (DSP) 24 reduces the data. It may be converted into audio data corresponding to the bit rate fs and re-recorded in the recording unit (memory) 30. For example, when re-encoding audio data with a recording time of 30 seconds and a bit rate of 64 kbps recorded in MP3 format into audio data with a low bit rate of 32 kbps, if the above-described series of processing is completed within a few seconds, The operator does not feel stress.

  Here, the update of various index information such as the recording mode (standard / long) relating to the audio data and the recording start address in the audio data recording area is performed after the conversion to the low bit rate in S39.

  As described above, according to the present embodiment, the optimum bit rate for the recording target is automatically selected during recording, so that the operator can concentrate on the recording action. Further, since the recording quality is kept constant for each recording target, there is no waste in using the memory.

  Although the present invention has been described based on the above embodiments, the present invention is not limited to the above-described embodiments, and various modifications and applications are naturally possible within the scope of the gist of the present invention.

  Further, the above-described embodiments include various stages of the invention, and various inventions can be extracted by appropriately combining a plurality of disclosed constituent elements. For example, even if some constituent requirements are deleted from all the constituent requirements shown in the embodiment, the problem described in the column of the problem to be solved by the invention can be solved, and the effect described in the column of the effect of the invention Can be extracted as an invention.

It is the block diagram which showed the main structures of the audio | voice recording / reproducing apparatus concerning one Embodiment of this invention. It is a block diagram which shows the internal structure of the IC recorder as an example of the audio | voice recording / reproducing apparatus which concerns on one Embodiment of this invention. It is a flowchart which shows the main operation | movement of the IC recorder of FIG. It is a figure shown about the detailed structure which concerns on a sound recording process among the structures of FIG. It is a figure for demonstrating an audio | voice signal level detection signal. It is a flowchart shown about a recording process.

Explanation of symbols

  DESCRIPTION OF SYMBOLS 1 ... Audio | voice signal level detection part, 2 ... Recording object determination part, 3 ... Sampling frequency selection part, 4 ... Encoding part, 20 ... Microphone (MIC), 21 ... Microphone amplifier (AMP), 22, 27 ... Low pass filter ( LPF), 22 ... Digital signal processing unit, 23 ... A / D converter (ADC), 24 ... Digital signal processing unit (DSP), 25 ... System control unit, 26 ... D / A converter (DAC), 28 ... Power amplifier (AMP), 29 ... Speaker, 30 ... Recording unit (memory), 31 ... Operation unit, 32 ... Display unit, 40 ... Operational amplifier, 42 ... Auto gain controller (AGC), 45 ... Magnitude comparator, 46 ... Selector, 47 ... CPU

Claims (4)

An audio signal level detection means for comparing the input audio signal with a predetermined determination reference value and detecting an audio signal greater than the determination reference value;
A recording target determination means for determining a recording target based on the number of times of input of the audio signal equal to or higher than the determination reference value detected by the voice signal level detection means within a predetermined time;
Sampling frequency selecting means for selecting a sampling frequency so that the recording quality is optimized based on the determination result by the recording target determining means;
Encoding means for encoding the input audio signal into audio data having a bit rate corresponding to the sampling frequency selected by the sampling frequency selection means;
An audio recording / reproducing apparatus comprising:   The sampling frequency selection means selects a sampling frequency corresponding to a high sound quality bit rate at the start of a recording operation, and after the recording target is determined by the recording target determination means, the determination result by the recording target determination means The audio recording / reproducing apparatus according to claim 1, wherein a sampling frequency based on the selection is selected.   The encoding means encodes the input audio signal at a high sound quality bit rate at the start of a recording operation, and after the recording target is determined by the recording target determination means, the bit rate at the start of the recording operation When the recording target is different from the bit rate after the recording target is determined, the audio data encoded at the bit rate at the start of the recording operation is re-encoded at the bit rate after the recording target is determined. The audio recording / reproducing apparatus according to claim 1. The recording target includes at least either oral recording or conference recording,
The sampling frequency selection means selects a sampling frequency corresponding to a low sound quality bit rate when the recording target is determined to be oral recording by the recording target determination means, and the recording target determination means selects the recording target. 2. The audio recording / reproducing apparatus according to claim 1, wherein a sampling frequency corresponding to a high-quality bit rate is selected when it is determined that the target is conference recording.

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