JP4735311B2 - Audio processing device - Google Patents

Description

  The present invention relates to an audio processing apparatus for re-recording based on a compression-encoded audio signal received from a digital broadcast.

  2. Description of the Related Art Conventionally, there is known a data reproducing apparatus that reads a video signal and an audio signal recorded by compression encoding from a recording medium such as a CD-ROM, a DVD-ROM, a semiconductor memory, and a hard disk, and decodes (decompresses) the signal. Yes. Video and audio reproduction is realized by sending the decompressed data to a video monitor, an audio speaker, or the like connected to the apparatus. Digital broadcasts can also be reproduced by receiving compressed and encoded video signals and audio signals and decoding them in the same way as reproduction from a recording medium.

Many standards are defined for compression coding. For example, MPEG (Motion Picture Experts Group) is known as a standard for compressing and encoding both video signals and audio signals.

  A conventional system and method for receiving and playing back a digital broadcast will be described below. In addition, the applicant has recognized the digital broadcasting system disclosed in Patent Document 1 and Patent Document 2, and in order to clarify the problem of the present invention, the contents of the prior art are simply described as conventional examples. explain.

  First, a conventional digital broadcasting system in Patent Document 1 will be described. If a station that distributes digital broadcasts is equipped with facilities that broadcast the same content, and the digital broadcast receiver is unable to view the digital broadcast that is currently being viewed due to a decrease in the strength of the received radio wave, the same program Another digital broadcasting station is selected based on the affiliated station information broadcasting. As a result, if another broadcast station of the same series can be selected and received for a program that was being viewed before the intensity of the received radio wave decreased, the same program as before can be continuously viewed.

Next, a conventional digital broadcast signal transmission system in Patent Document 2 will be described. Extract the frame synchronization information from the first signal source received from the transmitting station, perform decoding based on this frame synchronization information, and re-encode and send this decoded signal based on the previous frame synchronization information To do. When switching from the first signal source to the second signal source, reset means for resetting the frame synchronization is provided, and switching is performed during the silence period of the audio signal. With these configurations, re-encoding can be performed in synchronization with the frame information of the original signal, so that the frame phase does not differ compared to the case where frame synchronization is not performed. Can do.
JP 2004-140690 A (page 4-8, FIG. 1) Japanese Unexamined Patent Publication No. 2003-209712 (page 3-6, FIGS. 1 and 3)

  In the system of Patent Document 1, which is a precedent example, when digital broadcasts have a means for quickly switching to broadcasts when the intensity of received radio waves has dropped due to sudden heavy rain or some kind of radio interference, It is possible. In order to record the audio reproduced from the digital broadcast on a recording medium such as a DVD, there are a case where the received data is recorded as it is and a case where the received data is re-encoded based on the received data. .

  Assuming recording of a broadcast program, it may be assumed that the radio wave is hard to reach when the intensity of the received radio wave of the received program decreases. In such a case, the intensity of received radio waves of other series of digital broadcasts may be lowered. In such a case, the program cannot be recorded by switching to the same program of the affiliated station.

  In general, when recording a digital broadcast program, if the intensity of the received radio wave continues to decrease, a decoded sound can be produced at all times, such as when a noise sound is generated due to a decoding error or when the decoded sound is reproduced intermittently. It falls into a state of disappearing. If these reproduced voices are re-encoded as they are, the portion recorded when the intensity of the received radio wave is reduced becomes an unpleasant voice full of voice noise. Therefore, if the re-encoding unit and the recording unit take measures such as detecting the intensity of the received radio wave and stopping the recording when the re-encoding unit and the recording unit decrease, it is possible to prevent recording of audio noise. However, in the case of a reservation for an absence recording program, the program recording is stopped due to a mistake in the operation setting such as a recording reservation by the user, a program recording failure due to an individual device failure, or a decrease in the intensity of received radio waves. It was difficult to understand what was happening, and it was difficult to handle complaints from users of program recording devices.

  Further, Patent Document 2 extracts frame synchronization information from a signal received from a broadcast station, and re-encodes audio decoded based on the frame synchronization information based on the frame synchronization information. Accordingly, since the re-encoded audio is synchronized with the original decoded audio in units of frames, the re-encoded audio can be produced without any deterioration in sound quality. However, when the intensity of the received radio wave decreases, it becomes difficult to continue decoding due to deterioration of the signal received from the broadcasting station, and if the level continues to decrease, decoding becomes impossible. If it cannot be decoded, the frame synchronization information cannot be extracted.

  For this reason, in a moving vehicle, if the strength of the received radio wave decreases due to an obstacle that interferes with broadcast reception, and the intensity of the received radio wave recovers after passing through the obstacle (the interpretation of Patent Document 2 shows that the second Therefore, it is necessary to re-establish the frame after a silent period by switching after recovery and providing reset means for resetting the frame synchronization. In order to resume recording when the level returns, it is necessary to re-synchronize the audio frame. Therefore, since the frame synchronization is reestablished in order to establish the synchronization at the time of return, the voice silent section is recorded for a long time. Considering the decrease in the strength of the received radio wave with respect to Patent Document 2, the frame synchronization is re-applied every time when the strength of the received radio wave is reduced, resulting in a recording with many silent intervals.

  The present invention was made in order to solve the above-described problems, captures a decrease in the intensity of the received radio wave, and maintains the synchronized state before reaching the intensity of the received radio wave that makes decoding reproduction impossible. By providing means for transitioning to an output pause state in units of audio frames, a device for smoothly switching the transition from resumption to recording interruption and leaving data processing recording at the time of interruption is provided.

An audio processing apparatus according to the present invention includes a receiving unit that receives a digital broadcast including audio data, a radio field intensity detecting unit that detects a radio field intensity of the digital broadcast, and a buffer unit that stores audio data included in the digital broadcast. Decoding means for decoding the audio data stored in the buffer means, data output means for outputting the audio data stored in the buffer means in units of frames, and audio data in units of frames output from the data output means Encoding means for encoding the radio wave intensity detection means when the radio wave intensity detection means detects radio wave intensity below a specified value and the buffer means cannot output audio data, or the radio wave intensity detection means does not exceed a specified value. And the decoding means If you can not decode the audio data outputted from said buffer means, outputs a predetermined data to said encoding means, said encoding means, when receiving said predetermined data, a frame unit output from the data output means Stop encoding audio data.

In the sound processing apparatus of the present invention, when the radio field intensity is lower than the specified value, the encoding means can encode other data instead of the audio data to be encoded. As a result, the processing of the encoder alone can be made faster.

(Embodiment 1)
First, the configuration of the first embodiment will be described with reference mainly to FIG. 1 which is a block diagram showing the configuration of the speech processing apparatus according to the first embodiment of the present invention. FIG. 1 shows an audio processing apparatus including a playback unit 10 that receives a digital broadcast and a recording unit 20 that records data received by the playback unit 10.

  The playback unit 10 is a set-top box (hereinafter abbreviated as STB), a part that receives satellite digital broadcasts with a digital broadcast recorder or the like, a tuner part of a personal computer (hereinafter abbreviated as PC), and the like. A recording device that encodes and records received data on a hard disk, an optical disk, a semiconductor memory, or the like.

  First, the flow of signal processing will be described. The antenna unit 1 receives radio waves transmitted from a plurality of digital broadcasting stations, and the input detection unit 2 demodulates the demodulating unit that selects one broadcasting station selected by the user from the broadcast radio waves received by the antenna unit 1. Then, error correction is performed and a transport stream is output. The transport stream is filtered into data such as video, audio, and program information for each packet ID by the transport decoder unit in the input detection unit 2 and temporarily recorded in the buffer 3. Video and audio data is decoded by the decoding unit 4 and output from an image display unit (not shown) or an audio output unit (not shown). Other data such as program information is sent from the buffer 3 to the reproduction information generation unit 5. Information such as a decoding error generated in the decoding unit 4 is also sent to the reproduction information reproducing unit 5. The input detection unit 2 detects a decrease in the intensity of the received radio wave during viewing that has occurred due to a sudden change in weather and sends the error level from the demodulation unit and transport decoder to the reproduction information generation unit 5.

  The audio decoded data decoded by the decoding unit 4 is output from the audio data output unit 6 as audio data effective for recording. Further, the reproduction information output unit 7 edits the reproduction information generated by the reproduction information generation unit 5 on the recording unit 20 and outputs it together with the audio data.

  In the recording unit 20, the re-input unit 12 receives the audio data and the reproduction information, and the encoding unit 13 knows the reproduction state on the reproduction unit 10 side from the reproduction information, and then the audio based on the decoded audio decoded data. Encoding is performed and recorded on a recording medium or the like.

  The digital data stream input to the input detection unit 2 is an audio / video signal that has been digitally compressed and encoded by a method such as MPEG such as digital broadcasting. Receives data from a digital broadcast tuner device, etc., separates audio / video signals that are multiplexed and recorded, extracts compressed video data and compressed audio data, and extracts video playback time information and audio playback time information Input analysis is performed.

  In MPEG, in order to realize synchronized playback of video and audio, the time of playback is managed for each decoding unit called video and audio access units (one frame for video and one audio frame for audio). A time stamp value is added. This time stamp value is called PTS (Presentation Time Stamp), and audio PTS (hereinafter referred to as “APTS”) for audio and video PTS (hereinafter referred to as “VPTS”) for video are defined. . When the PTS matches the above-mentioned system time reference STC, the access unit to which the PTS is added is output for reproduction. The accuracy of PTS is a value measured with a 90 kHz clock, and is represented by a 33-bit length. The reason for measuring at 90 kHz is that it is a common multiple of the frame frequency of both NTSC and PAL video systems, and to obtain higher accuracy than one sample period of audio.

  FIG. 2 shows a detailed functional block configuration example of audio decoding. The input detection unit 2 separates and extracts digitally compressed and encoded audio and video data from the data stream, extracts APTS added to the audio pack as audio playback time information, and adds it to the video pack. VPTS is extracted as video playback time information. The buffer 3 includes a compressed audio buffer unit 31 and an audio playback time information management unit 32. The compressed audio buffer unit 31 is a semiconductor memory that can store the audio data separated by the input detection unit 2. The buffer size varies depending on the compression rate of the encoded audio data to be handled, and is, for example, about several kilobytes to several tens of kilobytes. The audio playback time information management unit 32 generates and manages a table that associates audio data stored in the compressed audio buffer unit 31 with audio playback time information.

  The audio decoding unit 41 analyzes attribute information (audio header information) of the compression-coded audio data and decodes the audio data stored in the compressed audio buffer unit 31 according to the APTS information. The audio decoding unit 41 is provided with a PCM (Pulse Code Modulation) buffer unit 42 and an audio output unit 43. The PCM buffer unit 42 stores the decoded audio data. The audio output unit 43 outputs audio data to the outside. The audio data is further output to the audio data output unit 6 that is output to the recording unit 20.

  As shown in FIG. 7, the decoded audio data is not output to the audio data output unit 6, but the stream format data stored in the buffer itself may be output to the audio data output unit 6. The difference between the configurations of FIG. 1 and FIG. 7 is that in the case of FIG. 1, the decoded audio data is output to the audio data output unit 6, but in the case of FIG. The data is output to the data output unit 6. In the case of FIG. 7, for example, the audio output unit 43 outputs the stream as an encoded digital signal system such as a Dolby digital system. In such a case, instead of outputting the audio decode data, the STC is referred to simultaneously with the decoding, and the stream format audio data is output from the buffer 3 when the APTS matches or exceeds.

  The audio decoding unit 41 can be realized by a separate semiconductor integrated circuit or the like and a control program thereof. However, as long as each function can be realized, it may be realized by one semiconductor integrated circuit and the control program.

  Video signals are handled in the same way as audio signals. The video data separated by the input detection unit 2 is temporarily stored in the compressed video buffer unit, and a table associating the video data with the video reproduction time information is generated and managed as a video reproduction time information management unit.

  The video decoding unit analyzes attribute information (video header information) of the compressed and encoded video data, and decodes the video data stored in the compressed video buffer unit according to the video playback time information. The video decoding unit includes a frame buffer unit and a video output unit, the frame buffer unit stores the decoded video data, and the video output unit outputs the video data.

  Decoding timing and reproduction timing of video data and audio data are realized by the system time base reference unit 40. The system time base reference unit 40 generates a system time base STC inside the playback unit 10. In order to generate STC, in the case of a transport stream (TS) used in BS digital broadcasting, PCR (Program Clock Reference) is used, and the system time base reference unit 40 At the arrival of the last byte, the value of PCR is set to the same value as the system time reference STC. Thereby, a reference time can be set. The decoding unit 4 outputs and reproduces the access unit to which the PTS is added when the STC matches the APTS. Since the frequency of the PTS is 90 kHz, the audio data is decoded so that the STC and the APTS can be synchronized within this accuracy range. Also for the video signal, the video data is decoded so that STC and VPTS can be synchronized. As a result, synchronized playback of video and audio can be realized.

  Audio data received from a digital broadcast uses an audio signal compression method such as AAC (Advanced Audio Coding) coding. Analog baseband speech is sampled at a sampling frequency of 48 KHz, framing is performed every 2048 samples, and speech compression is performed by a technique such as MDCT (Modified Discrete Cosine Transform) conversion, quantization coding, or the like. When encoding and decoding are repeated, the voice quality gradually deteriorates. However, when encoding and decoding are performed with the same frame configuration, it is known that there is little degradation when the same frame configuration is not performed. When re-encoding received audio data, It is better to perform this in units of frames of decoded audio data.

  The operation of the speech processing apparatus when the intensity of the received radio wave has decreased will be described using FIG. 3 and FIG. Before the received radio wave intensity decreases, the decoding unit 4 prepares audio data decoded in units of audio frames and outputs the audio data from the audio data output unit 6. In step S401, the intensity of the received radio wave is measured. In step S402, it is determined whether the radio wave intensity is equal to or higher than a specified value at which no transport decoding error or audio decoding occurs. If it is equal to or greater than the specified value, the audio data classified by packet ID in S403 is stored in the buffer 3. In S404, the audio data is decoded.

  When the received radio wave intensity is below the specified value, a decoding error may occur. However, since the received audio data is stored in the buffer 3, the data in the buffer is sent to the decoding unit 4 in S405, and it is determined in S406 whether decoding output is possible in units of audio frames. If decoding is possible, it is confirmed in S407 that no decoding error has occurred. If there is no decoding error, the process returns to S401.

  The capacity of the buffer 3 is desired to be a capacity for storing several audio frames. This is because the audio data to be output is prepared in units of one audio frame. When the broadcasting method is different and the encoding method used is different, the audio frame size is different depending on each encoding method. The capacity must be based on the encoding method that can be received.

  On the other hand, if it is impossible to output data in S406 or if a decoding error occurs in S407, the audio data output unit 6 creates pause data and outputs it in the unit where the audio frame data was scheduled to be output in S408. To do. This pause data output is continued until the intensity of the received radio wave returns to a specified value or more. While the received radio wave intensity is decreasing, a decoding error or the like is detected, but since the buffer is played back before the decoding error occurs, the recording unit 20 switches to the pause data. No abnormal sound such as voice is sent.

  The return from the state where the intensity of the received radio wave is reduced will be described with reference to FIG. S421 is a state in which the intensity of the received radio wave is low and pause data is being output. From this state, in S422, it is determined whether or not the intensity of the received radio wave has recovered to the specified value level. On the other hand, if the data is recovered, the storage of the audio data in the buffer 3 is resumed in S423, and in S424, it is confirmed that no decoding error has occurred. In S425, the audio output decoded in units of audio frames is output. Resume. The return from the pause state to normal audio data output is performed at a position that is a multiple of the audio frame unit sampled at a sampling frequency of 48 KHz and framed every 2048 samples. This is because broadcasting from the broadcasting station side should continue if there is no decrease in the intensity of the received radio wave, so if it returns at a position that is a multiple of the audio frame unit, the audio quality is the same as before the interruption. Will be obtained. There is also a phase relationship with the audio data information before return. At the time of return, there is no need to re-synchronize for audio frame playback and re-encoding, and the re-encoding process for synchronized recording can be resumed quickly.

  That is, even during a period when the intensity of the received radio wave is reduced, the synchronization state can be quickly restored without performing the reset process by maintaining the synchronization state before the interruption. However, if the program that the user requested to be recorded ends and another program starts while the strength of the received radio wave is decreasing, the synchronized state continues between the programs. Since there is no guarantee, it is necessary to reset the frame and re-synchronize the frame.

  Furthermore, when the received radio wave strength in Fig. 3 is restored, the reason for returning to the normal decode output state after confirming that there is no decoding error even if the specified value exceeds the specified value exceeds the specified value and a decoding error occurs. This is because if the stable state is not achieved, the strength of the received radio wave is likely to decrease again due to unstable weather conditions.

  The specified value is determined as follows. Set before reaching the field strength that causes a decoding error that cannot be corrected. Even if it falls below the specified value, it is necessary to allow at least one to several audio frame data in the buffer 3 before a decoding error occurs and audio data cannot be output. That is, a value slightly higher than the radio wave intensity at which a decoding error occurs is set.

  FIG. 4 shows an example of a data output format for transmitting audio data and reproduction information. For example, this is a case where the IEC958 format used for optical output or the like is used, and one sample is the reciprocal of the sampling frequency. Audio-decoded linear PCM data (hereinafter abbreviated as LPCM) and stream data are stored as audio data in the audio data portion of one sample. The stream data is output in the state of the stream format data without decoding and converting the stream format data, which is an AAC encoded digital signal used in domestic BS digital broadcasting, into LPCM.

  The reproduction information may use the spare bit area in FIG. 4 or the reproduction information bit area in FIG. 4 which is a place where the audio attribute information is transmitted. For example, a channel status area (not shown) or a user data area (not shown) in the reproduction information bit area may be used. By using an area other than the preamble area or the audio data area, information on the reproducing unit 10 side can be transmitted to the recording unit 20 side without providing another transmission path. In addition to the audio attribute information (audio sample word length, pre-emphasis, the number of audio channels, and data defining the sampling frequency), the reproduction information bits collect decoding error information collected by the reproduction information generation unit 5, A value indicating the strength of the received radio wave, decoding time information, and the like are given. As for the pause data, by outputting the pause data defined in IEC958 as audio data, the semiconductor and output format used in IEC958 can be used as they are.

  This transmission may be a dedicated bus inside the apparatus, a transfer using IEC958, or a digital interface such as HDMI or IEEE1394. What is required is to transmit reproduction information simultaneously with transmission of audio data. In the case where the transmission path is configured as a separate path, when the transmission path becomes unusable due to any failure, the failure content cannot be transmitted to the recording unit 20 side.

  The reason why the pause data is continuously sent while the intensity of the received radio wave is low is that it is convenient for the case where the decoded audio is output by the optical output etc. and the monitor output audio is output via the connection amplifier. . When the transmitted audio is pause data, the connection amplifier performs output mute processing while maintaining the previous synchronization state, and performs standby processing for returning from the pause. In other words, the same audio data as that on the recording unit 20 side can be output to the external amplifier, and the output audio data for recording and amplifier output can be shared.

  Next, data processing on the recording unit 20 side will be described. The recording unit 20 performs audio encoding based on the audio data input from the re-input unit 12. If the audio data is LPCM, the audio data is recorded in the LPCM format or by performing audio encoding which is a compression encoding method designated by the user. When AV recording is performed at the same time as video encoding data, recording that can be synchronized with AV at the time of reproduction can be performed by recording the PTS information recorded on the video side and the audio side together. On the other hand, when the audio signal is recorded in the stream data format, PTS information for synchronizing with a desired video signal is added and recording with AV synchronization is performed without performing encoding processing again. Can do.

  A case where pause data is sent as audio data along the way will be described. When the audio data is in the LPCM format, for example, the encoding unit 13 can continue the encoding recording by replacing the LPCM data of the volume level 0 with the encoded data. That is, silence data continues to be recorded during the period when the intensity of the received radio wave is reduced. Further, while the intensity of the received radio wave is decreasing due to the user's selection, the audio encoding itself may be temporarily stopped to stop recording. However, if you leave information that allows the user to confirm the period when the recording was canceled after the period when the recording was canceled, even if the user did not witness the recording time, such as an answering record, , You can trace what caused recording failure. Also, when the audio data is in the stream data format, the period of decrease in the received radio wave intensity is overcome by replacing the data as in the LPCM or by taking a recording suspension method. For example, in the case of stream data in the AAC format, during the pause data reception period, data obtained by AAC encoding the LPCM data at the volume level 0 is prepared in advance, and the volume level 0 is kept while the pause data reception is continued. Is replaced with AAC encoded data. Since the side that sends pause data guarantees that pause data is output in units of audio frames, the recording unit may be replaced with AAC encoded data of 0 volume level in units of audio frames.

  When audio data is sent in the LPCM format using the configuration example of FIG. 1, the LPCM data at a volume level 0 may be sent instead of pause data output in units of audio frames. In this way, the encoder on the recording unit 20 side continuously encodes audio data while avoiding noise recording if the sound of the volume 0 level is encoded all the time while the received radio wave intensity is decreasing. Can continue. It is assumed that the reproduction information bit data described above is continuously given even during output of audio data.

  Further, when audio data is sent as stream data such as AAC format using the configuration example of FIG. 7, the volume level, not pause data, is set in units of audio frames on the playback unit 10 side while the received radio wave intensity is reduced. A format in which data obtained by AAC encoding zero LPCM data may be used. In this way, the encoder on the recording unit 20 side only needs to encode the stream data all the time even when the received radio wave intensity is decreasing. At this time, it is assumed that the data of the reproduction information bit described above is continuously given even during output of the audio data.

  By receiving the reproduction information from the reproduction unit 10 in the recording unit 20, the encoder unit 13 can know in advance what processing is expected next. For example, intensity information of received radio waves can be obtained. That is, it is possible to know whether the intensity of the received radio wave is in a recovery state or not recovered. In the recovery state, it can be seen that the return from the pause state to the normal recording state is close, and by preparing the return processing from the end of the pause data replacement or the temporary recording interruption, the normal recording state can be quickly restored. A transition can be reached. If an error is detected during decoding, such as a CRC (Cyclic Redundancy Check) error, it is impossible to decode the stream data even if it is recorded as it is. It turns out that it is better to continue to select the process of interruption. In addition, when time information or the like is transmitted, the voice processing device can leave time information such as when recording is interrupted while the intensity of the received radio wave is decreasing. Therefore, it is understood that the recording is not interrupted but the recording is interrupted due to a decrease in the intensity of the received radio wave.

  In particular, the reproduction information while the intensity of the received radio wave is lowered, that is, while the pause data is output, indicates the reason why the reproduction unit 10 outputs the pause data. In addition, pause data in units of audio frames is to accelerate recovery from a decrease in received radio wave intensity as much as possible. By continuing to output data continuously, it is possible to maintain the frame synchronization, so that the sound processing device installed in a running car, etc., may be affected by radio wave interference caused by a running obstacle. If it changes every moment, it can be restored to a recordable state in a little.

  The above has described the reproduction and recording of audio data, but the same method can be used for video data. In the case of video, since the period of one video frame is specified in the NTSC system or the PAL system, even when the received radio wave intensity decreases and decoding becomes impossible, the playback unit 10 does not receive the frame before the decoding error. Processing for re-outputting data (repeat reproduction) and processing for replacement with blue back data may be performed and output every video frame period. If reproduction information such as radio wave intensity and decoding error is sent during the blanking period, the recording process at the recording unit 20 side can be continued.

  Furthermore, if synchronization is performed in units of audio frames, restoration of AV synchronized recording can be stopped without sending PTS information. In the case of digital broadcasting, not all audio frames have APTS information. Therefore, when establishing synchronization based on the value of APTS, even if the decoding error does not occur, it is not known at what timing audio reproduction may be started until APTS information arrives. First, the audio frame data with the APTS is compared with the system time. In other words, if the APTS is no longer updated due to a decrease in the strength of the received radio wave, the return from the decrease in the strength of the received radio wave first receives an audio frame with APTS information and re-synchronizes the audio frame. You must resume recording.

  However, in the case of the present invention, since the synchronization state before the intensity reduction of the received radio wave is maintained in units of audio frames, recording is resumed without waiting for the audio frame of the APTS information and without re-synchronizing the audio frame. It is something that can be done.

  The audio processing apparatus according to the present invention has an interface between the playback device side and the recording device side, and performs recording continuation on the recording device side based on the received radio wave intensity condition on the playback device side. It can be used for the purpose of selecting the encoded audio signal recording that matches the intensity condition of the broadcast reception radio wave on the playback device side while keeping it.

The figure which shows the structure of the apparatus in Embodiment 1 of this invention. The figure which shows the audio | voice decoding structure in Embodiment 1 of this invention. The figure which shows the data processing inside an apparatus at the time of the radio field intensity fall of the apparatus in Embodiment 1 of this invention The figure which shows the example of a structure of the audio data output format in Embodiment 1 of this invention. The figure which shows the process at the time of the intensity | strength fall of the received radio wave of the apparatus in Embodiment 1 of this invention The figure which shows the process at the time of the intensity | strength return of the received radio wave of the apparatus in Embodiment 1 of this invention The figure which shows the other connection structure of the apparatus in Embodiment 1 of this invention.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Antenna part 2 Input detection part 3 Buffer 4 Decoding part 5 Reproduction information generation part 6 Audio data output part 7 Reproduction information output part 10 Reproduction part 12 Re-input part 13 Encoder part 20 Recording part

Claims (1)

Receiving means for receiving a digital broadcast including audio data;
Radio wave intensity detecting means for detecting the radio wave intensity of the digital broadcast;
Buffer means for storing audio data included in the digital broadcast;
Decoding means for decoding the audio data stored in the buffer means;
Data output means for outputting the audio data decoded by the decoding means in units of frames;
Encoding means for encoding audio data in frame units output from the data output means,
The data output means includes
When the radio wave intensity detecting means detects a radio wave intensity below a specified value and the buffer means cannot output audio data,
Or
When the radio wave intensity detection means detects a radio wave intensity below a specified value, and the decoding means cannot decode the audio data output from the buffer means,
Outputting predetermined data to the encoding means;
The encoding means includes
An audio processing apparatus that stops encoding audio data in units of frames output from the data output means when the predetermined data is received.

Images