JP4613674B2 - Audio playback device - Google Patents

Description

  The present invention relates to an audio reproducing apparatus that reproduces compression-coded audio data in cooperation with an output device connected by a digital interface or the like.

  2. Description of the Related Art Conventionally, 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, and a hard disk, decodes (expands), and outputs the data is known. Video and audio reproduction can be realized by the data reproduction device sending the decompressed data to a monitor, an acoustic speaker, or the like connected to the device.

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 playback system and playback method will be described below. Note that the applicant has recognized the audio equipment, the control device, and the video receiving system disclosed in Patent Document 1, Patent Document 2, and Patent Document 3, and in order to clarify the problem of the present invention, The contents of the technology will be briefly described as a conventional example.

  First, a conventional audio device in Patent Document 1 will be described. FIG. 11 is an explanatory diagram for illustrating the configuration of this device. The flow of audio processing of the audio device in FIG. 11 is as follows. An encoded digital signal from an external source device side (not shown) is input from the digital interface receiver 202 of the destination-side audio device, and the digital signal processing device 204 corresponding to decoding of a plurality of encoded digital signals is used. Decoding and sound field reproduction signal processing are performed, and a reproduction audio signal is output from a D / A (abbreviation of D / A converter) 205. In the case of this apparatus, it is discriminated whether it is a specific encoded digital signal from various encoded digital signals, and signal processing based on a sound field reproduction mode suitable for the specified encoded digital signal is executed. For example, in an audio device such as an AV amplifier, a decoding process is performed on an encoded digital signal input via an optical cable or a coaxial cable according to the type.

  Next, a conventional control device and control method in Patent Document 2 will be described. It comprises an audio decoder that decodes audio data input from the IEEE1394 cable, an audio data output unit that outputs audio signals output from the audio decoder, and a CPU that controls the IEEE1394 interface. In the case of this control device, it has device-specific data that stores the type of playback device-encoded digital signal of the source device on the partner side that is connected to the control device, from the signal on the bus input from the IEEE1394 interface. When the control device on the Nation device reads the device-specific data, if the encoded digital signal on the bus is of a decodable type, it captures the signal from the bus, performs the decoding process, and outputs the decoded audio To do.

  In all of these, a specific encoded digital signal is supplied from the source device side, and the destination device side decodes and reproduces the received encoded digital signal. In the audio device of Patent Document 1, an audio device that is a destination device determines a decoded sound and a sound field reproduction mode according to the type of input encoded digital data. In the control device of Patent Document 2, device-specific information is acquired from the source device side, and when a decodable encoded digital signal runs on the bus, it is input and decoded and reproduced.

A video receiving system in Patent Document 3 will be described. FIG. 12 is an explanatory diagram showing the configuration of this system. The processing flow of the video receiving system in FIG. 12 is as follows. The channel selection operation of the television 100 switches the input to the video and audio signals output from the STB 112, which is a set-top box (hereinafter abbreviated as STB) connected to the television 100, and can also perform the channel selection operation on the STB 112 side. Is. This is because the information on the STB 112 on the source device side connected by HDMI (High Definition Multimedia Interface) or the like is obtained by the TV 100 side on the destination device side and the tuner tuning operation on the source side is controlled. Through the operation on the destination device side, the operation is easily performed.
Japanese Patent Laid-Open No. 9-130898 (page 3-6, FIGS. 1 and 2) Japanese Unexamined Patent Publication No. 2000-151752 (page 3-5, FIG. 1, FIG. 3, FIG. 4) JP 2004-304220 A (page 6-16, FIG. 1)

  When an encoded digital signal that cannot be decoded on the destination side is sent from the source side, the problem that the destination device side cannot reproduce the sound has not been solved. In the video reception system disclosed in Patent Document 3, the process from the destination device side to the channel selection operation on the source device side is performed one step further, but this changes the decoding reproduction condition of the encoded digital signal. It was not a thing but a channel selection channel was changed to the last. Therefore, if the program specified by channel selection is not an encoded digital signal that can be decoded on the destination side, it is impossible to decode and reproduce the sound.

  The present invention has been made to solve the above-described problems, and the source side has a function of selectively reproducing an encoded digital signal by obtaining a decoding condition that can be reproduced on the destination side. An apparatus capable of decoding and reproducing on the destination device side is provided.

The audio reproduction device according to the first and second aspects of the present invention (corresponding to claims 1 and 2) is acquired by an information acquisition unit that acquires information for specifying an audio reproduction condition from a connected device, and an information acquisition unit The control means for setting the reproduction condition of the audio data based on the information obtained , the reproduction means for reproducing the audio data according to the reproduction condition set by the control means, and the sampling frequency among the information acquired by the information acquisition means If the sampling frequency of the audio data is higher than the sampling frequency of the audio data, the device is provided with transmission means for performing the up-sampling process on the audio data reproduced by the reproduction means, and can be decoded on the audio output means side. Since the data can be selected on the playback means side, the user can play back audio that meets the playback conditions of the audio output means without operating the playback means. It is.

According to a third aspect of the present invention (corresponding to claim 3 ), the control means further receives the sound reproduction channel information of the device and switches the number of sound channels to be reproduced. Since the number of output channels included in the playback means can be selected on the playback means side, the user can perform sound playback that meets the playback conditions of the sound output means without changing the channel number setting on the playback means side. Is.

In the sound reproducing apparatus according to the fourth aspect of the present invention (corresponding to claim 4 ), the control further receives sound reproduction sampling information of the device and switches the sampling frequency to be reproduced, and transmits it to the sound output means. Since it is possible to output after obtaining the transmission conditions and sampling frequency conditions of the digital analog converter (hereinafter referred to as DAC) on the audio output means side on the playback means side, it becomes possible to output as high sampling data as possible on the playback means side, High sound quality reproduction is possible.

An audio reproduction device according to the fifth aspect of the present invention (corresponding to claim 5 ) further comprises storage means for recording audio reproduction conditions of the device and device information for specifying the device, and when connected to the device, By comparing the device information recorded in the storage unit with the previously connected device information, the playback unit plays back the audio playback condition recorded in the storage unit. Therefore, even if the user switches the audio output means from time to time, the data unique to the connection partner device can be stored on the reproduction means side, so that the complexity of setting the connection conditions is reduced.

According to a sixth aspect of the present invention (corresponding to claim 6 ), when the device can decode the audio data, the reproduction means outputs the compression-encoded audio data, and the audio output Only when the means cannot be decoded, the playback means decodes the data into uncompressed data and outputs the sound, so that the audio output means can be used even if a device that does not support the decoding of the compressed data is connected. The output setting can be completed without taking time and effort.

An audio reproducing device according to the seventh aspect of the present invention (corresponding to claim 7 ) outputs audio data having a high compression bit rate number that is compression-encoded from content, and outputs it with higher sound quality. For a user who wants to make a connection, the system automatically prioritizes a connection setting with a small compressed ratio, that is, a high compression bit rate and a large amount of audio information.

The audio playback apparatus according to the eighth aspect of the present invention (corresponding to claim 8 ) outputs audio with priority given to audio data having a large amount of audio playback channel information from the content. The setting close to the playback status is made effective.

An audio reproducing apparatus according to the ninth aspect of the present invention (corresponding to claim 9 ) outputs the audio data in which the audio reproducing condition is losslessly compressed and encoded with priority, and is included in the compressed audio data. Therefore, in order to select the lossless compression encoded data, the higher sound quality reproduction is selected with priority and the reproduction is realized.
The sound reproducing device according to the tenth aspect of the present invention (corresponding to claim 10) is to preferentially output sound data whose sound reproducing conditions are high in sound reproducing sampling frequency, and to select the sound reproducing with higher priority, Reproduction is realized.

  The audio reproducing apparatus of the present invention has a function of selecting and reproducing an encoded digital signal and transmitting audio data by acquiring reproducible decoding condition information possessed by audio output means on the destination side Can do. In other words, there is an advantage that encoded audio encoded data that can be reproduced can be selected and converted into a format that can be received on the audio output means side from the output information on the audio output means side that is not conscious of the user.

(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 audio reproduction device according to the first embodiment of the present invention. FIG. 1 shows a configuration including a regenerator 10 and an output device 20. The player 10 is a DVD player, DVD recorder, personal computer (hereinafter abbreviated as PC), STB, or the like, or a hard disk recorder when a hard disk is used in place of the optical disk, and the output machine 20 is an AV amplifier, AV output devices such as TV monitors.

  First, the flow of AV signal processing will be described. In the playback device 10, there are input devices such as a disk playback unit 1 and an antenna unit 2. The disc reproducing unit 1 performs binarization processing on a data stream read from an optical disc such as a DVD-ROM via an optical pickup, and outputs it as a digital data stream. The antenna unit 2 receives a digital broadcast data stream via an antenna or the like. The input unit 5 separates the digital data stream input from the disc playback unit 1 or the antenna unit 2 into video data and audio data, and the decoding unit 6 decodes each of them as necessary from the video signal and the audio signal. Is output as an AV signal.

  The AV signal is transmitted from the transmission unit 7 to the output device 20. The AV signal sent from the transmitter 7 is received from the receiver 12 in the output machine 20 and sent to the output unit 13.

  The output unit 13 receives the encoded digital signal before decoding and performs decoding processing to reproduce it as an audio signal, or receives decoded data decoded by the decoding unit 6 and reproduces it as an audio signal.

  The digital data stream input to the input unit 5 is contents encoded by various encoding methods such as MPEG, and audio / video signals that are digitally compressed and encoded such as digital broadcasting. Receives data from optical disc devices, digital broadcast tuner devices, etc., separates multiplexed and recorded audio / video signals, separates compressed video data and compressed audio data, video playback time information, and audio playback Input analysis for extracting time information 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 the decoding unit 6. In response to input from the input unit 5, the decoding unit 6 receives a compressed video buffer unit 22, a video playback time information management unit 23, a system time reference reference unit 24, a video decoding unit 25, a compressed audio buffer unit 26, The audio reproduction time information management unit 27 and the audio decoding unit 28 are included.

  The input unit 5 separates and extracts the digital compression encoded audio and video data from the data stream, extracts the VPTS added to the video pack as video playback time information, and APTS added to the audio pack Are extracted as audio playback time information.

  First, components related to video processing will be described. The compressed video buffer unit 22 primarily stores the video data separated by the input unit 5. The video playback time information management unit 23 generates and manages a table in which the video data stored in the compressed video buffer unit 22 is associated with the video playback time information.

  The video decoding unit 25 analyzes attribute information (video header information) of the compression-encoded video data, and decodes the video data stored in the compressed video buffer unit 22 according to the video playback time information. The video decoding unit 25 is provided with a frame buffer unit 31 and a video output unit 32. The frame buffer unit 31 stores the decoded video data. The video output unit 32 outputs video data to the outside. This video data is further output to the transmission unit 7 connected to the output device 20.

  Next, components related to audio processing will be described. The compressed audio buffer unit 26 is a semiconductor memory that can store the audio data separated by the input unit 5. 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 27 generates and manages a table associating the audio data stored in the compressed audio buffer unit 26 with the audio playback time information.

  The audio decoding unit 28 analyzes attribute information (audio header information) of the compression-encoded audio data and decodes the audio data stored in the compression audio buffer unit 26 according to the audio reproduction time information. The audio decoding unit 28 is provided with a PCM (Pulse Code Modulation) buffer unit 33 and an audio output unit 34. The PCM buffer unit 33 stores the decoded audio data. The audio output unit 34 outputs audio data to the outside. The audio data is further output to the transmission unit 7 connected to the output device 20.

  In some cases, the optical output terminal connected to the audio output unit 34 outputs the stream as an encoded digital signal system such as a Dolby Digital system. In such a case, the audio is not temporarily decoded and is temporarily stored in a stream buffer (not shown), and the STC is referred to, and the audio is output when the APTS matches or exceeds.

  The video decoding unit 23 and the audio decoding unit 28 described above can be realized by separate semiconductor integrated circuits and the like and their control programs. However, as long as each function can be realized, it may be realized by one semiconductor integrated circuit and the control program.

Next, a configuration for reproducing video data and audio data output from the video output unit 32 and the audio output unit 34 in synchronization will be described. The decoding timing and the reproduction timing are realized by the system time base reference unit 24.
The system time base reference unit 24 generates a system time base STC inside the regenerator 10. In order to generate an STC, an SCR (System Clock Reference) value defined in a program stream (PS) used in a DVD disc or the like is used. The SCR is described in the pack header. When the input data stream is a transport stream (TS) used in BS digital broadcasting, a PCR (Program Clock Reference) is used.

  The system time base reference unit 24 sets the value of the SCR to the same value as the system time base STC when the last byte of the video data arrives. Thereby, a reference time can be set. Further, by combining the setting of the STC value and a PLL (phase lock loop) circuit (not shown), it is possible to have an STC whose frequency completely matches the reference time system clock of the decoding unit 6. The system clock frequency of the system time reference STC is 27 MHz. Each PTS (frequency: 90 kHz) is referred to by dividing the STC by a counter or the like.

  When the system time reference STC matches VPTS, the video decoding unit 25 outputs and reproduces the access unit to which the PTS is added. The audio decoding unit A4 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, each of the video data and the audio data is decoded so that the STC can be synchronized with the VPTS and the APTS within this accuracy range. As a result, synchronized playback of video and audio can be realized.

  Video decoding and audio decoding are performed so that VPTS and APTS are synchronized with the STC. In particular, if the VPTS is set to the APTS between the time of 50 milliseconds preceding and the time of 30 milliseconds delay, the lip sync will not be noticed.

  The transmission unit 7 transmits the AV data decoded by the decoding unit 6 to the reception unit 12 side. This transmission / reception will be described using HDMI as an example, but a digital interface such as IEEE1394 may be used. Also, a wired or wireless network line such as a LAN may be used. The digital interface between the playback device 10 and the output device 20 needs to have a function of transmitting information on the output device 20 side in order to send audio data to the output device 20 side.

  Next, the configurations of the transmission unit 7 and the reception unit 12 will be described using HDMI as an example here with reference to FIG. The HDMI includes a transmitter that is the transmission unit 7 and a receiver that is the reception unit 12. The transmitter is the playback side of video / audio sources such as STB, DVD, VCR, and the receiver is the output side of TVs, displays, AV amplifiers, and the like.

  In the transmission unit 7 (source unit 81 in FIG. 3), the reception unit 12 (sink unit 82 in FIG. 3) is connected by an HDMI cable. AV data 86 is unidirectional from the transmitter unit 83 to the receiver unit 84 side, but a DDC line 87 and a CEC line 88 described later are bidirectional. A CEC (Consumer Electronics Control) line 88 transmits device control signals, and transmits commands such as power on / off and output channel setting change. A DDC (Display Data Channel) line 87 is used to transmit a control signal such as a display or an amplifier.

The source unit 81 transmits a control command for obtaining information for specifying the sink unit 82 from the DDC line 87. Then, an EDID (Extended) corresponding to the device specific information unit 11 is obtained.
Display Identification Data) From the ROM 85, the information is transmitted to the source unit 81 as device specific information together with the device ID of its own device.

  FIG. 4 shows an example of the data structure of EDID. It is mainly composed of four blocks: a video data block, an audio data block, a speaker arrangement data block, and a manufacturer specific information (Vendor Specific Data) block.

  In the description of this embodiment, blocks related to audio will be described in detail. The audio data block stores audio information that can be reproduced by the receiver. Various kinds of decodable audio decoder type information, the number of reproduction channels related to the decode information, sampling frequency, quantized data bit length, compression rate information, and the like are stored. The speaker arrangement block stores speaker arrangement information (front speaker, rear speaker, center speaker, etc.) that can be output by the receiver.

  These EDID information may be stored as EDIDROM 85 information in the memory unit 4 as a storage unit together with the device ID. When the power is turned on, the transmitter can identify the receiver as long as it knows the device ID of the receiver.

  In this way, the information acquisition unit 8 receives the device ID from the device specific information unit 11. The CPU unit 3 serving as a control unit refers to a table defined for the acquired device in an internal memory serving as the memory unit 4 or an external memory card. For example, as shown in FIG. 6, the table describes correspondences between device IDs and various types of reproduction audio information. And based on apparatus ID and its table, the outputable format information which the output machine 20 has can be obtained. If the acquired information is stored as a default setting value in the data processing apparatus, AV reproduction can always be performed in the same state as long as the device connection is not changed. If there is a change in the connected device ID, etc., the information on the partner device side is received each time, and the source side that has only to change the setting can update the contents of the memory unit 4 by receiving the EDID information as device specific information. That's fine.

  As these externally connected devices, monitor output devices such as TVs, interface devices such as audio output amplifiers and AV amplifiers having an AV selector function, portable output devices, in-vehicle AV playback devices, and the like are assumed.

  If the audio data received by the receiving unit 12 is PCM data, it is sent to the output unit 13 as it is. On the other hand, in the case of stream-format data that is an encoded encoded digital signal, the data is sent to the output unit 13 after being decoded into PCM data. Then, reproduced sound is output from a speaker unit (not shown) connected to the output unit 13.

  Next, a processing routine in which the CPU unit 3 of the player 10 selects an output format output from the transmission unit 7 will be described with reference to FIG. In step S401, the input unit 2 acquires the decoded audio type information of digital TV audio (hereinafter abbreviated as DTV) input from the antenna 2 in step S401. In the case of BS digital broadcasting in Japan, encoding is performed by an encoding method called AAC (Advanced Audio Coding).

  The device ID information on the output machine 20 side is received via the HDMI interface described above, and is compared with the information stored in the memory unit 4 in step S402. If there is matching information, the outputable format information corresponding to the device ID is read. On the other hand, if there is no matching device ID, output audio information such as an outputable format on the output machine 20 side is acquired in step S403.

  Here, the description will be made assuming that the information of the contents of the Audio Descriptor 92 in FIG. 4 is associated with the device ID and stored in the player 10 side.

  In step S404, it is determined whether an AAC format exists in the output format. When stream-type data, which is an encoded digital signal called an AAC system, is transmitted and the output device 20 determines that it can be decoded, in step S405, the decoding unit 6 converts the AAC system stream. The CPU unit 3 designates a setting for outputting to the transmission unit 7 in the state of the stream format data that is an encoded digital signal without performing decoding conversion into linear PCM (hereinafter abbreviated as LPCM).

  On the other hand, if the output unit 20 cannot receive the signal using the AAC method, it is determined in step S406 whether the LPCM can receive the signal using the multi-channel method. If it can be received by the multi-channel system, in step S407, the decoding unit 6 sets 5.1ch or the like surround broadcast encoded by the AAC system to be decoded and output in the multi-channel output format, and sets the output format. Processing of function F500 is performed. The output format setting function F500 will be described later.

  When the output device 20 cannot receive data by the multi-channel method, the LPCM 2-channel output setting is selected. In step S408, the 5.1ch surround broadcast encoded in the AAC format is also down-mixed into a stereo 2-channel sound by a channel number converting means (not shown) (multiple channels of output sound are converted to 2). By performing the process of synthesizing and adding to the output sound of the channel, the output setting is performed in the 2-channel output format, and the process of the output format setting function F500 is performed.

  The output format function F500 will be described with reference to FIG. Although the setting of the number of output channels is determined before this function is called, the sampling rate of audio data output by the LPCM is selected from these. First, in step S501, whether or not the output machine 20 can handle a sampling frequency of 176 KHz (more precisely, 176.4 KHz, which is four times 44.1 KHz, but will be abbreviated as 176 KHz here) or 192 KHz. Judging. If it can be handled on the output machine 20 side, it is determined in step S502 whether the decoded audio data is 176 KHz or higher. If it is 176 KHz or higher, output setting is performed at the sampling rate as it is at the source rate in step S503. In the case of BS digital broadcasting in Japan, since it is often 48 KHz, it is determined in step S502 that it is less than 176 KHz. In step S504, sampling processing is performed on data output at 192 KHz, which is four times 48 KHz, by performing filter processing for upsampling by the sampling conversion means in the decoding unit 6, and output as audio data at a sampling frequency of 192 KHz. Set to do.

  In step S501, if the output machine 20 side cannot handle the sampling frequency data of 176 KHz or 192 KHz, in step S505, 88 KHz (more precisely, 88.2 KHz that is twice 44.1 KHz) or 96 KHz Determine whether the sampling frequency can be handled. If it can be handled, it is determined in step S506 whether the decoded audio data is 176 KHz or higher. If it is 176 KHz or higher, the output machine 20 cannot perform processing at the sampling frequency as it is, and therefore, the source downsampling output is set in step S507 by the sampling conversion means. The decoding unit 6 is set to perform a down-sampling filter process for outputting at 88 KHz or 96 KHz.

  If it is determined in step S506 that the sampling frequency is less than 176 KHz, it is determined in step S508 whether the decoded audio data is 88 KHz or higher. If it is 88 kHz or higher, output setting is performed at the sampling rate as it is at the source rate in step S509. On the other hand, in the case of a source of less than 88 KHz, in step S510, sampling processing is performed on data output at a double sampling rate by performing a filtering process that performs upsampling in the decoding unit 6, and at a sampling frequency of 88 KHz or 96 KHz. It is set to output as audio data.

  In step S505, when the output machine 20 can handle only data with a sampling frequency of less than 88 KHz, an output that performs output at either a sampling frequency of 32 KHz, 44 KHz (exactly 44.1 KHz), or 48 KHz. Settings are made. Even if the source rate is a sampling frequency exceeding 48 KHz, by performing a downsampling process in the decoding unit 6, it is output as sampling frequency data that can be handled on the output machine 20 side.

  In the first embodiment, the case where the original encoded digital signal reproduced on the reproducing apparatus 10 side is only one type that has been compression-encoded such as the AAC method has been described. In addition to the AAC system, the audio compression coding system includes a DTS system (a speech coding system developed by Digital Theater Systems), a Dolby Digital system, and the like. In the first embodiment, one type of compression coding is used. The reproduction method when the content that has been encoded and digitized only by the method has been explained.

  Whether or not the playback device 10 obtains playback information of the output device 20 and there is an environment where the encoded digital signal can be received and decoded first, or whether or not it can be received and played back in multi-channel (3 channels or more) in the LPCM format. If not, the reception / reproduction with 2 channels is selected. Furthermore, when playback with higher sound quality is required by the user, if the encoded digital signal is decoded on the playback device 10 side, if the original sampling frequency is high, it is not reduced to a lower rate. It is preferable to output. Further, by performing filter processing such as upsampling on the playback device 10 side, it is possible to realize playback of higher-quality sound. Therefore, in order to achieve sound output with high sound quality, it is meaningful that the playback device 10 side acquires information on the sampling frequency setting that can be received and played back on the output device 20 side, and outputs it at a higher sampling frequency.

  The same applies to the bit length. If the encoded digital signal is decoded, the bit length (for example, 24 bits) is preferably transmitted as 24 bits if reception and reproduction are possible on the output device 20 side. Of course, when connected to another output device and the other party can only receive and reproduce in 16 bits, it is desirable that the player 10 side obtains the information and outputs it in 16 bits.

  By outputting the original encoded digital signal audio information as much as possible to the output machine 20 as much as possible, a reproduction environment that is as close as possible to the original sound intended by the content producer is to be realized.

  As described above, according to the first embodiment of the present invention, the output information such as the decoding information on the audio output means side has the function of setting the reproduction condition as the reproduction selection on the reproduction means side, so that the user can reproduce the reproduction means. It is possible to provide an audio reproducing apparatus that can enjoy music and the like without changing the reproducing conditions on the audio output means side or the audio output means side.

(Embodiment 2)
A sound reproducing apparatus according to Embodiment 2 of the present invention will be described. The following description is centered on differences from the first embodiment. A configuration example is the same as that of the first embodiment, and will be described with reference to FIGS. 1, 2, and 3.

  Hereinafter, a processing routine in which the CPU unit 3 of the player 10 selects an output format output from the transmission unit 7 will be described with reference to FIG. When playing back from the content, the encoded digital signal can be played back by selecting and playing back an arbitrary method from, for example, the compression-encoded AAC method and the LPCM method that is simply uncompressed and encoded. Processing contents will be described.

  In step S421, the input unit 2 acquires the decoded audio type information of an encoded digital signal input from the reproduction of the optical disc 1 or the like. Here, encoding is performed by the AAC system that is compression-encoded and the LPCM system that is encoded without compression.

  As in the first embodiment, the device ID information on the output machine 20 side is received via the HDMI interface described above, and is compared with the information stored in the memory unit 4 in step S422. If there is matching information, the outputable format information corresponding to the device ID is read. On the other hand, if there is no matching device ID, output audio information such as an outputable format on the output machine 20 side is acquired in step S423.

  Then, in step S424, it is determined whether or not reception is possible using the LPCM multi-channel method. If reception is possible using the multi-channel method, in step S425, decoding output is set in a multi-channel output format using LPCM, and processing of an output format setting function F500 is performed.

  If it is determined in step S424 that reception cannot be performed using the multi-channel method, it is determined in step S426 whether the AAC method exists. When stream format data, which is an encoded digital signal called AAC format, is transmitted and it is determined that the output device 20 can decode the stream format data, the decoding unit 6 converts the AAC format stream in step S427. The CPU unit 3 designates a setting for outputting to the transmission unit 7 while maintaining the state of the stream format data that is the encoded digital signal without converting into the LPCM.

  When the output machine 20 cannot receive the data in the AAC system, the LPCM 2-channel output setting is selected. In step S428, down-mix processing of LPCM multi-channel to stereo 2-channel audio (processing to synthesize and add multi-channel output audio to 2-channel output audio) is performed to set output in a 2-channel output format, The output format setting function F500 is processed.

  In the case where both non-compressed encoded data and compressed encoded data can be selected at the time of content reproduction, this is done while acquiring reproduction environment information on the output device 20 side via HDMI or the like. By selecting compression and multi-channel playback, high sound quality playback is to be realized.

  When the output unit 20 cannot receive the multi-channel method using LPCM and the playback unit 10 has a function of re-encoding the decoded audio data with multi-channel, the output unit 20 side It is desirable to re-encode with an encoding method that can be decoded. That is, when the content is reproduced, even if the encoding method cannot be decoded on the output device 20 side, the reproduction device 10 side has re-encoding means (encoder) that can be decoded on the output device 20 side. The audio data encoded by the re-encoding means may be output from the transmission unit 7.

  Furthermore, when a plurality of playback contents are combined and played back and output to the output device 20 side, the following configuration is made. First, the reproduction condition is acquired from the device specific information unit 11 of the output device 20, and it is determined whether reception / reproduction is possible with the LPCM multi-channel. If possible, a plurality of encoded data is decoded and synthesized by the playback device 10 and output from the transmitter 7 in the form of LPCM multi-channel. Next, if the LPCM multi-channel cannot be received / reproduced, the encoding / converting means (not shown) capable of encoding the multi-channel on the regenerator 10 side can be used to decode the output unit 20 side. It outputs from the transmission part 7 in the re-encoded form. When there is only a coding conversion means to a coding method that cannot be decoded on the output device 20 side in the playback device 10, or when there is no coding conversion means, a plurality of contents are each converted into 2-channel output audio. After downmixing, the signals are combined and output from the transmission unit 7 in the form of a two-channel output format. In this way, even when a plurality of encoded sounds are reproduced simultaneously on the player 10 side, the synthesized sound can be enjoyed in a multi-channel state as much as possible by acquiring the reproduction condition information on the output device 20 side. be able to.

(Embodiment 3)
A sound reproducing apparatus according to Embodiment 3 of the present invention will be described. The following description is centered on differences from the first embodiment. A configuration example is the same as that of the first embodiment, and will be described with reference to FIGS. 1, 2, and 3.

  Hereinafter, a processing routine in which the CPU unit 3 of the player 10 selects an output format output from the transmission unit 7 will be described with reference to FIG. A description will be given of processing contents when an encoded digital signal can be reproduced by selecting an arbitrary method from a plurality of compression-encoded methods (for example, DTS method and Dolby digital method) when reproducing from content. . It is assumed that there are a plurality of compression-encoded data, but no non-compression encoding data.

  In step S461, the input unit 2 acquires the decoded audio type information of the encoded digital signal input from the reproduction of the optical disc 1 or the like. Here, encoding is performed by a compression-coded DTS method and a compression-coded Dolby digital method.

As in the first embodiment, the device ID information on the output machine 20 side is received via the HDMI interface described above, and is compared with the information stored in the memory unit 4 in step S462. If there is matching information, the outputable format information corresponding to the device ID is read. On the other hand, if there is no matching device ID, output audio information such as an outputable format on the output machine 20 side is acquired in step S463.
In step S464, it is determined whether or not the DTS method exists in the output format. When data in a stream format, which is an encoded digital signal called the DTS method, is transmitted and it is determined that the output device 20 can decode the data, in step S465, the DTS as further publicly described is used. Among the methods, a more detailed decoding method such as the DTS-HD method is examined.

  For example, FIG. 5 shows information of the Audio Descriptor 921 stored in the memory unit 4. Here, in the case of the DTS system, which is an encoded digital signal that has been compression-encoded, a case where more detailed decoding constraint condition information is referred to is shown.

  FIG. 5 shows the number of playback channels, the sampling rate, and the upper limit compression bit rate as various constraints in the two methods of the Dolby Digital method and the DTS method. These are obtained from the output device 20 side via HDMI or the like. FIG. 5 shows that, for example, a plurality of detailed playback conditions are shown in the DTS system. If the playback environment on the output device 20 side, that is, the decoding conditions, has the largest number of reproducible channels and the sampling rate and the compression bit rate can be played back, that side is selected with priority. Thus, it is possible to prepare a sound reproduction environment with higher sound quality.

  In step S465, when content that is in the DTS format is played back, playback with a high sampling rate and a high compression bit rate, called the DTS-HD format, is possible on the output device side, and the encoded digital signal is DTS- It is determined whether or not the HD format has been encoded, and if both are satisfied, in step S466, stream data of the DTS-HD format encoded digital signal is set to be output. On the other hand, if either is not met, in FIG. 9, the stream data of the encoded digital signal in the normal DTS format described as DTS (core) is set to be output.

  Similarly, in step S468, it is determined on the output device 20 side whether a digital signal encoded by the Dolby Digital method can be received and reproduced. If possible, in step S469, Dolby Digital Plus (abbreviated as DD +), which is a multi-channel and high compression bit rate, can be received and reproduced in the Dolby Digital method, and the encoded digital signal read from the content is It is determined whether it is encoded by the DD + system. If the condition is met, in step S470, the stream data of the encoded digital signal in the DD + system is set to be output. If the conditions do not match, it is set so that an encoded digital signal is output by AC-3 (trademark) which is a normal Dolby digital system.

  When the DTS system and the Dolby Digital system cannot be received on the output device 20 side, these are decoded by the decoding unit 6 on the playback device 10 side to be LPCM audio data. Then, it is determined in step S472 whether or not the LPCM format can be received by the multi-channel method. If reception is possible using the multi-channel method, in step S473, the decoding unit 6 performs decoding output setting in the multi-channel output format, and performs processing of the output format setting function F500.

  When the output device 20 cannot receive data by the multi-channel method, the LPCM 2-channel output setting is selected. In step S474, the output of the multi-channel playback signal is also set in a 2-channel output format by performing a downmix process (a process of synthesizing and adding a multi-channel output sound to a 2-channel output sound) to a stereo 2-channel sound. Then, the output format setting function F500 is processed.

  When there is no uncompressed encoded data and a plurality of compressed encoded data can be selected at the time of content reproduction, when both the DTS method and the Dolby digital method can be selected, a method with a high compressed encoding rate (For example, when the DTS system is 1.5 Mbit / sec and the Dolby Digital system is 448 Kbit / sec, it indicates the DTS system) so that the user can set the priority selection criteria in advance. An algorithm may be adopted.

  This means that when there is no uncompressed encoded data and a plurality of compressed encoded data can be selected at the time of content reproduction, the reproduction environment information on the output device 20 side is acquired via HDMI etc. It is intended to achieve higher sound quality playback based on criteria such as whether the rate is high, the sampling rate is high, or the number of playback channels is large.

(Embodiment 4)
A sound reproducing apparatus according to Embodiment 4 of the present invention will be described. The following description is centered on differences from the first embodiment. A configuration example is the same as that of the first embodiment, and will be described with reference to FIGS. 1, 2, and 3.

  Hereinafter, a processing routine in which the CPU unit 3 of the player 10 selects an output format output from the transmission unit 7 will be described (not shown). In this embodiment, when playing back from content, the encoding digital signal selects an arbitrary method from a plurality of compression-encoded methods (eg, DTS method and Dolby digital method) and an uncompressed encoding method. Next, the processing content when it can be reproduced will be described.

  In this case, the input unit 2 first acquires the decoded audio type information of an encoded digital signal input from the reproduction of the optical disc 1 or the like. Here, it is assumed that the data is encoded by the compression-coded DTS method and Dolby digital method and the LPCM method encoded by non-compression.

  As in the first embodiment, the device ID information on the output device 20 side is received via the HDMI interface described above and compared with the information stored in the memory unit 4. If there is matching information, the outputable format information corresponding to the device ID is read. On the other hand, if there is no matching device ID, output audio information such as an outputable format on the output machine 20 side is acquired.

  First, it is determined whether or not reception is possible using the LPCM multi-channel method. If reception is possible using the multi-channel method, the decoding unit 6 performs decoding output setting using the LPCM multi-channel output format.

  If it cannot be received by the multi-channel method, as described in Embodiment 3, it is determined whether or not there is an environment in which an encoded digital signal using the DTS method or the Dolby digital method can be decoded. If both the DTS method and the Dolby digital method are not possible, the LPCM 2-channel output setting is selected. Multi-channel in LPCM is down-mixed to stereo 2-channel audio (processing to synthesize and add multi-channel output audio to 2-channel output audio), and output is set in a 2-channel output format.

  In the case where both non-compressed encoded data and compressed encoded data can be selected at the time of content reproduction, this is done while acquiring reproduction environment information on the output device 20 side via HDMI or the like. By selecting compression and multi-channel playback, high sound quality playback is to be realized.

(Embodiment 5)
A sound reproducing apparatus according to Embodiment 5 of the present invention will be described. The following description is centered on differences from the first embodiment. A configuration example is the same as that of the first embodiment, and will be described with reference to FIGS. 1, 2, and 3.

  Hereinafter, a processing routine in which the CPU unit 3 of the player 10 selects an output format output from the transmission unit 7 will be described (not shown). In this embodiment, when reproducing from content, an arbitrary method is selected from a plurality of compression-encoded encoding methods (for example, DTS method and Dolby digital method) and a lossless compression encoding method. Next, the processing content when it can be reproduced will be described.

  In this case, the input unit 2 first acquires the decoded audio type information of an encoded digital signal input from the reproduction of the optical disc 1 or the like. Here, it is assumed that the encoding is performed by the compression-coded DTS method and the Dolby digital method and the MLP (Meridian Lossless Packing) method encoded by lossless compression. In this lossless compression, if compressed encoded information is also decoded, it is compressed by an encoding method that can be completely restored to original uncompressed encoded data. On the other hand, the DTS method and the Dolby Digital method employ a compression method that does not completely return to the original uncompressed encoded data even after decoding. This side is called lossy compression.

  As in the first embodiment, the device ID information on the output device 20 side is received via the HDMI interface described above and compared with the information stored in the memory unit 4. If there is matching information, the outputable format information corresponding to the device ID is read. On the other hand, if there is no matching device ID, output audio information such as an outputable format on the output machine 20 side is acquired.

  First, it is determined whether or not reception is possible using the multi-channel method based on MLP. If reception is possible using the multi-channel method, the decoding unit 6 performs decoding output setting using the MLP method.

  If it cannot be received by the MLP method, as described in the third embodiment, it is determined whether or not there is an environment in which an encoded digital signal using the DTS method or the Dolby digital method can be decoded. If both the DTS method and the Dolby digital method are not possible, the LPCM 2-channel output setting is selected. Multi-channel decoded for each encoding method is downmixed to stereo 2-channel audio (processing to add and output multi-channel output audio to 2-channel output audio) and set to output in 2-channel output format .

  This is because, in the case where both lossy encoded data and lossless encoded data can be selected at the time of content reproduction, the reproduction environment information on the output device 20 side is acquired via HDMI or the like. Thus, high-quality sound reproduction is realized by selecting reversible compression reproduction and multi-channel reproduction.

  In the above description, the method information that can be decoded on the output device 20 side is acquired on the playback device 10 side, decoded in a format that can be played on the output device 20 side, output processing is set, and output. In addition, a method for realizing a process for performing selection so that an environment for reproduction with as high a sound quality as possible has been described.

  However, when the bandwidth of the transmission path transmitted from the player 10 side to the output device 20 side is small, it is more convenient to output in the state of an encoded stream instead of the LPCM. In such a case, first, it is determined whether or not the signal in the encoded stream that is the encoded digital signal can be received and reproduced, and then, the LPCM transfer with two channels is selected. Good. In other words, when there is a limitation on the transmission band, priority is given to audio data that has been compressed and encoded and has a low compression bit rate, so that the amount of audio information can be transmitted as little as possible. This can contribute to a reduction in traffic of data transmission.

  Examples of applications on the playback means side of this audio playback device include STBs, digital satellite broadcast receivers and their recording equipment, related equipment for DVD players or DVD recorders, other equipment having playback functions such as optical disks, hard disk recorders, personal computers Examples of applications on the audio output means side include AV amplifiers, television monitors and projectors having video and audio outputs, personal computers, and the like.

  The audio reproduction device according to the present invention has a digital interface between the reproduction device side and the output device side, and is based on a plurality of encoded audio signals digitally encoded on the reproduction device side based on the audio reproduction condition on the output device side. In order to reproduce, it can use for the use of selecting the encoding audio | voice signal matched with the output possible condition of an output device.

The figure which shows the structure of the audio | voice reproduction apparatus in Embodiment 1 of this invention. The figure which shows the decoding structure of the video and audio in Embodiment 1 of this invention Diagram showing the configuration of HDMI The figure which shows the example of a structure of the apparatus specific information in Embodiment 1 of this invention The figure which shows the example of another structure of the apparatus specific information in Embodiment 1 of this invention The figure which shows the example of a structure of the content recorded on the memory in Embodiment 1 of this invention The figure which shows the decoding process in the case of reproducing | regenerating the digital broadcast in Embodiment 1 of this invention The figure which shows the decoding process in the case of reproducing | regenerating the digital broadcast in Embodiment 2 of this invention The figure which shows the decoding process in the case of reproducing | regenerating the disk medium in Embodiment 3 of this invention The figure which shows the sampling audio | voice process in Embodiment 1 of this invention The figure which shows the structure of the audio equipment of a prior art example The figure which shows the structure of the video receiving system of a prior art example

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 Disc reproduction | regeneration part 2 Antenna part 3 CPU part 4 Memory part 5 Input part 6 Decoding part 7 Transmission part 8 Information acquisition part 10 Playing machine 11 Equipment specific information part 12 Reception part 13 Output part 20 Output machine

Claims (10)

Information acquisition means for acquiring information for specifying the audio reproduction condition from the connected device, control means for setting the reproduction condition of the audio data based on the information acquired by the information acquisition means,
Playback means for playing back audio data according to the playback conditions set by the control means;
Transmitting means that, when the sampling frequency of the information acquired by the information acquiring means is higher than the sampling frequency of audio data, transmits the audio data reproduced by the reproducing means to the device after performing upsampling processing An audio playback device comprising: 2. The audio reproducing apparatus according to claim 1, wherein the reproduction condition set by the control means is a decoded audio type. 2. The audio reproduction device according to claim 1, wherein the reproduction condition set by the control means is the number of audio channels. 2. The audio reproduction device according to claim 1, wherein the reproduction condition set by the control means is a sampling frequency. The apparatus further comprises a storage means for recording the audio playback condition of the device acquired by the information acquisition means and the device information for identifying the device, and compares the device information recorded in the storage means when connected to the device. Accordingly, when the information matches the previously connected device information, the reproduction means reproduces the sound reproduction condition recorded in the storage means. . The audio reproduction apparatus according to claim 1, wherein the reproduction unit further includes an encoding conversion unit that reproduces data by changing an encoding method. The audio reproducing apparatus according to claim 2, wherein the reproducing means preferentially outputs audio data having a high compression bit rate number that is compression-encoded from content. 4. The audio reproduction apparatus according to claim 3, wherein the reproduction means preferentially outputs audio data having a large amount of audio reproduction channel information from the content. 3. The audio reproducing apparatus according to claim 2, wherein the reproducing means preferentially outputs audio data in which an audio reproducing condition is lossless compression encoded. 4. The audio reproduction apparatus according to claim 3, wherein the reproduction unit preferentially outputs audio data having a high audio reproduction sampling frequency from the content.

Images