CN107507617A - A kind of system and method realized DSD audios and solved firmly - Google Patents

A kind of system and method realized DSD audios and solved firmly Download PDF

Info

Publication number
CN107507617A
CN107507617A CN201710698979.8A CN201710698979A CN107507617A CN 107507617 A CN107507617 A CN 107507617A CN 201710698979 A CN201710698979 A CN 201710698979A CN 107507617 A CN107507617 A CN 107507617A
Authority
CN
China
Prior art keywords
data
dsd
forms
voice data
converter module
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Granted
Application number
CN201710698979.8A
Other languages
Chinese (zh)
Other versions
CN107507617B (en
Inventor
戴建成
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Shenzhen Ling Ling Digital Technology Development Co Ltd
Original Assignee
Shenzhen Ling Ling Digital Technology Development Co Ltd
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Shenzhen Ling Ling Digital Technology Development Co Ltd filed Critical Shenzhen Ling Ling Digital Technology Development Co Ltd
Priority to CN201710698979.8A priority Critical patent/CN107507617B/en
Publication of CN107507617A publication Critical patent/CN107507617A/en
Application granted granted Critical
Publication of CN107507617B publication Critical patent/CN107507617B/en
Active legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Classifications

    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16Vocoder architecture
    • G10L19/167Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16Vocoder architecture
    • G10L19/173Transcoding, i.e. converting between two coded representations avoiding cascaded coding-decoding
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16Vocoder architecture
    • G10L19/18Vocoders using multiple modes
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16Vocoder architecture
    • G10L19/18Vocoders using multiple modes
    • G10L19/22Mode decision, i.e. based on audio signal content versus external parameters

Landscapes

  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Signal Processing For Digital Recording And Reproducing (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Abstract

The present invention discloses a kind of system and method realized DSD audios and solved firmly, and the system includes:Application processor module is used to obtain and parse original audio data, and the data parameters in data bus address and analysis result and voice data are sent into audio processing modules by data/address bus;Whether data parameters include data sampling rate, digit, code stream, are that DSD forms, DSD forms, rear end D/A converter module need the mode supported;Audio processing modules read data parameters and voice data according to the data bus address, carry out decoding process to the voice data according to data parameters, and decoded voice data is sent into D/A converter module;D/A converter module, for decoded voice data to be converted into simulated audio signal output;Asynchronous double crystal oscillator clocks are used to export clock signal to audio processing modules.The present invention, which realizes, supports the DSD audios of all specifications to play, and realizes the accurate broadcasting of fully sampled rate, reaches good sound effect.

Description

A kind of system and method realized DSD audios and solved firmly
Technical field
The present invention relates to audio decoding techniques field, more particularly to a kind of system and method realized DSD audios and solved firmly.
Background technology
With the rapid development of electronic information technology, music is inseparable in the life of people, and people want to tonequality Seek more and more higher.Currently used method for playing music is that various audio formats are decoded into PCM (Pulse-Code Modulation, pulse code modulation) play out, but in order to obtain more preferable sound quality, Sony and Philips exist The audio of DSD (Direct Stream Digital, direct bit streaming digital) form is developed within 1996, based on PDM (pulse-density modulation) pulse density modulated is realized, simulated audio signal, and PDM are represented by density The precision sampled every time is all 1bit, therefore can provide more outstanding sound effect, and DSD has minimum quantizing noise, is surpassed High signal to noise ratio, therefore quality is more stable.
Realizing the method for DSD audios in the prior art has three kinds:A kind of is to realize DSD audios by USB xmos modes Play, but xmos power consumptions are big, volume is big, cost is high, are not suitable for portable HiFi players and use, and need to pass through USB Interface realizes DSD transmission, and many MCU USB interface is few, and versatility is not strong;Second of I2S (Inter- using standard IC Sound) bus (also known as integrated circuit built-in audio bus) transmission voice data, its DSD audio decoded by processor Digital analog converter is sent into PCM data to play out, because being converted to PCM data, therefore can not give play to direct bit The advantage of streaming digital data, and decoded using MCU, it is high to MCU performance requirements, and can only typically up to support DSD The decoding of 128 forms, DSD256 and DSD512 are not decoded, and can not play primary DSD audios;The third is to pass through DSD data, are carried out packing encapsulation, the PCM data for the I2S standards that disguise oneself as is passed by DoP (DSD overPCM) mode in MCU Defeated, then DAC further decodings in rear end go out DSD data and played out, although the form that such a mode does not pass through DSD to PCM is changed, But because MCU does not support the data transfer of the DSD audio streams higher than sample rate (2.8224MHz), therefore can only support DSD64, sample rate are 2.8224MHz and DSD128, and sample rate is 2.8224MHz 32bit DSD audio streaming, it is impossible to is propped up It is the higher DSD musics that 11.2896MHz and DSD512 sample rates are 22.5792MHz to hold DSD256 sample rates, and also not Primary (Native DSD) patterns of DSD can be supported, it can thus be appreciated that such a method supports that DSD forms are not complete, it is impossible to support higher rule The DSD audios of lattice play and Native DSD patterns.
The content of the invention
The technical problems to be solved by the invention, which are the provision of, a kind of supports the DSD audios of all specifications to play, prop up Hold the system and method realized DSD audios and solved firmly of fully sampled rate.
In order to solve the above technical problems, the present invention uses following technical scheme:
On the one hand, there is provided a kind of system realized DSD audios and solved firmly, the system include application processor module, passed through Audio processing modules that data/address bus is connected with the application processing module, D/A converter module and be the audio processing modules Asynchronous double crystal oscillator clocks of clock signal are provided;
The application processor module is used to obtain and parse original audio data, by data bus address and analysis result In data parameters and voice data be sent to audio processing modules;The data parameters include data sampling rate, digit, code Stream, whether it is that DSD forms, DSD forms, rear end D/A converter module need the mode supported, the voice data is DSD forms DSD native datas or the PCM data of non-DSD forms that decodes of the application processor module;
Audio processing modules read the data parameters and voice data according to the data bus address, according to the number Decoding process is carried out to the voice data according to parameter, and decoded voice data is sent to D/A converter module;
D/A converter module, the decoded voice data for audio processing modules to be sent are converted into analogue audio frequency letter Number output.
Wherein, the parsing original audio data includes:If the original audio data is the audio number of non-DSD forms According to the original audio data being then decoded into PCM data, and parse the sample rate, digit and code stream of the PCM data; If the original audio data is the voice data of DSD forms, i.e. DSD native datas, then the DSD native datas are parsed DSD forms.
Wherein, it is described that decoding process is carried out to the voice data according to the data parameters, and by decoded audio Data, which are sent to D/A converter module, to be included:
If whether in the data parameters is that DSD forms are non-DSD forms, i.e. voice data is PCM data, then by institute State PCM data and be sent to D/A converter module, sample rate, digit and code stream logarithmic mode modular converter in data parameters It is configured;
If whether in the data parameters is that DSD forms are DSD forms, i.e., voice data is DSD native datas, if after The mode that end D/A converter module needs to support is the primary modes of DSD, then DSD native datas is transferred to digital-to-analogue conversion by bit Module;If the mode that rear end D/A converter module needs to support is DoP modes, DSD native datas are encapsulated according to DoP agreements Into DoP data, the DoP data are made up of data parameters+DSD native datas, give the DoP data outputs to digital-to-analogue conversion mould Block;The mode that rear end D/A converter module needs to support is PCM modes, then DSD native datas according to DSD coded system algorithms PCM data is decoded, the PCM data is converted into corresponding sample rate, digit, code stream by the DSD forms in data parameters, The PCM data, sample rate, digit and code stream are sent to D/A converter module, and according to the sample rate and code stream logarithmic mode Modular converter is configured.
Wherein, the audio processing modules be additionally operable to be sent to the PCM data of D/A converter module, DSD native datas, Or DoP data conversions export into spdif forms.
Wherein, asynchronous double crystal oscillator clocks include 22.5792MHz the first crystal oscillator and 24.576MHz the second crystal oscillator; Or the first crystal oscillator and 49.152MHz the second crystal oscillator including 45.1584MHz.
Wherein, the audio processing modules are realized based on FPGA;Application processor module passes through high speed data bus and sound Frequency processing module connects.
Wherein, the data parameters and voice data by data bus address and analysis result are sent to audio frequency process Module specifically includes:
The analysis result of the original audio data is packaged according to customized data bus transmission protocol format, The data bus transmission protocol format is:Data bus address+data parameters+voice data;
A group data for bag are sent to audio processing modules according to data bus transmission agreement and data bus address;
The audio processing modules read the data parameters and voice data according to the data bus address to be included:Sound Frequency processing module reads the data of described group of bag according to the data bus address, according to data bus transmission protocol analysis The data of group bag.
On the other hand a kind of method realized DSD audios and solved firmly is provided, this method includes:
Application processor module obtains and parses original audio data, by the data in data bus address and analysis result Parameter and voice data are sent to audio processing modules;The data parameters include data sampling rate, digit, code stream, whether be DSD forms, DSD forms, rear end D/A converter module need the mode supported, the voice data is that the DSD of DSD forms is primary The PCM data for the non-DSD forms that data or the application processor module decode;
Audio processing modules read the data parameters and voice data according to the data bus address, according to the number Decoding process is carried out to the voice data according to parameter, and decoded voice data is sent to D/A converter module;
It is defeated that the decoded voice data that D/A converter module sends audio processing modules is converted into simulated audio signal Go out;
Wherein, the application processor module is connected by data/address bus with audio processing modules;Audio processing modules Clock signal is provided by asynchronous double crystal oscillator clocks.
Wherein, the parsing original audio data includes:If the original audio data is the audio number of non-DSD forms According to the original audio data being then decoded into PCM data, and parse the sample rate, digit and code stream of the PCM data; If the original audio data is the voice data of DSD forms, i.e. DSD native datas, then the DSD native datas are parsed DSD forms.
Wherein, it is described that decoding process is carried out to the voice data according to the data parameters, and by decoded audio Data, which are sent to D/A converter module, to be included:
If whether in the data parameters is that DSD forms are non-DSD forms, i.e. voice data is PCM data, then by institute State PCM data and be sent to D/A converter module, sample rate, digit and code stream logarithmic mode modular converter in data parameters It is configured;
If whether in the data parameters is that DSD forms are DSD forms, i.e., voice data is DSD native datas, if after The mode that end D/A converter module needs to support is the primary modes of DSD, then DSD native datas is transferred to digital-to-analogue conversion by bit Module;If the mode that rear end D/A converter module needs to support is DoP modes, DSD native datas are encapsulated according to DoP agreements Into DoP data, the DoP data are made up of data parameters+DSD native datas, give the DoP data outputs to digital-to-analogue conversion mould Block;The mode that rear end D/A converter module needs to support is PCM modes, then DSD native datas according to DSD coded system algorithms PCM data is decoded, the PCM data is converted into corresponding sample rate, digit, code stream by the DSD forms in data parameters, The PCM data, sample rate, digit and code stream are sent to D/A converter module, and according to the sample rate and code stream logarithmic mode Modular converter is configured.
Compared with prior art, beneficial effects of the present invention are:The system provided by the invention for realizing that DSD audios solve firmly is led to Cross asynchronous double crystal oscillator clocks and provide clock signal to processing module, realize the support of fully sampled rate, clock is more accurate, and effect is more It is good, and using the transmission of data/address bus progress voice data, the advantage of the direct bit streaming digital datas of DSD has been given play to, in fact The DSD audios for having showed all specifications of support play, and realize the accurate broadcasting of fully sampled rate, have reached good sound effect.
Brief description of the drawings
Technical scheme in order to illustrate the embodiments of the present invention more clearly, institute in being described below to the embodiment of the present invention The accompanying drawing needed to use is briefly described, it should be apparent that, drawings in the following description are only some implementations of the present invention Example, for those of ordinary skill in the art, on the premise of not paying creative work, it can also be implemented according to the present invention The content of example and these accompanying drawings obtain other accompanying drawings.
Fig. 1 is a kind of knot of the embodiment of the system realized DSD audios and solved firmly provided in the specific embodiment of the invention Structure block diagram.
Fig. 2 is a kind of side of the embodiment of the method realized DSD audios and solved firmly provided in the specific embodiment of the invention Method flow chart.
Embodiment
For make present invention solves the technical problem that, the technical scheme that uses and the technique effect that reaches it is clearer, below The technical scheme of the embodiment of the present invention will be described in further detail with reference to accompanying drawing, it is clear that described embodiment is only It is part of the embodiment of the present invention, rather than whole embodiments.Based on the embodiment in the present invention, those skilled in the art exist The every other embodiment obtained under the premise of creative work is not made, belongs to the scope of protection of the invention.
A kind of system for realizing that DSD audios solve firmly provided in an embodiment of the present invention is made with reference to Fig. 1 further detailed Thin description.Fig. 1 is refer to, it is a kind of reality of the system realized DSD audios and solved firmly provided in the specific embodiment of the invention Apply the block diagram of example.As shown in figure 1, the system includes in certain embodiments:Application processor module 20, pass through data Audio processing modules 10 that bus is connected with the application processing module 20, D/A converter module 40 and be the audio frequency process mould Block 10 provides asynchronous double crystal oscillator clocks 30 of clock signal.Application processor module 20 is used to obtain and parse original audio number According to the data parameters in data bus address and analysis result and voice data are sent into audio processing modules 10;The number Include data sampling rate, digit according to parameter, code stream, whether be that DSD forms, DSD forms, rear end D/A converter module need to support Mode, the voice data for DSD forms primary (Native DSD) data of DSD or the application processor module decoding The PCM data of the non-DSD forms gone out.Audio processing modules 10 read the data parameters and sound according to the data bus address Frequency evidence, decoding process is carried out to the voice data according to the data parameters, and decoded voice data is sent to D/A converter module 40.D/A converter module 40, for the decoded voice data conversion for sending audio processing modules 10 Exported into simulated audio signal, D/A converter module 40 outputs analog signal to power amplifier and played out.Asynchronous double crystal oscillator clocks 30 are used to export clock signal to audio processing modules 10, and audio processing modules 10 are according to the clock signal gathered data.Wherein, Application processor module 20 is connected by data/address bus and audio processing modules 10, data/address bus includes but be not limited to I2S buses, I2C buses, spi bus, pci bus, high speed GPIO data/address bus, CAN or the contour data of general-purpose serial bus USB are total Line and normal data line.
The system provided by the invention for realizing that DSD audios solve firmly provides clock by asynchronous double crystal oscillator clocks to processing module Signal, the support of fully sampled rate is realized, clock is more accurate, better, and carries out voice data using data/address bus Transmission, has given play to the advantage of the direct bit streaming digital datas of DSD, has realized and support the DSD audios of all specifications to play, and realizes The accurate broadcasting of fully sampled rate, has reached good sound effect.
In certain embodiments, the parsing original audio data includes:If the original audio data is non-DSD forms Voice data, then the original audio data is decoded into PCM data, and parse the sample rate of the PCM data, digit And code stream;If the original audio data is the voice data of DSD forms, i.e. DSD native datas, then it is former to parse the DSD The DSD forms of raw data.
In certain embodiments, it is described that decoding process is carried out to the voice data according to the data parameters, and will solution Voice data after code, which is sent to D/A converter module, to be included:
If whether in the data parameters is that DSD forms are non-DSD forms, i.e. voice data is PCM data, then by institute State PCM data and be sent to D/A converter module, sample rate, digit and code stream logarithmic mode modular converter in data parameters It is configured;
If whether in the data parameters is that DSD forms are DSD forms, i.e., voice data is DSD native datas, if after The mode that end D/A converter module needs to support is the primary modes of DSD, then DSD native datas is transferred to digital-to-analogue conversion by bit Module;If the mode that rear end D/A converter module needs to support is DoP modes, DSD native datas are encapsulated according to DoP agreements Into DoP data, the DoP data are made up of data parameters+DSD native datas, give the DoP data outputs to digital-to-analogue conversion mould Block;The mode that rear end D/A converter module needs to support is PCM modes, then DSD native datas according to DSD coded system algorithms PCM data is decoded, the PCM data is converted into corresponding sample rate, digit, code stream by the DSD forms in data parameters, The PCM data, sample rate, digit and code stream are sent to D/A converter module, and according to the sample rate and code stream logarithmic mode Modular converter is configured.It follows that no matter DSD forms are PCM modes, DoP modes or the primary modes of DSD, the system It can play, return to the advantage of the processing direct bit streaming digital datas of DSD, more preferable sound quality can be obtained.In some realities Apply in example, audio processing modules are additionally operable to that the PCM data, DSD native datas or DoP data of D/A converter module will be sent to It is converted into spdif forms and is exported by spdif interfaces 50.
In certain embodiments, the data parameters and voice data by data bus address and analysis result are sent Specifically included to audio processing modules:Solution according to customized data bus transmission protocol format to the original audio data Analysis result packages, and the data bus transmission protocol format is:Data bus address+data parameters+voice data;Group The data of bag send audio processing modules 10 to according to data bus transmission agreement and data bus address, data/address bus Location, data parameters and voice data first organize bag and are then forwarded to the progress decoding process of audio processing modules 10, can be not easy data Lose, voice data is actual voice data, including the DSD native datas of DSD forms or the application processor module solution The PCM data for the non-DSD forms that code goes out.In certain embodiments, the audio processing modules 10 are according to the data/address bus The data parameters and voice data are read in location to be included:Audio processing modules 10 read described group according to the data bus address The data of bag, the data of bag are organized according to data bus transmission protocol analysis.Audio processing modules 10 receive application processor The data and data bus address for the described group of bag that module 20 is sent, the number of described group of bag is read according to the data bus address According to organizing the data of bag according to data bus transmission protocol analysis, the parameter in data parameters is to the audio number According to progress decoding process, and decoded voice data is sent to D/A converter module 40.After D/A converter module 40 decodes Voice data be converted into simulated audio signal and export to play out to power amplifier.
Wherein, DSD forms include DSD64, DSD128, DSD256 and DSD512, and rear end rear end D/A converter module needs The mode of support includes DSD modes, DoP modes and PCM modes, and data bus transmission agreement is customized, can change and Adjustment, very flexible and extension, its data transmitted not only supports PCM data also to support DSD native datas, therefore reaches The DSD of full format is supported to solve purpose firmly.The system provided by the invention for realizing that DSD audios solve firmly passes through asynchronous double crystal oscillator clocks Clock signal is provided to audio processing modules, and data bus transmission protocol format is:Data bus address+data parameters+sound Frequency evidence, the support of fully sampled rate is realized, clock is more accurate, better, and carries out voice data using data/address bus Transmission, has given play to the advantage of the direct bit streaming digital datas of DSD, has realized and support the DSD audios of all specifications to play, and realizes The accurate broadcasting of fully sampled rate, has reached good sound effect.
In certain embodiments, the data bus transmission protocol format is specially:8bit data bus address+8bit numbers According to parameter+voice data;Wherein, 8bit data bus address is the data/address bus physical address of 16 systems, to application Reason device module writes data;The parameter of 8bit data parameters includes:The sample rates of data, digit, code stream, whether be DSD forms, DSD forms, rear end D/A converter module need the mode supported;Voice data is actual voice data, including DSD forms The PCM data for the non-DSD forms that DSD native datas or the application processor module decode.In certain embodiments, it is described Asynchronous double crystal oscillator clocks 30 include 22.5792MHz the first crystal oscillator 301 and 24.576MHz the second crystal oscillator 302;Or including 45.1584MHz the first crystal oscillator 301 and 49.152MHz the second crystal oscillator 302.Wherein, 22.5792MHz and 45.1584MHz First crystal oscillator 301 is used for the audio for sampling 44.1KHz, 88.2KHz, 176.4KHz and 352.8KHz audio format, 24.576MHz and 49.152MHz the second crystal oscillator 302 is used for the audio lattice for sampling 48KHz, 96KHz, 192KHz and 384KHz The audio of formula, it is achieved thereby that the support to fully sampled rate, clock is more accurate, better.
In certain embodiments, audio processing modules (Field-Programmable Gate Array, are showed based on FPGA Field programmable gate array) realize, decoding process is carried out to voice data by FPGA audio processing modules, cost is low, the speed of service It hurry up, power consumption is lower, size is small.In certain embodiments, application processor module 20 passes through high speed data bus and audio frequency process Module 10 is connected, and audio processing modules 10 are connected by I2S buses with D/A converter module 40.Using data/address bus DSD's Each format data transmission parses data bus transmission agreement by audio processing modules 10 and data is carried out to audio processing modules 10 Processing, cost is low, and size is small, and power consumption is lower, avoids the problem of cost of other manner is high, and size is big, and power consumption is big.
The system provided in an embodiment of the present invention for realizing that DSD audios solve firmly, which can be applicable to, to be required to tonequality, it is necessary to support In all products for playing DSD forms, especially portable HiFi players product, including but not limited to portable player, On the products such as desk-top HiFi players, digital broadcasting product, music fever mobile phone.
The system provided in an embodiment of the present invention for realizing that DSD audios solve firmly realize DSD64, DSD128, DSD256, The music of DSD512 high code stream DSD forms, supports all DSD modes, including PCM modes, DoP modes and DSD primary The broadcasting of mode, the advantage of the direct bit streaming digital datas of DSD is given play to, more preferable sound quality has been obtained, has reached more Outstanding sound effect improves;And by self-defining data bus transfer agreement, it is not high to application processor performance requirement, therefore The application processor or host computer of main flow are all suitable for, versatility and flexibility are stronger, more practical, while realize simple, agreement Can flexibly it change, application scalability is stronger;DSD each format data transmission is given using data/address bus the sound realized based on FPGA Frequency processing module, data are handled by audio processing modules parsing data bus transmission agreement, cost is low, and size is small, work( Consumption is lower, avoids the problem of cost of other manner is high, and size is big, and power consumption is big;Using asynchronous double crystal oscillator clocks, realize pair 32Khz, 44.1KHz, 88.2KHz, 176.4KHz, 352.8KHz and 48KHz, 96KHz, 192KHz, 384KHz audio are adopted entirely The support of sample rate, clock is more accurate, and effect more, and not only supports the data playback of DSD forms, also supports other PCM lattice The transmission and broadcasting of formula, reach best effect.
Fig. 2 is refer to, it is a kind of reality of the method realized DSD audios and solved firmly provided in the specific embodiment of the invention Apply the method flow diagram of example.The embodiment of method is what the embodiment based on said system was realized, not detailed in embodiment of the method Most content refer to the embodiment of said system.As shown in Fig. 2 the method comprising the steps of S101~step S103, it is specific in Hold as follows:
Step S101:Application processor module is obtained and parses original audio data, and data bus address and parsing are tied Data parameters and voice data in fruit are sent to audio processing modules;The data parameters include data sampling rate, digit, code Stream, whether it is that DSD forms, DSD forms, rear end D/A converter module need the mode supported, the voice data is DSD forms DSD native datas or the PCM data of non-DSD forms that decodes of the application processor module.
In certain embodiments, the parsing original audio data includes:If the original audio data is non-DSD forms Voice data, then the original audio data is decoded into PCM data, and parse the sample rate of the PCM data, digit And code stream;If the original audio data is the voice data of DSD forms, i.e. DSD native datas, then it is former to parse the DSD The DSD forms of raw data.
In certain embodiments, the data parameters and voice data by data bus address and analysis result are sent Specifically included to audio processing modules:Solution according to customized data bus transmission protocol format to the original audio data Analysis result packages, and the data bus transmission protocol format is:Data bus address+data parameters+voice data;Group The data of bag send audio processing modules to according to data bus transmission agreement and data bus address.In certain embodiments, Data bus transmission protocol format is specially:8bit data bus address+8bit data parameters+voice data;Wherein, 8bit numbers According to the data/address bus physical address that bus address is 16 systems, for writing data to application processor module;8bit data parameters Parameter include:The sample rates of data, digit, code stream, whether it is DSD forms, DSD forms, rear end D/A converter module needs The mode of support;Voice data is actual voice data, including the DSD native datas of DSD forms or the application processor The PCM data for the non-DSD forms that module decodes.In certain embodiments, asynchronous double crystal oscillator clocks include 22.5792MHz the first crystal oscillator and 24.576MHz the second crystal oscillator;Or the first crystal oscillator including 45.1584MHz and 49.152MHz the second crystal oscillator.Wherein, 22.5792MHz and 45.1584MHz the first crystal oscillator be used for sample 44.1KHz, The audio of 88.2KHz, 176.4KHz and 352.8KHz audio format, 24.576MHz and 49.152MHz the second crystal oscillator are used In the audio of sampling 48KHz, 96KHz, 192KHz and 384KHz audio format, it is achieved thereby that the support to fully sampled rate, Clock is more accurate, better.
Step S102:Audio processing modules read the data parameters and voice data according to the data bus address, Decoding process is carried out to the voice data according to the data parameters, and decoded voice data is sent to digital-to-analogue conversion Module;Wherein, the clock signal of audio processing modules is provided by asynchronous double crystal oscillator clocks.
In certain embodiments, the audio processing modules according to the data bus address read the data parameters and Voice data includes:Audio processing modules read the data of described group of bag according to the data bus address, according to data/address bus Host-host protocol parses the data of described group of bag.
In certain embodiments, it is described that decoding process is carried out to the voice data according to the data parameters, and will solution Voice data after code, which is sent to D/A converter module, to be included:
If whether in the data parameters is that DSD forms are non-DSD forms, i.e. voice data is PCM data, then by institute State PCM data and be sent to D/A converter module, sample rate, digit and code stream logarithmic mode modular converter in data parameters It is configured;
If whether in the data parameters is that DSD forms are DSD forms, i.e., voice data is DSD native datas, if after The mode that end D/A converter module needs to support is the primary modes of DSD, then DSD native datas is transferred to digital-to-analogue conversion by bit Module;If the mode that rear end D/A converter module needs to support is DoP modes, DSD native datas are encapsulated according to DoP agreements Into DoP data, the DoP data are made up of data parameters+DSD native datas, give the DoP data outputs to digital-to-analogue conversion mould Block;The mode that rear end D/A converter module needs to support is PCM modes, then DSD native datas according to DSD coded system algorithms PCM data is decoded, the PCM data is converted into corresponding sample rate, digit, code stream by the DSD forms in data parameters, The PCM data, sample rate, digit and code stream are sent to D/A converter module, and according to the sample rate and code stream logarithmic mode Modular converter is configured.
In certain embodiments, audio processing modules (Field-Programmable Gate Array, are showed based on FPGA Field programmable gate array) realize, decoding process is carried out to voice data by FPGA audio processing modules, cost is low, the speed of service It hurry up, power consumption is lower, size is small.
Step S103:The decoded voice data that D/A converter module sends audio processing modules is converted into analog audio Simulated audio signal is exported and played out to power amplifier by frequency signal output, D/A converter module.
In the present embodiment, application processor module is connected by data/address bus with audio processing modules;Audio processing modules Clock signal provided by asynchronous double crystal oscillator clocks.As a preferred embodiment, application processor module passes through high speed number It is connected according to bus with audio processing modules, audio processing modules are connected by I2S buses with D/A converter module.It is total using data Bundle of lines DSD each format data transmission parses data bus transmission agreement logarithm to audio processing modules by audio processing modules According to being handled, cost is low, and size is small, and power consumption is lower.
In summary, the present embodiment provide realize method that DSD audios solve firmly by asynchronous double crystal oscillator clocks to audio at Manage module and clock signal is provided, realize the support of fully sampled rate, clock is more accurate, better, and is entered using data/address bus The transmission of row voice data, the advantage of the direct bit streaming digital datas of DSD is given play to, realized the DSD for supporting all specifications Audio plays, and realizes the accurate broadcasting of fully sampled rate, has reached good sound effect.
The technical principle of the present invention is described above in association with specific embodiment.These descriptions are intended merely to explain the present invention's Principle, and limiting the scope of the invention can not be construed in any way.Based on explanation herein, the technology of this area Personnel would not require any inventive effort the other embodiments that can associate the present invention, and these modes are fallen within Within protection scope of the present invention.

Claims (10)

1. a kind of system realized DSD audios and solved firmly, it is characterised in that the system includes application processor module, passes through number The audio processing modules that are connected according to bus with the application processing module, D/A converter module and carried for the audio processing modules For asynchronous double crystal oscillator clocks of clock signal;
The application processor module is used to obtain and parse original audio data, by data bus address and analysis result Data parameters and voice data are sent to audio processing modules;The data parameters include data sampling rate, digit, code stream, are No is that DSD forms, DSD forms, rear end D/A converter module need the mode supported, and the voice data is the DSD of DSD forms The PCM data for the non-DSD forms that native data or the application processor module decode;
Audio processing modules read the data parameters and voice data according to the data bus address, are joined according to the data It is several that decoding process is carried out to the voice data, and decoded voice data is sent to D/A converter module;
D/A converter module, it is defeated that the decoded voice data for audio processing modules to be sent is converted into simulated audio signal Go out.
A kind of 2. system realized DSD audios and solved firmly according to claim 1, it is characterised in that the original sound of parsing Frequency evidence includes:If the original audio data is the voice data of non-DSD forms, the original audio data is decoded into PCM data, and parse the sample rate, digit and code stream of the PCM data;If the original audio data is DSD forms Voice data, i.e. DSD native datas, then parse the DSD forms of the DSD native datas.
3. a kind of system realized DSD audios and solved firmly according to claim 2, it is characterised in that described according to the number Decoding process is carried out to the voice data according to parameter, and decoded voice data is sent into D/A converter module to include:
If whether in the data parameters is that DSD forms are non-DSD forms, i.e. voice data is PCM data, then by described in PCM data is sent to D/A converter module, and the sample rate, digit and code stream logarithmic mode modular converter in data parameters are entered Row is set;
If whether in the data parameters is that DSD forms are DSD forms, i.e. voice data is DSD native datas, if rear end number The mode that mould modular converter needs to support is the primary modes of DSD, then DSD native datas is transferred to D/A converter module by bit; If the mode that rear end D/A converter module needs to support is DoP modes, DSD native datas are packaged into DoP according to DoP agreements Data, the DoP data are made up of data parameters+DSD native datas, by the DoP data outputs to D/A converter module;Afterwards The mode that end D/A converter module needs to support is PCM modes, then DSD native datas is decoded according to DSD coded systems algorithm Go out PCM data, the PCM data is converted into corresponding sample rate, digit, code stream by the DSD forms in data parameters, by this PCM data, sample rate, digit and code stream are sent to D/A converter module, and are changed according to the sample rate and code stream logarithmic mode Module is configured.
A kind of 4. system realized DSD audios and solved firmly according to claim 3, it is characterised in that the audio frequency process mould Block is additionally operable to the PCM data, DSD native datas or DoP data conversions for being sent to D/A converter module is defeated into spdif forms Go out.
5. a kind of system realized DSD audios and solved firmly according to claim 1, it is characterised in that the asynchronous twin crystal shakes Clock includes 22.5792MHz the first crystal oscillator and 24.576MHz the second crystal oscillator;Or the first crystal oscillator including 45.1584MHz With 49.152MHz the second crystal oscillator.
A kind of 6. system realized DSD audios and solved firmly according to claim 1, it is characterised in that the audio frequency process mould Block is realized based on FPGA;Application processor module is connected by high speed data bus with audio processing modules.
A kind of 7. system realized DSD audios and solved firmly according to claim 1, it is characterised in that:
It is specific that the data parameters and voice data by data bus address and analysis result are sent to audio processing modules Including:
The analysis result of the original audio data is packaged according to customized data bus transmission protocol format, it is described Data bus transmission protocol format is:Data bus address+data parameters+voice data;
A group data for bag are sent to audio processing modules according to data bus transmission agreement and data bus address;
The audio processing modules read the data parameters and voice data according to the data bus address to be included:At audio The data that module reads described group of bag according to the data bus address are managed, bag is organized according to data bus transmission protocol analysis Data.
A kind of 8. method realized DSD audios and solved firmly, it is characterised in that methods described includes:
Application processor module obtains and parses original audio data, by the data parameters in data bus address and analysis result And voice data is sent to audio processing modules;Whether the data parameters include data sampling rate, digit, code stream, are DSD lattice Formula, DSD forms, rear end D/A converter module need the mode supported, the voice data is the DSD native datas of DSD forms Or the PCM data of non-DSD forms that the application processor module decodes;
Audio processing modules read the data parameters and voice data according to the data bus address, are joined according to the data It is several that decoding process is carried out to the voice data, and decoded voice data is sent to D/A converter module;
The decoded voice data that D/A converter module sends audio processing modules is converted into simulated audio signal output;
Wherein, the application processor module is connected by data/address bus with audio processing modules;The clock of audio processing modules Signal is provided by asynchronous double crystal oscillator clocks.
A kind of 9. method realized DSD audios and solved firmly according to claim 8, it is characterised in that the original sound of parsing Frequency evidence includes:If the original audio data is the voice data of non-DSD forms, the original audio data is decoded into PCM data, and parse the sample rate, digit and code stream of the PCM data;If the original audio data is DSD forms Voice data, i.e. DSD native datas, then parse the DSD forms of the DSD native datas.
10. a kind of method realized DSD audios and solved firmly according to right wants 9, it is characterised in that described according to the data Parameter carries out decoding process to the voice data, and decoded voice data is sent into D/A converter module included:
If whether in the data parameters is that DSD forms are non-DSD forms, i.e. voice data is PCM data, then by described in PCM data is sent to D/A converter module, and the sample rate, digit and code stream logarithmic mode modular converter in data parameters are entered Row is set;
If whether in the data parameters is that DSD forms are DSD forms, i.e. voice data is DSD native datas, if rear end number The mode that mould modular converter needs to support is the primary modes of DSD, then DSD native datas is transferred to D/A converter module by bit; If the mode that rear end D/A converter module needs to support is DoP modes, DSD native datas are packaged into DoP according to DoP agreements Data, the DoP data are made up of data parameters+DSD native datas, by the DoP data outputs to D/A converter module;Afterwards The mode that end D/A converter module needs to support is PCM modes, then DSD native datas is decoded according to DSD coded systems algorithm Go out PCM data, the PCM data is converted into corresponding sample rate, digit, code stream by the DSD forms in data parameters, by this PCM data, sample rate, digit and code stream are sent to D/A converter module, and are changed according to the sample rate and code stream logarithmic mode Module is configured.
CN201710698979.8A 2017-08-15 2017-08-15 System and method for realizing DSD audio hard solution Active CN107507617B (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
CN201710698979.8A CN107507617B (en) 2017-08-15 2017-08-15 System and method for realizing DSD audio hard solution

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
CN201710698979.8A CN107507617B (en) 2017-08-15 2017-08-15 System and method for realizing DSD audio hard solution

Publications (2)

Publication Number Publication Date
CN107507617A true CN107507617A (en) 2017-12-22
CN107507617B CN107507617B (en) 2022-04-22

Family

ID=60691061

Family Applications (1)

Application Number Title Priority Date Filing Date
CN201710698979.8A Active CN107507617B (en) 2017-08-15 2017-08-15 System and method for realizing DSD audio hard solution

Country Status (1)

Country Link
CN (1) CN107507617B (en)

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN113744746A (en) * 2021-09-07 2021-12-03 广州飞傲电子科技有限公司 Audio data conversion playing method and device and audio player
CN113763971A (en) * 2021-09-07 2021-12-07 广州飞傲电子科技有限公司 Audio decoding control method and device and audio decoding control equipment
CN115278458A (en) * 2022-07-25 2022-11-01 邓剑辉 Multi-channel digital audio processing system based on PCIE interface

Citations (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20050010399A1 (en) * 2003-06-17 2005-01-13 Cirrus Logic, Inc. Circuits and methods for reducing pin count in multiple-mode integrated circuit devices
CN105047200A (en) * 2015-07-21 2015-11-11 重庆邮电大学 FPGA-based FLAC hardware decoder and decoding method
CN105513603A (en) * 2014-10-16 2016-04-20 北京海格神舟通信科技有限公司 DSP-based low-speed voice coding and decoding module
CN105869647A (en) * 2016-05-05 2016-08-17 西安睿芯微电子有限公司 Smart phone native DSD audio decoding method and system and smart phone
US20160322056A1 (en) * 2015-05-01 2016-11-03 Rohm Co., Ltd. Dsd decoder and audio system
CN106648537A (en) * 2016-12-29 2017-05-10 维沃移动通信有限公司 Audio data decoding control method and mobile terminal

Patent Citations (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20050010399A1 (en) * 2003-06-17 2005-01-13 Cirrus Logic, Inc. Circuits and methods for reducing pin count in multiple-mode integrated circuit devices
CN105513603A (en) * 2014-10-16 2016-04-20 北京海格神舟通信科技有限公司 DSP-based low-speed voice coding and decoding module
US20160322056A1 (en) * 2015-05-01 2016-11-03 Rohm Co., Ltd. Dsd decoder and audio system
CN105047200A (en) * 2015-07-21 2015-11-11 重庆邮电大学 FPGA-based FLAC hardware decoder and decoding method
CN105869647A (en) * 2016-05-05 2016-08-17 西安睿芯微电子有限公司 Smart phone native DSD audio decoding method and system and smart phone
CN106648537A (en) * 2016-12-29 2017-05-10 维沃移动通信有限公司 Audio data decoding control method and mobile terminal

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN113744746A (en) * 2021-09-07 2021-12-03 广州飞傲电子科技有限公司 Audio data conversion playing method and device and audio player
CN113763971A (en) * 2021-09-07 2021-12-07 广州飞傲电子科技有限公司 Audio decoding control method and device and audio decoding control equipment
CN113744746B (en) * 2021-09-07 2023-08-08 广州飞傲电子科技有限公司 Audio data conversion playing method and device and audio player
CN115278458A (en) * 2022-07-25 2022-11-01 邓剑辉 Multi-channel digital audio processing system based on PCIE interface

Also Published As

Publication number Publication date
CN107507617B (en) 2022-04-22

Similar Documents

Publication Publication Date Title
CN1934646B (en) Method and system for synchronizing audio processing modules
CN101958139A (en) High definition lossless audio playing (HDAP) system
CN107507617A (en) A kind of system and method realized DSD audios and solved firmly
CN102063908A (en) Audio data transmission method between PC and mobile phone
JP4621368B2 (en) Controller and method for controlling interface with data link
CN208805783U (en) A kind of keyboard of integrated sound card function
CN104581522A (en) High-definition lossless audio playing system (HDAP)
CN202102715U (en) Multi-carrier real-time play controlling system for audio media
CN102332293A (en) Vehicle-mounted digital music player
CN202145381U (en) Record and playback device used in vehicle-mounted sound and miniature player
CN211654305U (en) Digital audio control system
CN203761375U (en) Audio processing device and audio playing equipment
CN1332365C (en) Method and device for sync controlling voice frequency and text information
CN1897110A (en) High-quality audio-frequency signal coverting playback device with solid memory as media
CN103152669A (en) Method for utilizing intelligent terminal to operate voice frequency effector
CN103065657A (en) Audio system with small scale integration (SSI) module and working method thereof
CN201532773U (en) Player used for vehicle-mounted acoustics or mini acoustics
CN108391202A (en) A kind of onboard audio dynamic sampling method and system
CN112597332A (en) Voice playing method and device embedded in MCU
CN203070758U (en) Audio recording and playing apparatus
CN209017278U (en) A kind of sound system with the enhancing of DSP audio
CN201449727U (en) Conversion device based on field programmable gate array
CN200997288Y (en) Phonetic reporting musical playing system
CN209201208U (en) A kind of radar video record playback platform based on CPCI framework
CN203366738U (en) Audio playing device and audio playing system

Legal Events

Date Code Title Description
PB01 Publication
PB01 Publication
SE01 Entry into force of request for substantive examination
SE01 Entry into force of request for substantive examination
GR01 Patent grant
GR01 Patent grant