CN102065366A - Method and device for processing multi-channel sound sample data - Google Patents
Method and device for processing multi-channel sound sample data Download PDFInfo
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- CN102065366A CN102065366A CN 201010525024 CN201010525024A CN102065366A CN 102065366 A CN102065366 A CN 102065366A CN 201010525024 CN201010525024 CN 201010525024 CN 201010525024 A CN201010525024 A CN 201010525024A CN 102065366 A CN102065366 A CN 102065366A
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Abstract
The invention provides a method and a device for processing multi-channel sound sample data, wherein, the method for processing the multi-channel sound sample data is applied between a processor and an external memory, and the processor is used for receiving and decoding a multi-channel sound signal to generate a plurality of sound sample data. The method comprises the following steps: the processor arranges the sound sample data according to a predetermined format; the processor starts a direct memory for access action so as to write the sound sample data into the external memory; and the external memory automatically switches a write-in address according to the predetermined format. By means of the method and the device provided by the invention, the data access burden of the sound signal processor in a sound playback system can be greatly reduced, and the delay effect can be adjusted according to the service environment.
Description
Technical field
The present invention is a kind of multi-channel sound sample data processing method and device, refers to be applied to the multi-channel sound sample data processing method and the device of sound playing system especially.
Background technology
Play out sound effect in order to reduce with telepresenc, the technological means of multichannel has been widely used in the audio frequency broadcast system, from stereophony the earliest, 5.1 sound channels of main flow even 6.1 sound channels up till now, purpose is exactly to simulate an auditory perception with spatial impression.See also Fig. 1, it is the schematic diagram of one 5.1 channel audio Play Systems, it is by two stereo preamplifier roads, the place ahead loud speaker 11,12, adds two three-dimensional surround channel loud speakers 13,14 of postposition, forms in conjunction with dead ahead independence center channel loud speaker 15 and supper bass channel loudspeaker 16 again.In the middle of the combination of 5.1 channel loudspeakers, loud speaker 11,12 and loud speaker 13,14 are responsible for four varying environment stereo sound effect contents that the orientation produced, 15 responsible leading roles' of center channel loud speaker sound (Voice) output, as for 16 of subwoofers in order to remedy and to strengthen bass performance in whole scene.
But because the restriction of real space, be not listening to environment and can allowing above-mentioned loud speaker be placed in desirable position to produce correct sound effect of each user, so according to actual needs the sound of different sound channels is carried out in various degree delay, just can effectively remedy the problems referred to above.For instance, when loud speaker 11 to first distance 110 of user's listening location greater than loud speaker 12 during to the second distance 120 of user's listening location, system can postpone the voice signal of output loud speaker 12, with the influence of counteracting distance difference, and then make the sound that reaches at last in user's ear reach the state of balance.So for the input of each loud speaker to actual range between user's listening location, Play System just can corresponding calculate each sound channel required time of delay, and then adjusts best result of broadcast by the user.
And in order to carry out necessary delay, Play System must store decoded finish and etc. voice data to be played, so a data buffer must be set in the Play System.And be to support each sound channel required different time of delays, this data buffer exists with the form of two-dimensional array, and wherein first dimension represent corresponding sound channel, and second ties up the degree of depth of the first-in first-out buffer that then is each sound channel.Therefore, the number of supporting sound channel along with Play System is big more, and the time of delay that can support is long more, and first dimension of above-mentioned data buffer is just big more with the number of second dimension, so must be provided with and manage a data buffer that capacity is very big in the Play System.
And if finish the setting and the management of this data buffer purely with firmware (firmware), Play System just must use the regional memory (local memory) of audio signal processor inside, for example be arranged at the static RAM (SRAM) 201 in the audio signal processor 20, but costing an arm and a leg of static RAM 201, excessive size can allow cost promote in a large number.Therefore, setting up hardware mode with the outside mostly at present finishes, the data buffer 22 shown in Fig. 2 (a) for example, data are managed and write to audio signal processor 20 to the data buffer of finishing with dynamic random access memory (DRAM) 22 by a bus 21, so can effectively save hardware cost.
Then express the data structure exemplary plot of this data buffer 22 as for Fig. 2 (b), for supporting 5.1 sound channels, wherein be provided with six first-in first-out buffers 220,221 ..., 225, and each first-in first-out buffer all has the index that writes of oneself, but shared one is read index.So when entering opening initialization in system, just to insert the delay data of right quantity in advance according to needed time of delay of each sound channel, example as shown in FIG., need the delay data of all inserting varying number in the first-in first-out buffer 220,221,224,225 of carryover effects, the first-in first-out buffer 222,223 that does not need carryover effects is not then inserted and is postponed to use data.
See also Fig. 2 (c) again, it is to utilize Fig. 2 (a), the flow chart of data processing figure that system environments shown in Fig. 2 (b) and data structure are carried out, and can know by Fig. 2 (c) and to find out, when audio signal processor 20 will carry out track selecting (step 32) and write data (step 34) before by 21 pairs of data buffers of bus 22, all must check (step 31 to data buffer 22,33), be ready to complete all in order to each first-in first-out buffer in the specified data buffer 22, the sample sound data that could begin that decoding is finished write in the corresponding first-in first-out buffer categorizedly, and utilize step 35,36 and 37, whether judgment data has write and has finished and whether will switch sound channel.And can find out by flow chart, audio signal processor 20 need carry out status checkout (step 31) by 21 pairs of data buffers of bus 22, add and also want judgment data whether to write to finish (step 35) and whether will switch sound channel (step 36), and to check all whether data buffer 22 is ready to complete (step 33) before writing the data of a sound channel at every turn, cause calculation resources to be overused, and then tie down voice codec operation originally, cause the system exception that can not expect.In addition, no matter be setting and the management of finishing above-mentioned data buffer with firmware or hardware, wherein all producing delay in advance with audio signal processor 20 inserts in the data buffer 22 with data again, so also will increase the workload of audio signal processor 20, transmit but also will take bus 21, and store in the data buffer 22, cause system resource to waste in a large number.And how to improve above-mentioned all existing disappearances, for developing main purpose of the present invention.
Summary of the invention
The present invention is a kind of multi-channel sound sample data processing method, be applied between a processor and the external memory storage, this processor is in order to receive a multi-channel sound signal and decode and produce a plurality of sample sound data, and this method comprises the following steps: that this processor arranges the data based predetermined format of described sample sound; And this processor startup one direct storage access action, in order to described sample sound data are write this external memory storage; Wherein, this external memory storage writes the address according to this predetermined format automatic switchover.
According to above-mentioned conception, multi-channel sound sample data processing method of the present invention, wherein this predetermined format is according to fixing sound channel order, each sound channel is arranged the sample sound data of fixed number.
According to above-mentioned conception, multi-channel sound sample data processing method of the present invention, wherein this external memory storage internal layout has N first-in first-out buffer, correspond respectively to N sound channel, and after the sample sound data of fixed number were write this first-in first-out buffer of one of them sound channel correspondence, the first-in first-out buffer that automatically switches to next sound channel correspondence write.
According to above-mentioned conception, multi-channel sound sample data processing method of the present invention wherein also comprises the following steps: when this N first-in first-out buffer initialization, sets N count value required different time of delays according to each sound channel.And when reading this first-in first-out buffer, check earlier corresponding this count value, if this count value is not 0, makes sample sound data that this first-in first-out buffer produces a delay usefulness automatically being read, and this count value is successively decreased 1 downwards.
Another aspect of the invention is a kind of multi-channel sound sample data processing unit, it comprises: a processor, decode and produce a plurality of sample sound data in order to receive a multi-channel sound signal, and the data based predetermined format of described sample sound can be arranged; One external memory storage is used to store described sample sound data; And a direct memory access controller, being electrically connected on this processor and this external memory storage, it is started by this processor and carry out a direct storage access and move, in order to described sample sound data are write this external memory storage; Wherein, this external memory storage writes the address according to this predetermined format automatic switchover.
According to above-mentioned conception, multi-channel sound sample data processing unit of the present invention, wherein this predetermined format is according to fixing sound channel order, each sound channel is arranged the sample sound data of fixed number.
According to above-mentioned conception, multi-channel sound sample data processing unit of the present invention, wherein this external memory storage comprises N first-in first-out buffer corresponding to N sound channel, respectively in order to writing the sample sound data of fixed number, and the first-in first-out buffer that automatically switches to next sound channel writes.
According to above-mentioned conception, multi-channel sound sample data processing unit of the present invention, wherein this external memory storage more comprises N counter, in order to set N count value required different time of delays according to each sound channel.And when this first-in first-out buffer is read, check corresponding this count value, if this count value is not 0, the sample sound data that this first-in first-out buffer produces a delay usefulness automatically to be being read, and this counter and automatically count value is successively decreased 1 downwards.
The present invention can reduce the data access burden of audio signal processor in the sound playing system in a large number, also can reach the effect of adjusting delay according to environment for use.
Description of drawings
Fig. 1 is the schematic diagram of one 5.1 channel audio Play Systems.
Fig. 2 (a) is the partial function module diagram of existing Play System.
Fig. 2 (b) is the data structure exemplary plot of data buffer in the existing Play System.
Fig. 2 (c) is the flow chart of data processing figure that utilizes existing system environment and data structure to carry out.
Fig. 3 is that the present invention is the multi-channel sound data processing function module diagram that the above-mentioned existing means disappearance of improvement develops out.
Fig. 4 (a) is the present invention when using the direct memory access (DMA) controller to carry out reading and writing data, is positioned at the data structure schematic diagram of audio signal processor end.
Fig. 4 (b) be N first-in first-out buffer 421,422 in the external memory storage 32 of the present invention ..., the module diagram of 42N.
Fig. 5 (a) is the data decode carried out of audio signal processor of the present invention and write flow chart.
Fig. 5 (b) is the audio signal processor of the present invention data playback of carrying out and read flow chart.
Being simply described as follows of symbol in the accompanying drawing:
Stereo preamplifier road loud speaker: 11,12
Rearmounted three-dimensional surround channel loud speaker: 13,14
Dead ahead independence center channel loud speaker: 15
Supper bass channel loudspeaker: 16
First distance: 110
Second distance: 120
Audio signal processor: 20
Static RAM: 201
Data buffer: 22
Bus: 21
First-in first-out buffer: 220,221 ..., 225
Audio signal processor: 30
Direct memory access (DMA) controller: 31
External memory storage: 32
N first-in first-out buffer: 421,422 ..., 42N.
Embodiment
See also Fig. 3, it is that the present invention is the multi-channel sound data processing function module diagram that the above-mentioned existing means disappearance of improvement develops out, wherein audio signal processor 30 is finished with an external memory storage 32 by a direct memory access controller 31 and is connected, audio signal processor 30 can be finished by common digital signal processor (D SP) or microprocessor, it is mainly in order to receive a bit stream data (bitstream, for example common DVD voice data) and decode, and then produce the sample sound data of corresponding multichannel, and the sample sound data of corresponding multichannel are play (playback).And be the resource of avoiding taking audio signal processor 30, the data that audio signal processor 30 of the present invention is not directly handled for external memory storage 32 write, and carry out the action that data allocations writes but use direct memory access (DMA) controller 31 instead.
See also Fig. 4 (a), it is the present invention when using direct memory access (DMA) controller 31 to carry out reading and writing data, is positioned at the data structure of audio signal processor 30 ends.With the data channel width of external memory storage 32 be 32, burst (burst) number be the length of 8 and sample sound data to be 32 be example, each sound channel is 8 sample sound data of access fixedly, and switch sound channel according to permanent order.In other words, audio signal processor 30 is after producing the sample sound data of corresponding multichannel, only need the data structure arrangement of the sample sound data of decoding gained, read the sound sample data by direct memory access (DMA) controller 31 according to said sequence and quantity again and write in the external memory storage 32 according to the predetermined format as Fig. 4 (a) shown in.
And the internal layout of external memory storage 32 is just like shown in Fig. 4 (b), the N that its inside a comprises first-in first-out buffer 421,422 ..., 42N, correspond respectively to N sound channel.And external memory storage 32 also is designed to write the address according to above-mentioned predetermined format automatic switchover, after just every pair of first-in first-out buffer write the sample sound data of receiving some, the first-in first-out buffer that just automatically switches to next sound channel correspondence write.With 5.1 sound channels and above-mentioned specification is example, and number of channels N is 6, and external memory storage 32 every pair of first-in first-out buffers write receive 8 sample sound data after, just automatically switch to the first-in first-out buffer of next sound channel.Thus, the inventive method and available data process chart that need not be shown in Fig. 2 (c) are such, and sound channel of every switching or whenever write the voice data of some just needs to check the state of data buffer 22.
Below please refer to Fig. 5 (a), Fig. 5 (b), to understand the flow chart of data processing that audio signal processor 30 of the present invention is carried out in more detail.
Wherein the data decode carried out for audio signal processor 30 of the present invention of Fig. 5 (a) with write flow chart.At first, 30 pairs one multi-channel sound signals of audio signal processor carry out the decoding of bit stream and continue to produce the sample sound data, and insert the selected buffer (step 501) that inside own has; And when the 31 performed direct memory access (DMA) actions of direct memory access (DMA) controller are finished (step 502), sample sound data in this selected buffer are rearranged according to the data structure shown in Fig. 4 (a), and then trigger direct memory access (DMA) controller 31 and carry out direct memory access (DMA) action (step 503), and then make the sample sound data can write external memory storage 32 shown in Fig. 4 (b) in regular turn.Then just selected again another buffer (step 504) and get back to step 501 and carry out the sample sound data decode and write in inside own.
The data playback of being carried out for audio signal processor 30 of the present invention as for Fig. 5 (b) with read flow chart.At first, audio signal processor 30 judges whether these direct memory access (DMA) controller 31 performed direct memory access (DMA) actions finish (step 601); If, carry out the direct memory access (DMA) action just behind inner selected buffer itself, start direct memory access (DMA) controller 31, in order to the sample sound data read that will store in the external memory storage 32 to this selected buffer (step 602); And handle sample sound data stored in this buffer and play (step 603) then.
And can find out by Fig. 5 (a), Fig. 5 (b), because 30 pairs of sample sound data of audio signal processor are arranged according to predetermined format, and external memory storage 32 is when the sample sound data that reception writes, can automatically switch according to this predetermined format and write the address, audio signal processor 30 only need move in appropriate time startup direct memory access (DMA) and carry out data access, therefore can significantly reduce the computational burden of audio signal processor 30, and then allow system move more efficiently, and then reach purpose of the present invention.
In addition, for saving audio signal processor 30 produces and insert right quantity needed time of delay in advance according to each sound channel delay with the action of data to external memory storage, the present invention in external memory storage 32, be provided with correspond to N first-in first-out buffer 421,422 ..., N the counter of 42N, when being used to this first-in first-out buffer initialization, just can inserting respectively and represent each sound channel count value of required different time of delays.With N=6 in scheming is example, be respectively 1 corresponding to the count value in 6 counters of 6 sound channels, 2,4,4,0,0, and the sample sound data that store in audio signal processor 30 will be with external memory storage 32 are read when playing (playback), promptly starting direct memory access (DMA) controller 31 carries out direct memory access (DMA) and moves (as described in the flow chart of Fig. 5 (b)) when reading first-in first-out buffer, can check corresponding this count value earlier, if this count value is not 0, make sample sound data that this first-in first-out buffer automatically produces a delay usefulness being read, and this count value is successively decreased 1 downwards; Otherwise,, then directly read the data in this first-in first-out buffer if this count value is 0.When for example reading for the first time, it only is 0 first-in first-out buffer 425 corresponding to the count value in the counter, 426 actual sound sample data is read out broadcast, and initial value is not 0 first-in first-out buffer 421,422,423,424 sample sound data that can produce a delay usefulness automatically allow direct memory access (DMA) controller 31 read, and automatically the initial value in the counter is successively decreased 1 downwards, therefore through after reading for the first time, initial value in 6 counters will become 0,1,3,3,0,0, by that analogy, this method can the delay that does not need audio signal processor 30 to produce in advance and insert right quantity with data to the action of external memory storage, just can finish the effect of delay.
In sum, technological means of the present invention can effectively be improved the disappearance of existing means, and then under a large amount of data accesses that reduce audio signal processor in the sound playing system are born, can also finish the effect of adjusting delay according to environment for use, thoroughly reach main purpose of the present invention.And the present invention can be widely used on the various digital audio Play System, for example the computer system of DVD player, tool DVD CD player or the Disc player in generation Pleistocene etc.The above only is preferred embodiment of the present invention; so it is not in order to limit scope of the present invention; any personnel that are familiar with this technology; without departing from the spirit and scope of the present invention; can do further improvement and variation on this basis, so the scope that claims were defined that protection scope of the present invention is worked as with the application is as the criterion.
Claims (13)
1. multi-channel sound sample data processing method, it is characterized in that, be applied between a processor and the external memory storage, this processor is in order to receive a multi-channel sound signal and this multi-channel sound signal is decoded and produced a plurality of sample sound data, and this multi-channel sound sample data processing method comprises the following steps:
This processor is arranged the data based predetermined format of described sample sound; And
This processor starts a direct storage access action, in order to described sample sound data are write this external memory storage;
Wherein, this external memory storage writes the address according to this predetermined format automatic switchover.
2. multi-channel sound sample data processing method according to claim 1 is characterized in that, this predetermined format is according to fixing sound channel order, each sound channel is arranged the sample sound data of fixed number.
3. multi-channel sound sample data processing method according to claim 1, it is characterized in that, this external memory storage internal layout has N first-in first-out buffer, correspond respectively to N sound channel, and after the sample sound data of fixed number were write the first-in first-out buffer of one of them sound channel correspondence, the first-in first-out buffer that automatically switches to next sound channel correspondence write.
4. multi-channel sound sample data processing method according to claim 3 is characterized in that, also comprises the following steps:
When described N first-in first-out buffer initialization, set N count value required different time of delays according to each sound channel.
5. multi-channel sound sample data processing method according to claim 4, it is characterized in that, when reading this first-in first-out buffer, check corresponding count value earlier, if this count value is not 0, make sample sound data that this first-in first-out buffer automatically produces a delay usefulness being read, and this count value is successively decreased 1 downwards.
6. multi-channel sound sample data processing method according to claim 4 is characterized in that, when reading this first-in first-out buffer, checks corresponding count value earlier, if this count value is 0, then directly reads the data in this first-in first-out buffer.
7. multi-channel sound sample data processing method according to claim 1 is characterized in that, when playing described sample sound data, this processor starts this direct memory access (DMA) action, to read this external memory storage.
8. a multi-channel sound sample data processing unit is characterized in that, comprising:
One processor in order to receiving a multi-channel sound signal and this multi-channel sound signal is decoded and produced a plurality of sample sound data, and is arranged the data based predetermined format of described sample sound;
One external memory storage is used to store described sample sound data; And
One direct memory access controller is electrically connected on this processor and this external memory storage, and this direct memory access (DMA) controller is started by this processor and carries out a direct storage access and move, in order to described sample sound data are write this external memory storage;
Wherein, this external memory storage writes the address according to this predetermined format automatic switchover.
9. multi-channel sound sample data processing unit according to claim 8 is characterized in that, this predetermined format is according to fixing sound channel order, each sound channel is arranged the sample sound data of fixed number.
10. multi-channel sound sample data processing unit according to claim 8, it is characterized in that, this external memory storage comprises N first-in first-out buffer corresponding to N sound channel, described first-in first-out buffer is respectively in order to writing the sample sound data of fixed number, and the first-in first-out buffer that automatically switches to next sound channel correspondence writes.
11. multi-channel sound sample data processing unit according to claim 10 is characterized in that this external memory storage also comprises N counter, in order to set N count value required different time of delays according to each sound channel.
12. multi-channel sound sample data processing unit according to claim 11, it is characterized in that, when this first-in first-out buffer is read, check corresponding count value, if this count value is not 0, the sample sound data that this first-in first-out buffer automatically produces a delay usefulness are being read, and this counter successively decreases 1 downwards with count value automatically.
13. multi-channel sound sample data processing unit according to claim 11 is characterized in that, when this first-in first-out buffer is read, checks corresponding count value, if this count value is 0, then the data in this first-in first-out buffer directly are read.
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| JP2000250806A (en) * | 1999-02-26 | 2000-09-14 | Matsushita Electric Ind Co Ltd | Device and method for switching system for digital signal |
| CN1497585A (en) * | 1998-11-16 | 2004-05-19 | �ձ�ʤ����ʽ���� | Voice coding device and decoding device, optical recording medium and voice transmission method |
| CN1575044A (en) * | 2003-06-06 | 2005-02-02 | 三菱电机株式会社 | Multichannel audio system, and head unit and slave unit used for the same |
| CN1622694A (en) * | 2003-11-24 | 2005-06-01 | 三星电子株式会社 | Method and equipment for playing multichannel digital sound |
| CN1792116A (en) * | 2003-03-18 | 2006-06-21 | 布陆泰科株式会社 | A multi-channel speaker system and a connection system thereof |
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| CN1497585A (en) * | 1998-11-16 | 2004-05-19 | �ձ�ʤ����ʽ���� | Voice coding device and decoding device, optical recording medium and voice transmission method |
| JP2000250806A (en) * | 1999-02-26 | 2000-09-14 | Matsushita Electric Ind Co Ltd | Device and method for switching system for digital signal |
| CN1792116A (en) * | 2003-03-18 | 2006-06-21 | 布陆泰科株式会社 | A multi-channel speaker system and a connection system thereof |
| CN1575044A (en) * | 2003-06-06 | 2005-02-02 | 三菱电机株式会社 | Multichannel audio system, and head unit and slave unit used for the same |
| CN1622694A (en) * | 2003-11-24 | 2005-06-01 | 三星电子株式会社 | Method and equipment for playing multichannel digital sound |
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