CN101930738A - Multi-track audio signal decoding method and device - Google Patents
Multi-track audio signal decoding method and device Download PDFInfo
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Abstract
The invention relates to a multi-track audio signal decoding method and a device which are applied to a playing system. The multi-track audio signal decoding method comprises the following steps: a decoder receives a multi-track audio signal to be decoded; a first decoding program is carried out on the multi-track audio signal to be decoded to generate a multi-track audio signal; a second decoding program is carried out on first single-track audio data by a high-efficiency module because the first single-track audio data in the multi-track audio signal belong to a first class, thereby generating a first single-track audio signal; and a third decoding program is carried out on part data of second single-track audio data by a low-frequency module because the second single-track audio data in the multi-track audio signal belong to a second class, thereby generating a second single-track audio signal, wherein the number of instructions required by the third decoding program is less than the number of instructions required by the second decoding program.
Description
Technical field
The present invention relates to a kind of multi-track audio signal decoding method and device, relate in particular to the multi-track audio signal decoding method and the device that are applied in a kind of Play System.
Background technology
For effectively reducing the data volume of numerical coding sound signal, many data compression methods are developed, and wherein advance rank audio coding technology (Advanced Audio Coding is hereinafter to be referred as AAC) and are to develop quite fast and use a kind of quite widely.And be to pursue lower bit rate and higher sound quality, high-level efficiency is advanced rank audio coding technology (High Efficiency AdvancedAudio Coding is called for short HE-AAC) and is just arisen at the historic moment.It mainly is on the basis of advancing rank audio coding technology that high-level efficiency is advanced rank audio coding technology, increase frequency range and duplicated (Spectral BandReplication, abbreviation SBR) algorithm reaches very high compression efficiency, approximately can save 30% bit rate (bit-rates), and then under the condition of low bit rate, still can possess high-quality sound.
See also Fig. 1 (a), it is the code translator function block schematic diagram commonly used of a HE-AAC, with original samples frequency f s, audiorange 0～fa is example, sound signal wherein to be decoded is earlier through behind the AAC code translator 10, and then solve pulse-code modulation (the pulse codemodulation that sampling frequency only has fs/2, PCM) signal is sent into this signal one high-level efficiency module 11 then and is handled.111 pairs of these a plurality of voice datas of quadrature mirror filter bank analysis module in this high-level efficiency module 11 (Quadrature MirrorFilters Banks analysis) carry out demodulation analysis, and then go up to produce frequency band range at the low-frequency band voice data of 0～fa/2 and represent the high frequency band voice data (frequency band range is at one group of coefficient of fa/2～fa) at frequency domain (frequency domain), then this low-frequency band voice data and represent this group coefficient of high frequency band voice data to send into frequency range replication module (SBR module) 112 and carry out frequency range and duplicate, and then produce a high frequency band voice data, and this low-frequency band voice data and this high frequency band voice data reduce and produce the pulse-code modulation signal that sampling frequency is fs through after merging module 113 and quadrature mirror filter bank synthesis module (Quadrature Mirror Filters BanksSynthesis) 114.
And in video-audio playing system now, around audio (Surround) has been indispensable function, and can provide multichannel numerical coding sound signal that many kinds of forms are arranged around audio, wherein common have 5.1 channel format, and it is that the audio-frequency signal coding that will adhere to 6 sound channels separately becomes a multichannel numerical coding sound signal to store and transmit.And after Play System is reduced into the sound signal of 6 sound channels in decoding, utilize a pair of preposition loudspeaker L as Fig. 1 (b) shown in and R, a center loudspeaker C, a pair of circulating loudspeaker RL afterwards and RR and a mega bass loudspeaker Sub to play again.Each loudspeaker is responsible for independently sound effect separately, is responsible for main sound channel as preposition loudspeaker L and R, and the place ahead sound field is provided.Center loudspeaker C is responsible for the voice dialogue in the film, and back circulating loudspeaker RL and RR produce complete sound field Sensurround, and mega bass loudspeaker Sub is responsible for the task of low frequency output.
And with the HE-AAC 5.1 audio coding technology of above-mentioned two kinds of technology combinations, be widely used in Video CD (DVD) at present, aspect such as digital broadcasting and Digital Television, but, in decode procedure commonly used, must be with the code translator shown in signal input Fig. 1 to be decoded (a), and utilize quadrature mirror filter bank analysis module 111, frequency range replication module 112, merge module 113 and repeat 6 decoding actions with quadrature mirror filter bank synthesis module 114, the sound signal that belongs to 6 sound channels can be deciphered respectively to restore and play, too much operation program will cause the burden of Play System, and how to improve above-mentioned defective, be the fundamental purpose of implementing this case.
Summary of the invention
This case discloses a kind of multi-track audio signal decoding method, is applied to a Play System, and this method comprises the following step: receive multi-channel audio signal to be decoded; Multi-channel audio signal to be decoded is carried out first translator and produces a multi-channel audio signal; Utilize first monaural audio data to carry out second translator in response to first monaural audio data in the multi-channel audio signal belongs to first classification, and then produce first monophonic audio signal; And in response to second monaural audio data in the multi-channel audio signal belong to second the classification and utilize this second monaural audio data to carry out the 3rd translator, and then produce second monophonic audio signal, wherein the required number of instructions of the 3rd translator is less than the required number of instructions of this second translator, preferably, multi-channel audio signal to be decoded is HE-AAC 5.1 sound signals to be decoded, and first translator is to utilize the AAC code translator to carry out, and multi-channel audio signal is 6 sound channel pulse-code modulation signals.Preferably, first classification is the voice data of L channel, R channel, back L channel and back R channel, and this second classification is the voice data of center channels and mega bass sound channel, and judges that by the gauge outfit of resolving each news frame in these HE-AAC 5.1 sound signals to be decoded it belongs to this first classification or this second classification.
According to above-mentioned conception, the described multi-track audio signal decoding method of this case, wherein this second translator comprises the following step: first monaural audio data is carried out demodulation analysis, and then the one group of coefficient that on frequency domain, produces a low-frequency band voice data and represent the high frequency band voice data; Utilize this low-frequency band voice data and represent this group coefficient of high frequency band voice data to carry out frequency range and duplicate, and then produce a high frequency band voice data; And will synthesize after this low-frequency band voice data and the merging of this high frequency band voice data, reduction produces first monophonic audio signal.
According to above-mentioned conception, the described multi-track audio signal decoding method of this case, wherein the 3rd translator comprises the following step: insert numerical value between each sampling spot with second monaural audio data and be 0 sampling spot, and then form sampling spot and increase signal; And sampling spot is increased signal carry out low-pass filtering treatment, in order to the high frequency composition filtering in this signal, and then produce second monophonic audio signal.
According to above-mentioned conception, the described multi-track audio signal decoding method of this case, wherein this second monaural audio data is the bass audio data of a selected frequency range, and the 3rd translator more comprises the following step: the bass audio data of this selected frequency range is handled and given up a high frequency coefficients and the data of this second monaural audio data.
Another aspect of the invention is a kind of multi-channel audio signal code translator, be applied to a Play System, the multi-channel audio signal code translator comprises: code translator, in order to receive a multi-channel audio signal to be decoded and multi-channel audio signal to be decoded carried out first translator and produces a multi-channel audio signal; The high-level efficiency module, signal is connected in code translator, in order to first monaural audio data that belongs to first classification in this multi-channel audio signal is carried out second translator, and then produces first monophonic audio signal; And low frequency module, signal is connected in code translator, in order to second monaural audio data that belongs to second classification in the multi-channel audio signal is carried out the 3rd translator, and then produce second monophonic audio signal, wherein the required number of instructions of the 3rd translator is less than the required number of instructions of this second translator.Preferably, multi-channel audio signal to be decoded is HE-AAC 5.1 sound signals to be decoded, and code translator is the AAC code translator, and multi-channel audio signal is one 6 sound channel pulse-code modulation signals, and first classification is the voice data of L channel, R channel, back L channel and back R channel, and this second classification is the voice data of center channels and mega bass sound channel, and this code translator judges that by the gauge outfit of resolving each news frame in these HE-AAC 5.1 sound signals to be decoded it belongs to this first classification or this second classification.Preferably, the high-level efficiency module comprises: the quadrature mirror filter bank analysis module, signal is connected in code translator, in order to first monaural audio data is carried out demodulation analysis, and then the one group of coefficient that on frequency domain, produces a low-frequency band voice data and represent the high frequency band voice data; Frequency range replication module, signal are connected in the quadrature mirror filter bank analysis module, utilize this low-frequency band voice data and represent this group coefficient of high frequency band voice data to carry out frequency range and duplicate, and then produce a high frequency band voice data; Merge module, signal is connected in quadrature mirror filter bank analysis module and frequency range replication module, in order to low-frequency band voice data and high frequency band voice data are merged; And the quadrature mirror filter bank synthesis module, signal is connected in the merging module, synthesizes in order to low-frequency band voice data after will merging and high frequency band voice data, and reduction produces first monophonic audio signal.
According to above-mentioned conception, the described multi-channel audio signal code translator of this case, its medium and low frequency module comprises: sampling spot increases device, and signal is connected in code translator, be used for inserting between each sampling spot of this second monaural audio data numerical value and be 0 sampling spot, and then form a sampling spot and increase signal; And interpolation filter, signal is connected in sampling spot increases device, sampling spot is increased signal carry out low-pass filtering treatment, in order to the high frequency composition filtering in this signal, and then produces second monophonic audio signal.
According to above-mentioned conception, the described multi-channel audio signal code translator of this case, wherein should low frequency module be that the bass audio data of a selected frequency range of this second monaural audio data is handled, and this code translator be to give this low frequency module with the bass audio data of this selected frequency range and a high frequency coefficients and the data of giving up this second monaural audio data.
Description of drawings
This case will be in conjunction with following accompanying drawing and explanation, feasible more deep understanding:
Fig. 1 (a) is the code translator function block schematic diagram commonly used of HE-AAC;
Fig. 1 (b) is the speaker system block schematic diagram of 5.1 sound channels;
Fig. 2 is that this case is to improve a HE-AAC 5.1 code translator function block schematic diagrams that conventional means develops out;
Fig. 3 (a) is that sampling frequency is the sampling spot increase signal waveform synoptic diagram of fs;
Fig. 3 (b) is that sampling frequency is the pulse-code modulation signal waveform synoptic diagram of fs;
Fig. 4 is the schematic flow sheet of this case multichannel numerical coding audio signal decoding method of developing out.
[primary clustering symbol description]
Each assembly that is comprised in this case accompanying drawing lists as follows:
Quadrature mirror filter bank analysis module 111
Frequency range replication module 112 merges module 113
Quadrature mirror filter bank synthesis module 114
Quadrature mirror filter bank analysis module 211
Frequency range replication module 212 merges module 213
Quadrature mirror filter bank synthesis module 214
Interpolation filter 222
Embodiment
See also Fig. 2, it is that this case is to improve a HE-AAC 5.1 code translator function block schematic diagrams that conventional means develops out, at first, after the processing of HE-AAC 5.1 sound signals to be decoded through an AAC code translator 20, can solve sampling frequency is pulse-code modulation (pulse code modulation, the PCM) signal that fs/2 belongs to 6 sound channels.The voice data that wherein belongs to L channel, R channel, back L channel and back R channel is handled being admitted to high-level efficiency module 21 respectively.211 pairs of these a plurality of voice datas of quadrature mirror filter bank analysis module in this high-level efficiency module 21 (QuadratureMirror Filters Banks analysis) carry out demodulation analysis, and then go up to produce frequency band range at the low-frequency band voice data of 0～fa/2 and represent the high frequency band voice data (frequency band range is at one group of coefficient of fa/2～fa) at frequency domain (frequency domain), then this low-frequency band voice data and represent this group coefficient of high frequency band voice data to send into frequency range replication module (SBR module) 212 and carry out frequency range and duplicate, and then produce a high frequency band voice data, and behind this low-frequency band voice data and this high frequency band voice data process merging module 213 and the quadrature mirror filter bank synthesis module (Quadrature Mirror Filters BanksSynthesis) 214, reduction produces and belongs to L channel, R channel, back L channel is the pulse-code modulation signal of fs with the sampling frequency of back R channel.
But, owing to voice dialogue and the low-frequency effect in center channels (central channel) in 6 sound channels and the responsible respectively film of mega bass sound channel (Low Frequency Effect Channel), its upper frequency limit is respectively 8kHz and 200Hz, mid point fa/2 (about 12kHz) than whole audiorange upper limit fa (about 24kHz) is low, in this embodiment, both are not sent into the quadrature mirror filter bank analysis module 211 in the above-mentioned high-level efficiency module 21, frequency range replication module 212, merge module 213 and quadrature mirror filter bank synthesis module 214 and carry out complexity decoding computing, handle and transfer to the lower alternative of other operand.So, utilize the gauge outfit (header) of each news frame (frame) in 20 pairs of above-mentioned HE-AAC 5.1 sound signals to be decoded of AAC code translator to resolve (parse), and then know that this news frame (frame) is after belonging to 6 center channels and mega bass sound channel in the sound channel, the news frames (frame) of will being correlated with change to be delivered to the low frequency module 22 that this case sets up and handles, for example, the bass audio data that AAC code translator 20 will be selected frequency range is given low frequency module 22 and is handled and give up high frequency coefficients and/or data, in order to the center channels and the mega bass sound channel of voice and bass are directly carried out simply inserting the some processing, in order to save the complexity decoding computing of high-level efficiency module 21, therefore reduce the burden of system.
In Fig. 2, include a sampling spot in the low frequency module 22 and increase a device 221 and an interpolation filter 222, sampling spot increases the center channels of device 221 and the sampling frequency of mega bass sound channel signal is fs/2 owing to send into, to insert a numerical value in each sampling spot be 0 sampling spot and sampling spot increases device 221, and then form shown in Fig. 3 (a), sampling frequency is that the sampling spot of fs increases the signal waveform synoptic diagram, and then this sampling spot is increased signal send into interpolation filter 222, because this interpolation filter 222 is a low-pass filter, in order to the high frequency composition filtering in the signal, and then produce shown in Fig. 3 (b), sampling frequency is fs and belongs to center channels and the pulse-code modulation signal waveform synoptic diagram of mega bass sound channel signal.And above-mentioned high-level efficiency module 21 all can utilize digital signal processor (Digital Signal Processor is called for short DSP) to finish with low frequency module 22.
See also Fig. 4 again, it is the interpretation method schematic flow sheet of this case multichannel numerical coding sound signal of developing out, at first, after utilizing AAC code translator 20 that HE-AAC 5.1 sound signals to be decoded are carried out one first translator, solving sampling frequency is the pulse-code modulation signal (step 41) that fs/2 belongs to 6 sound channels.And resolve in these HE-AAC 5.1 sound signals to be decoded the gauge outfit of each news frame, judge whether it belongs to center channels signal or mega bass sound channel signal (step 42); If not, will interrogate frame delivers to high-level efficiency module 21 and carries out one second translator (step 43) and produce the pulse-code modulation signal that sampling frequency is fs; If, then will interrogate frame (frame) and change and deliver to the low frequency module 22 that this case sets up and carry out the 3rd translator (step 44), be the pulse-code modulation signal of fs in order to produce sampling frequency, therefore reduce the burden of system.
And compare with low frequency module 22 with the high-level efficiency module 21 of Fig. 2, utilize identical digital signal processor (Digital Signal Processor, be called for short DSP) the identical signal of processing, the required MIPS of second translator that required per second 1,000,000 instructions (MillionInstructions per second is called for short MIPS) the greater efficiency module 21 of the 3rd translator that low frequency module 22 is carried out is carried out minuses 30MIPS approximately.Therefore, present embodiment can effectively reduce the operand of system, and then improves the defective of conventional means.
In sum, the present invention effectively reduces the hardware complexity and the cost of multichannel numerical coding audio signal playback system, can be widely used in fields such as Video CD (DVD), digital broadcasting and Digital Television.Therefore the present invention can be by many modifications and variations that those skilled in the art did, the neither claimed scope of the application's claim of taking off.
Claims (16)
1. a multi-track audio signal decoding method is applied to Play System, and this interpretation method comprises the following step:
Receive a multi-channel audio signal to be decoded;
This multi-channel audio signal to be decoded is carried out first translator and produces a multi-channel audio signal;
Belong to first classification in response to first monaural audio data in this multi-channel audio signal, and utilize this first monaural audio data to carry out second translator, and then produce one first monophonic audio signal; And
Belong to one second classification in response to second monaural audio data in this multi-channel audio signal, and utilize this second monaural audio data to carry out the 3rd translator, and then produce one second monophonic audio signal, wherein the required number of instructions of the 3rd translator is less than the required number of instructions of this second translator.
2. multi-track audio signal decoding method according to claim 1 is characterized in that, wherein this multi-channel audio signal to be decoded is that a high-level efficiency to be decoded is advanced rank audio coding technology 5.1 sound signals.
3. multi-track audio signal decoding method according to claim 2 is characterized in that, wherein this first translator is to utilize an AAC code translator to carry out, and this multi-channel audio signal is one 6 sound channel pulse-code modulation signals.
4. multi-track audio signal decoding method according to claim 2 is characterized in that, wherein this second translator comprises the following step:
This first monaural audio data is carried out demodulation analysis, and then the one group of coefficient that on frequency domain, produces a low-frequency band voice data and represent the high frequency band voice data;
Utilize this low-frequency band voice data and represent this group coefficient of high frequency band voice data to carry out frequency range and duplicate, and then produce a high frequency band voice data; And
To synthesize after this low-frequency band voice data and the merging of this high frequency band voice data, reduction produces this first monophonic audio signal.
5. multi-track audio signal decoding method according to claim 2 is characterized in that, wherein the 3rd translator comprises the following step:
Insert a numerical value between each sampling spot and be 0 sampling spot, and then form a sampling spot and increase signal this second monaural audio data; And
This sampling spot is increased signal carry out low-pass filtering treatment, in order to the high frequency composition filtering in this signal, and then produce this second monophonic audio signal.
6. multi-track audio signal decoding method according to claim 2, it is characterized in that, wherein this first classification is the voice data of L channel, R channel, back L channel and back R channel, and this second classification is the voice data of center channels and mega bass sound channel, and advances in audio coding technology 5.1 sound signals of rank the gauge outfit of each news frame and judge that it belongs to this first classification or this second and classifies by resolving this high-level efficiency to be decoded.
7. according to claim 1 a described multi-track audio signal decoding method, it is characterized in that wherein this second monaural audio data is the bass audio data of a selected frequency range.
8. multi-track audio signal decoding method according to claim 7, it is characterized in that wherein the 3rd translator more comprises the following step: the bass audio data of this selected frequency range is handled and given up a high frequency coefficients and the data of this second monaural audio data.
9. a multi-channel audio signal code translator is applied to a Play System, and this code translator comprises:
One code translator is in order to receive a multi-channel audio signal to be decoded and this multi-channel audio signal to be decoded is carried out first translator and produces a multi-channel audio signal;
One high-level efficiency module, signal is connected in this code translator, in order to one first monaural audio data that belongs to first classification in this multi-channel audio signal is carried out second translator, and then produces one first monophonic audio signal; And
One low frequency module, signal is connected in this code translator, in order to one second monaural audio data that belongs to second classification in this multi-channel audio signal is carried out the 3rd translator, and then produce one second monophonic audio signal, wherein the required number of instructions of the 3rd translator is less than the required number of instructions of this second translator.
10. multi-channel audio signal code translator according to claim 9 is characterized in that, wherein this multi-channel audio signal to be decoded is that a high-level efficiency to be decoded is advanced rank audio coding technology 5.1 sound signals.
11. multi-channel audio signal code translator according to claim 10 is characterized in that, wherein this code translator is an AAC code translator, and this multi-channel audio signal is one 6 sound channel pulse-code modulation signals.
12. multi-channel audio signal code translator according to claim 10 is characterized in that, wherein this high-level efficiency module comprises:
One quadrature mirror filter bank analysis module, signal is connected in this code translator, in order to this first monaural audio data is carried out demodulation analysis, and then the one group of coefficient that on frequency domain, produces a low-frequency band voice data and represent the high frequency band voice data;
One frequency range replication module, signal are connected in this quadrature mirror filter bank analysis module, utilize this low-frequency band voice data and represent this group coefficient of high frequency band voice data to carry out frequency range and duplicate, and then produce a high frequency band voice data;
One merges module, and signal is connected in this quadrature mirror filter bank analysis module and this frequency range replication module, in order to this low-frequency band voice data and this high frequency band voice data are merged; And
One quadrature mirror filter bank synthesis module, signal are connected in this merging module, synthesize in order to this low-frequency band voice data after will merging and this high frequency band voice data, and reduction produces this first monophonic audio signal.
13. multi-channel audio signal code translator according to claim 10 is characterized in that, wherein should comprise by low frequency module:
One sampling spot increases device, and signal is connected in this code translator, is 0 sampling spot in order to insert a numerical value between each sampling spot with this second monaural audio data, and then forms a sampling spot and increase signal; And
One interpolation filter, signal are connected in this sampling spot increases device, this sampling spot is increased signal carry out low-pass filtering treatment, in order to the high frequency composition filtering in this signal, and then produces this second monophonic audio signal.
14. multi-channel audio signal code translator according to claim 10, it is characterized in that, wherein this first classification is the voice data of L channel, R channel, back L channel and back R channel, and this second classification is center channels and mega bass channel audio data, and this code translator advances in audio coding technology 5.1 sound signals of rank the gauge outfit of each news frame and judges that it belongs to this first classification or this second and classifies by resolving this high-level efficiency to be decoded.
15. multi-channel audio signal code translator according to claim 9 is characterized in that, should low frequency module be that a bass audio data of selecting frequency range of this second monaural audio data is handled wherein.
16. multi-channel audio signal code translator according to claim 15, it is characterized in that wherein this code translator is to give this low frequency module with the bass audio data of this selected frequency range and a high frequency coefficients and the data of giving up this second monaural audio data.
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