CN101079296A - Audio frequency decoder and audio frequency decoding method - Google Patents
Audio frequency decoder and audio frequency decoding method Download PDFInfo
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- CN101079296A CN101079296A CN 200610026745 CN200610026745A CN101079296A CN 101079296 A CN101079296 A CN 101079296A CN 200610026745 CN200610026745 CN 200610026745 CN 200610026745 A CN200610026745 A CN 200610026745A CN 101079296 A CN101079296 A CN 101079296A
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Abstract
The invention discloses an AF decoder and AF decoding method, which comprises the following parts: analytical unit, which receives the external AF code flow to debag; decoding unit, which decodes the AF code flow to change IDCT and generate PCM sampling value; the resampling unit, which resamples the PCM sampling value according to preset sampling frequency rate to output; control unit, which controls the working of AF decoder, wherein the resampling unit can trim AF output velocity through transmitting the sampling rate of AF sampling point, which is fit for different AF play equipments with different codes without leading phase-position shake. The invention saves cost greatly without using VCO, which also decreases the chip area.
Description
Technical field
The present invention relates to audio decoder and audio-frequency decoding method, relate in particular to a kind of audio decoder and the audio-frequency decoding method that can adjust the decoding/playback rate of audio data stream.
Background technology
The audio frequency and video transmission technology is widely used in various information technical fields such as video conference, Digital Television, the networking telephone.Because audio, video data has mass property, industry generally adopts encoding and decoding technique to realize the transmission of audio, video data, for example, encode at transmitting terminal according to certain coding rule, and at receiving end by decoding with the corresponding decoding rule of transmitting terminal.Usually transmitting terminal can adopt certain encoded clock when coding, and receiving end also needs to be provided with the decode clock that is consistent with encoded clock when recovering audio, video data, thereby guarantees the continuous orderly broadcast of audio, video data.The for example extensive at present audio/video encoding standard that adopts, promptly mobile motion picture expert group version (MPEG) series standard, it reaches effective compression making full use of the redundancy on the room and time aspect the video compress, then mainly utilizes the subjective noise apperceive characteristic of people's ear to reach the purpose of compression aspect audio compression.For example, effective identification frequency range of people's ear is at 20～20KHz, this scope of can correspondingly giving prominence to the key points when carrying out the voice data compression is ignored the signal beyond this scope with interior signal, the while can also utilize the non-flatness of sound spectrum to reach the purpose of compression from an aspect.
But the sample frequency code system of the transmitting terminal of audio decoding apparatus and receiving end may be different, and for example the sampled point output speed of the sample frequency of transmitting terminal coding and receiving end is inconsistent, may need to carry out the conversion of sample frequency.On the other hand, owing to the reasons such as obstruction of channel, may cause not matching of receiving end local clock and transmitting terminal encoded clock.For example, carry out data transmission with the form of program stream (ProgramStream) or transport stream (Transport Stream) by various channels among the MPEG2, for example, satellite transmits, wire transmission or terrestrial transmission.Usually because the obstruction of channel causes the fluctuation transmitted, may cause not matching of receiving end local clock and transmitting terminal encoded clock.The delay of the scrambler of transmitting terminal simultaneously etc. also may cause not matching of receiving end local clock and transmitting terminal encoded clock.Clock do not match the error that produces usually can make play discontinuous.Thereby people have adopted the whole bag of tricks to eliminate this error, for example VCXO (VCO), phaselocked loop (PLL) etc.In general audio decoder, for realize audio frequency and video synchronously, usually wait until that error accumulation arrives when a certain amount of, for example deviation surpasses a frame, just adjust or etc. pending.At this moment, usually need to make the measure that frame-skipping etc. may influence system performance largely, so relatively influence output effect, and may cause the very serious phenomenon of jolting of consequence.If in phaselocked loop, adopt the frequency divider that can carry out fractional frequency division, can carry out the frequency adjustment of small step distance to a certain extent.Adopt fractional frequency division that certain limitation is also arranged, for example when realizing the frequency trim of high-precision requirement, use a large amount of circle filtering electric capacity, thereby be unfavorable for the integrated of chip for stable needs that keeps phaselocked loop.And, when the ratio of output frequency and phase detectors frequency is big, can produce bigger noise, the shake of the phase place of bringing therefrom is difficult to avoid.In addition, the cost of VCXO is unfavorable for the control to the whole cost of audio decoder chip than higher.
Summary of the invention
For overcoming the defective that exists in the prior art, the present invention proposes and a kind ofly make output signal have higher frequency control precision and realize lower-cost audio decoder and corresponding audio-frequency decoding method.According to an aspect of the present invention, provide a kind of audio decoder, comprising:
Resolution unit is in order to receive outside audio code stream and to unpack;
Decoding unit is decoded to described audio code stream, and carries out obtaining the PCM sampled value after idct transform and the windowing process;
The resampling unit is exported after described PCM sampled value resampled by predetermined sampling frequency ratio;
Control module is in order to control the work of described audio decoder.
In the above-mentioned audio decoder, described resampling unit comprises:
Pick-up unit in order to the conversion of sample frequency and/or the error of sample frequency are detected, and produces a frequency and adjusts the ratio reference value;
The frequency ratio control device is adjusted ratio reference value, output sampling frequency rate ratio according to described frequency;
Frequency adjusting device according to the value of frequency ratio control device output, adopts the filtering method reconfiguration waveform that sample frequency is carried out conversion and/or adjustment.
In the above-mentioned audio decoder, described frequency ratio control device comprises X ratio register and Y-ratio register, wherein, a value in two values of the value of the value of X ratio register and Y-ratio register is set to variable, another value is arranged to configurable constant, or two values all are set to variable.
In the above-mentioned audio decoder, described frequency ratio control device also comprises a computing unit, in order to after read clock PTS in the code stream of described pick-up unit output and the local clock RTC calculating, result of calculation is sent into one of X ratio register and Y-ratio register.
In the above-mentioned audio decoder, the value of one of described X ratio register and described Y-ratio register is set to the former sample frequency value of the coding side of code stream indication, and another value of described X ratio register and Y-ratio register is set to the broadcast sample frequency value of decoding unit.
In the above-mentioned audio decoder, the value of one of described X ratio register and described Y-ratio register is set to the ratio of the broadcast sample frequency value of the former sample frequency value of coding side of code stream indication and decoding unit, and another value of described X ratio register and Y-ratio register is set to a configurable constant.
According to a further aspect in the invention, provide a kind of audio-frequency decoding method, comprise the steps:
(a) receive outside audio code stream and unpacking;
(b) audio code stream that unpacks is decoded, and carry out obtaining the PCM sampled value after idct transform and the windowing process;
(c) after being resampled by predetermined sampling frequency ratio, described PCM sampled value exported.
In the said method, step (c) further comprises:
(c1) conversion of sample frequency and/or the error of sample frequency are detected, and produce a frequency adjustment ratio reference value;
(c2) adjust ratio reference value, output sampling frequency rate ratio according to described frequency;
(c3), adopt the filtering method reconfiguration waveform that sample frequency is carried out conversion and/or adjustment according to the sample frequency ratio of output.
In the said method, when step (c1) detects sample frequency conversion is arranged, finish code system conversion to different sample frequency; And when step (c1) detects sample frequency error is arranged, finish error correction to sample frequency.
In the said method, when step (c1) detects sample frequency and conversion arranged simultaneously sample frequency has error, code system conversion and error correction synthesized carry out a filtering.
In the said method, when step (c1) detects sample frequency and conversion arranged simultaneously sample frequency has error, filtering is separately carried out in code system conversion and error correction, again from synthesizing with the convolution form in logic.
In the said method, when step (c1) detects sample frequency error is arranged, obtain the error amount of sample frequency by following formula:
ΔF＝[(RTC-PTS)*C
F]；
Wherein, C wherein
FBe the step-length coefficient, RTC represents the local clock of receiving end audio decoder, and PTS represents the indicated read clock of code stream that receiving end receives.
In the said method, in step (c1), obtain frequency by following formula and adjust the ratio reference value:
F
s＝F
s0+ΔF*D
f
Wherein, the sampling frequency offset value of Δ F for obtaining, F
S0Be the crude sampling frequency values according to local clock, F
sBe detected output sampling frequency rate value, D
fFor adjusting precision.
In the said method, when step (c1) detects sample frequency conversion is arranged, the former sample frequency of the coding side of code stream indication and the broadcast sample frequency of decoding end are adjusted the ratio reference value as frequency.
Resampling of the present invention unit can be finely tuned the audio frequency output speed by conversion audio frequency sampling point sampling rate, simultaneously can be by resampling reconfiguration waveform adapting to the audio-frequence player device of different code systems, and can not cause phase jitter.The present invention does not need to use VCO, has significantly reduced the realization cost.Other element in simultaneously can the multiplexed audio demoder is realized this function, thereby also from having reduced chip area on the one hand, has reduced the realization cost.
Description of drawings
The following drawings is the aid illustration to exemplary embodiment of the present, to the elaboration of the embodiment of the invention, be to disclose feature of the present invention place, but do not limit the present invention in conjunction with the following drawings for further, same-sign is represented respective element or step among the embodiment among the figure, wherein:
Fig. 1 is the structural representation of the audio decoder of one embodiment of the present of invention.
Fig. 2 is the principle of work waveform synoptic diagram of a resampling unit in the audio decoder shown in Figure 1.
Fig. 3 is the structured flowchart of a resampling unit of audio decoder shown in Figure 1.
Fig. 4 is the process flow diagram of the audio-frequency decoding method of one embodiment of the present of invention.
Fig. 5 is the detail flowchart of step S3 in the audio-frequency decoding method shown in Figure 4.
Embodiment
A kind of audio decoder that the present invention discloses can be integrated in the mode of system single chip on the audio/video decoding chip, is used for the electronic product that needs are realized audio/video decoding, as digital TV, set-top box, DVD; Also can make with the audio decoder circuit form separately.
Fig. 1 shows the structural representation of the audio decoder of one embodiment of the present of invention.As shown in Figure 1, audio decoder of the present invention comprises resolution unit 11, decoding unit 12, resampling unit 13 and control module 14.In one embodiment of the invention, control module 14 is a control register, and ppu 15 is by the work of control register control audio demoder.In other embodiments of the invention, also can be directly with processor as control module 11, and without ppu.Resolution unit 11 receives outside audio code stream, sends into decoding unit 12 after unpacking; 12 pairs of audio code streams of decoding unit are decoded, and idct transform and windowing (Window) processing, obtain the PCM sampled value; Output after the 13 pairs of PCM sampled values in resampling unit resample.
Resampling unit 13 carries out filtering according to certain resampling algorithm to sampled value, with reconstructed sample point output waveform, and adjusts the sampled point output speed.Its principle of work as shown in Figure 2, arrow is arranged the time point of expression sampling in order among the figure, show for ease of clear, among the figure only with less sampled point as example.Waveform A is arranged under original sample frequency,, obtain the waveform B identical by waveform reconstruction, but the sample frequency of waveform B improves with waveform A when actual audio sample point output speed during greater than the playback rate of audio decoder; When actual audio sample point output speed during, obtain the waveform C identical by waveform reconstruction, but the sample frequency of waveform C reduces with waveform A less than the playback rate of audio decoder.Here the restructing algorithm of Cai Yonging for example can be interpolation algorithm, short time discrete Fourier transform algorithm or frequency domain prediction algorithm etc., and perhaps the reasonable combination of any several algorithms wherein combines with fourier transform algorithm as the time domain interpolation algorithm.For example in one embodiment of the invention, adopt the time domain interpolation algorithm, the method for utilizing interpolation progressively to approach is finished the conversion of sample frequency, thereby reaches the purpose of adjusting the sampled point output speed.
As shown in Figure 3, resampling unit 13 is provided with pick-up unit 131, frequency ratio control device 132 and frequency adjusting device 133.
Pick-up unit 131 can detect the conversion of sample frequency and/or the error of sample frequency.In some embodiments of the invention, pick-up unit 131 determines whether needs adjustment sample frequency according to the information in the code stream that receives.For example, pick-up unit 131 can determine whether that needs carry out the adjustment of sample frequency according to the header information of the basic stream bag of audio code stream.Frequency ratio control device 132 calculates frequency adjustment reference value or directly is recorded as frequency adjustment reference value according to the output valve of pick-up unit 131.Need the transformed samples frequency and when carrying out code system conversion and/or existing the error of sample frequency to correct, according to the value of frequency ratio control device 132 sample frequency is adjusted when being checked through by frequency adjusting device 133.Step-length and the accuracy rating adjusted can be arranged to variable.For example the precision of frequency adjusting device 133 adjustment can be utilized software programming setting as required.Adjust accuracy value and can be made as fixed-point number, the error rounding of can going ahead of the rest when adjusting is given up mantissa.Also can be made as floating number adjusting accuracy value, precision can be very high when using like this, can eliminate fully in theory to receive the error of playing the relative transmitting terminal encoded clock of end local clock.The sample frequency adjustment is mainly finished by the wave filter reconfiguration waveform.The wave filter here is the wave filter of decoding unit in the multiplexed audio demoder directly, thereby can save cost from an aspect.
Pick-up unit is to the former sample frequency F of coding side
S0Output sampling frequency rate F with decoding end
SCompare.As former sample frequency F
S0With output sampling frequency rate F
SWhen unequal,, the frequency ratio control device adjusts reference value for providing frequency.
In one embodiment of the invention, frequency ratio control device 132 comprises X ratio register 1321, Y-ratio register 1322 and computing unit 1323.X ratio register 1321 and Y-ratio register 1322 can be arranged in the control register of audio decoder.The value of X ratio register 1321 and Y-ratio register 1322 for example can be write by software by the outside, perhaps by pick-up unit testing result are directly imported.In one embodiment of the invention, X ratio register 1321 and Y-ratio register 1322 all are set to variable; In other embodiments of the invention, Y-ratio register 1322 is arranged to a configurable constant, and X ratio register 1321 is set to variable; Perhaps X ratio register 1321 is set to configurable constant, and Y-ratio register 1322 is set to variable.The relation of the value of Y-ratio register 1322 and X ratio register 1321 is as follows:
Y/X=F
S/ F
S0(formula one)
In the error that detects sample frequency and/or when needing the transformed samples frequency, frequency adjusting device 133 carries out frequency transformation according to the X ratio register 1321 of frequency ratio control device and the value of Y-ratio register 1322.
Δ t
2/ Δ t
1=F
S0/ F
S=X/Y (formula two)
x
n'=x
0+ (n-1) Δ t
2/ Δ t
1=x
0+ (n-1) X/Y (n=0,1,2......) (formula three)
y
n'=f (x
n') (formula four)
Below be that example is described further with the audio decoder that meets mpeg standard.
1. error correction example
The MPEG2 code stream becomes single code stream to transmit by suitable transmission medium a plurality of program multiplexings of being made up of related audio code stream and video code flow.Usually audio code stream in the program of packing and video code flow are provided with common time base so that can play simultaneously during decoding.The MPEG2 code stream is made of system layer and compression layer.Compression layer comprises the audio and video data streams that needs are play.System layer mainly comprises audio-visual synchronization, multiplexing, the information such as bag ID, error-detecting of stream.Usually audio code stream in the program of packing and video code flow are provided with common time base so that can play simultaneously during decoding.Set an end-to-end fixed delay time model among the MPEG2, supposed that the transmission from the scrambler to the demoder of all digital pictures and voice data all used the time of same length.Be provided with the program clock reference (PCR) that requires fixed delay time in the system layer.The head of compression layer comprises demonstration timestamp/decoded time stamp (PTS/DTS).The local clock of demoder (RTC) has roughly the same frequency with the local clock of scrambler, is generally 27MHz.The local clock of the local clock of demoder and scrambler is consistent by PCR.In demoder, adopt PCR to correct local clock, and adjust audio-visual synchronization by PTS and play.The PCR territory is 42, comprises 33 PCR base segment and extension two parts of 9.Base segment characterizes the unit of 90MHz, and the extension characterizes the unit of 27MHz.The PTS territory is 33, characterizes the unit of 90KHz, is 1/300th of 27MHz.In meeting the audio decoder of mpeg standard, can utilize the value in PTS territory to eliminate the error of sample frequency.
Pick-up unit finds that the two is not consistent after the PTS in the code stream and local clock RTC are compared, and then PTS in the code stream and local clock RTC is outputed to frequency ratio control device 132 simultaneously.In one embodiment of the invention, after frequency ratio control device 132 calculates, result of calculation is sent into X ratio register 1321 and Y-ratio register 1322 respectively.
According to following formula five, can obtain the error amount Δ F of sample frequency, wherein C
FBe the segmentation quantizing factor, available form of tabling look-up is converted into frequency domain value with the time clock correction of time domain, for example can be with C
FBe set to 1/128.Segmentation quantizing factor C
FFor example can change by software setting.In the formula five, RTC represents the local clock of decoding end audio decoder, and PTS represents the indicated read clock of code stream that decoding end receives.
Δ F=[(RTC-PTS) * C
F]; (formula five)
According to following formula six, can obtain the output sampling frequency rate value of decoding end.F in the formula six
sBe output sampling frequency rate value, F
S0Be the crude sampling frequency values according to local clock, D
fFor adjusting precision, for example can be with D
fBe set to 2
-16Adjust precision D
fFor example can change by software setting.In one embodiment of the invention, can be with F
S0Send into the value of frequency ratio control device 132, with F as X ratio register 1321
sSend into the value of frequency ratio control device 132 as Y-ratio register 1322.
F
s=F
S0+ Δ F*D
f(formula six)
In another embodiment of the present invention, the Y-ratio register 1322 of frequency ratio control device 132 can be arranged to configurable constant, for example a 0X10000.X ratio register 1321 can be set to variable, and control module is to F
S0/ F
sAfter calculating, the result of gained is write X ratio register 1321.
2. transformed samples frequency and the example of carrying out code system conversion
In some embodiments of the invention, pick-up unit can also determine whether that needs carry out the code system conversion according to the header information of the basic stream of audio code stream between different sample frequency.At this is that example is described further with the audio decoder that meets mpeg standard still.
For receiving, audio frequency defined three kinds of different audio sampling frequencies: 48KHz, 44.1KHz, 32KHz among the mpeg standard file ISO/IEC11172-3 with Play System.ISO/IEC13818-3 has further defined half frequency sampling frequency, i.e. 24KHz, 22.05KHz, 16KHz in addition.In some other audio frequency reception and Play System, also defined the high frequency sample frequency, as 96KHz, 88.2KHz, 64KHz, and the very high frequency(VHF) sample frequency, as 192KHz, 176.4KHz, 128KHz.Frequency conversion apparatus of the present invention can carry out conversion between different audio sampling frequencies, for example be transformed into 44.1KHz from 48KHz; Also can in the deviation range up and down of given frequency, carry out the adjustment of error, for example adjust to 44.1KHz from 44.105KHz to same audio sampling frequency.
According to mpeg standard file ISO/IEC11172-3, comprise the sample frequency territory (sampling_frequency) of 2 (bit) in the header information (Header) of the basic stream (ES) of audio code stream.The sample frequency that defines in this standard is corresponding as shown in table 1 with basic stream head.Mpeg standard file ISO/IEC13812-3 further flows the implication that head has been set the ID territory of 1 (bit) substantially at audio code stream.The ID thresholding is 1, represents three kinds of common sample frequency, 44.1,48,32KHz; The ID thresholding is 0, expression three and half sample frequency frequently, 22.05,24,16KHz.
| Sample frequency (KHz) | The sample frequency thresholding |
| 44.1 | 00 |
| 48 | 01 |
| 32 | 10 |
| Keep | 11 |
Table 1
The sampled point playback rate that detects the sampled point output speed of the code stream that receives and audio decoder when pick-up unit 131 is inconsistent, and the former sample frequency of the coding side of code stream indication and the broadcast sample frequency of demoder are offered the frequency ratio control device.For example, can be directly the detected output sampling frequency rate of pick-up unit value F
SSend into Y-ratio register 1322, the former sample frequency value F of coding side
S0Send into X ratio register 1321.Carry out waveform reconstruct by frequency adjusting device 133 according to the value of frequency ratio control device 132, thereby adjust sample frequency.
In another embodiment of the present invention, the Y-ratio register 1322 of frequency ratio control device 132 can be arranged to configurable constant, for example a 0X10000; X ratio register 1321 can be set to the ratio of the broadcast sample frequency of the former sample frequency of coding side of code stream indication and decoding end.Carry out waveform reconstruct by frequency adjusting device 133 according to the value of frequency ratio control device 132, thereby adjust sample frequency.
In one embodiment of the invention, need the transformed samples frequency to carry out code system conversion and/or when existing the error of sample frequency to correct when being checked through, the application of two kinds of resamplings can be synthesized and carried out a filtering in same resampling unit, wherein new sample frequency value F
SThe sample frequency value that needs in two kinds of application to adjust can be synthesized and obtains.For example the broadcast sample frequency of code system conversion is 48KHz, and the Δ F of error correction is 0.005KHz, then can be directly the two addition, and the new sample frequency 48.005KHz after obtaining synthesizing.
In another embodiment of the present invention, the application of two kinds of resamplings can adopt the form of cascade to realize.From formal, can carry out filtering to a kind of application of resampling earlier and finish frequency change, for example first 48.005KHz with coding side transforms to 48KHz and reaches error correction, again filtering is carried out in the application of another kind resampling and is finished frequency change, for example again 48KHz is transformed to 32KHz.Can link to each other with the logical relation of convolution between two kinds of resamplings.
Fig. 4 is the process flow diagram of the audio-frequency decoding method of one embodiment of the present of invention.Referring to Fig. 4, a kind of audio-frequency decoding method of the present invention comprises the steps:
Step S1, the audio code stream that reception is outside also unpacks;
Step S2 decodes to the audio code stream that unpacks, and carries out obtaining the PCM sampled value after idct transform and the windowing process;
Step S3 is exported after described PCM sampled value resampled by predetermined sampling frequency ratio.
Fig. 5 is the detail flowchart of step S3 in the audio-frequency decoding method shown in Figure 4.Wherein, step S3 further comprises:
Step S31 detects the conversion of sample frequency and/or the error of sample frequency, and produces a frequency adjustment reference value;
Step S32 adjusts ratio reference value output sampling frequency rate ratio according to described frequency;
Step S33 according to the ratio of new sample frequency and former sample frequency, adopts the filtering method reconfiguration waveform, and sample frequency is carried out conversion and/or adjusted back output sampled point.
Wherein, resampling step is to adopt the output waveform of filtering method reconstructed sample point and adjust the sampled point output speed according to certain algorithm.This algorithm comprises that one or more algorithms in interpolation algorithm, short time discrete Fourier transform algorithm and the frequency domain prediction algorithm scheduling algorithm carry out filtering with reconfiguration waveform.
When the PCM sampled value was resampled, the resampling ratio of sample frequency was variable.In one embodiment of the invention, the accuracy rating of sample frequency can be utilized software programming setting as required.
The resampling ratio of sample frequency can according to audio code stream provide information be provided with.In meeting some embodiment of mpeg standard, for the adjustment of the error of sample frequency, the value in the PTS territory that is comprised according to the header information of the basic stream bag of audio code stream by the resampling ratio of sample frequency decides; For the conversion between the different sample frequency, the value in the sampling_frequencey territory that is comprised according to the header information of the basic stream of audio code stream by the resampling ratio of sample frequency decides, and the value in the ID territory that can comprise in conjunction with the header information of basic stream.
The foregoing description is just in order further more clearly to describe the present invention, but not limitation of the present invention.Be to be understood that the present invention is not limited to the elaboration that the foregoing description is done, anyly all should be encompassed within the spirit and scope of claim of the present invention based on modification of the present invention and equivalent of the present invention.
Claims (14)
1. audio decoder comprises:
Resolution unit is in order to receive outside audio code stream and to unpack;
Decoding unit is decoded to described audio code stream, and carries out obtaining the PCM sampled value after idct transform and the windowing process;
The resampling unit is exported after described PCM sampled value resampled by sample frequency ratio;
Control module is in order to control the work of described audio decoder.
2. audio decoder as claimed in claim 1 is characterized in that, described resampling unit comprises:
Pick-up unit in order to the conversion of sample frequency and/or the error of sample frequency are detected, and produces a frequency and adjusts the ratio reference value;
The frequency ratio control device is adjusted ratio reference value output sampling frequency rate ratio according to described frequency.
Frequency adjusting device according to the value of frequency ratio control device output, adopts the filtering method reconfiguration waveform that sample frequency is carried out conversion and/or adjustment.
3. audio decoder as claimed in claim 2, it is characterized in that, described frequency ratio control device comprises X ratio register and Y-ratio register, wherein, a value in two values of the value of the value of X ratio register and Y-ratio register is set to variable, another value is arranged to configurable constant, or two values all are set to variable.
4. audio decoder as claimed in claim 3, it is characterized in that, described frequency ratio control device also comprises a computing unit, in order to after read clock PTS in the code stream of described pick-up unit output and the local clock RTC calculating, result of calculation is sent into one of X ratio register and Y-ratio register.
5. audio decoder as claimed in claim 3, it is characterized in that, the value of one of described X ratio register and described Y-ratio register is set to the former sample frequency value of the coding side of code stream indication, and another value of described X ratio register and Y-ratio register is set to the broadcast sample frequency value of decoding unit.
6. audio decoder as claimed in claim 3, it is characterized in that, the value of one of described X ratio register and described Y-ratio register is set to the ratio of the broadcast sample frequency value of the former sample frequency value of coding side of code stream indication and decoding unit, and another value of described X ratio register and Y-ratio register is set to a configurable constant.
7. an audio-frequency decoding method comprises the steps:
(a) receive outside audio code stream and unpacking;
(b) audio code stream that unpacks is decoded, and carry out obtaining the PCM sampled value after idct transform and the windowing process;
(c) after being resampled by predetermined sampling frequency ratio, described PCM sampled value exported.
8. method as claimed in claim 7 is characterized in that, step (c) further comprises:
(c1) conversion of sample frequency and/or the error of sample frequency are detected, and produce a frequency adjustment ratio reference value;
(c2) adjust ratio reference value, output sampling frequency rate ratio according to described frequency;
(c3), adopt the filtering method reconfiguration waveform that sample frequency is carried out conversion and/or adjustment according to the sample frequency ratio of output.
9. method as claimed in claim 8 is characterized in that, when step (c1) detects sample frequency conversion is arranged, finishes the code system conversion to different sample frequency; And when step (c1) detects sample frequency error is arranged, finish error correction to sample frequency.
10. method as claimed in claim 9 is characterized in that, when step (c1) detects sample frequency and conversion arranged simultaneously sample frequency has error, code system conversion and error correction is synthesized carry out a filtering.
11. method as claimed in claim 9 is characterized in that, when step (c1) detects sample frequency and conversion arranged simultaneously sample frequency has error, filtering is separately carried out in code system conversion and error correction, again from synthesizing with the convolution form in logic.
12. as the arbitrary described method of claim 8 to 11, it is characterized in that, when step (c1) detects sample frequency error is arranged, obtain the error amount of sample frequency by following formula:
ΔF＝[(RTC-PTS)*C
F]；
Wherein, C wherein
FBe the step-length coefficient, RTC represents the local clock of receiving end audio decoder, and PTS represents the indicated read clock of code stream that receiving end receives.
13. method as claimed in claim 12 is characterized in that, in the step (c1), obtains frequency by following formula and adjusts the ratio reference value:
F
s＝F
s0+ΔF*D
f
Wherein, the sampling frequency offset value of Δ F for obtaining, F
S0Be the crude sampling frequency values according to local clock, F
sBe detected output sampling frequency rate value, D
fFor adjusting precision.
14. as the arbitrary described method of claim 8 to 11, it is characterized in that, when step (c1) detects sample frequency conversion is arranged, the former sample frequency of the coding side of code stream indication and the broadcast sample frequency of decoding end adjusted the ratio reference value as frequency.
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| CN103399724A (en) * | 2013-07-08 | 2013-11-20 | 江苏省广播电视集团有限公司 | Digital audio loudness measuring card |
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| CN105684078A (en) * | 2013-09-30 | 2016-06-15 | 奥兰治 | Resampling an audio signal for low-delay encoding/decoding |
| CN106528040A (en) * | 2016-11-02 | 2017-03-22 | 福建星网视易信息系统有限公司 | Method and apparatus for improving audio quality of android device |
| CN106797490A (en) * | 2014-09-11 | 2017-05-31 | 盖茨埃尔有限公司 | The hybrid modulation device being multiplexed using numeric field |
| CN112509591A (en) * | 2020-12-04 | 2021-03-16 | 北京百瑞互联技术有限公司 | Audio coding and decoding method and system |
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| Publication number | Priority date | Publication date | Assignee | Title |
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| JP3211771B2 (en) * | 1998-05-26 | 2001-09-25 | 日本電気株式会社 | Voice transceiver |
| JP3594514B2 (en) * | 1999-07-12 | 2004-12-02 | 株式会社ソニー・コンピュータエンタテインメント | Encoder, decoder, audio data processing device, audio data processing system, audio data compression method, audio data decompression method, audio data processing method, and recording medium |
| US20040010329A1 (en) * | 2002-07-09 | 2004-01-15 | Silicon Integrated Systems Corp. | Method for reducing buffer requirements in a digital audio decoder |
| US6650258B1 (en) * | 2002-08-06 | 2003-11-18 | Analog Devices, Inc. | Sample rate converter with rational numerator or denominator |
| CN1485849A (en) * | 2002-09-23 | 2004-03-31 | 上海乐金广电电子有限公司 | Digital audio encoder and its decoding method |
| CN100499621C (en) * | 2004-11-12 | 2009-06-10 | 上海乐金广电电子有限公司 | Carrier wave recovering equipment of OOB QPSK receiver |
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2006
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| CN103345376A (en) * | 2013-07-08 | 2013-10-09 | 南京琅声声学科技有限公司 | Method for monitoring digital audio signal volume |
| CN103399724A (en) * | 2013-07-08 | 2013-11-20 | 江苏省广播电视集团有限公司 | Digital audio loudness measuring card |
| CN103399724B (en) * | 2013-07-08 | 2015-12-09 | 江苏省广播电视集团有限公司 | A kind of DAB loudness measurement card |
| CN105684078A (en) * | 2013-09-30 | 2016-06-15 | 奥兰治 | Resampling an audio signal for low-delay encoding/decoding |
| CN107481726A (en) * | 2013-09-30 | 2017-12-15 | 皇家飞利浦有限公司 | Resampling is carried out to audio signal for low latency coding/decoding |
| CN105684078B (en) * | 2013-09-30 | 2020-03-27 | 皇家飞利浦有限公司 | Audio signal resampling method/apparatus and audio signal encoder/decoder |
| CN104753388A (en) * | 2013-12-30 | 2015-07-01 | 三星电机株式会社 | Driving apparatus, driving circuit and driving method of piezoelectric actuator |
| CN106797490A (en) * | 2014-09-11 | 2017-05-31 | 盖茨埃尔有限公司 | The hybrid modulation device being multiplexed using numeric field |
| CN106528040A (en) * | 2016-11-02 | 2017-03-22 | 福建星网视易信息系统有限公司 | Method and apparatus for improving audio quality of android device |
| CN112509591A (en) * | 2020-12-04 | 2021-03-16 | 北京百瑞互联技术有限公司 | Audio coding and decoding method and system |
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