CN106653035B - method and device for allocating code rate in digital audio coding - Google Patents

method and device for allocating code rate in digital audio coding Download PDF

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CN106653035B
CN106653035B CN201611217367.4A CN201611217367A CN106653035B CN 106653035 B CN106653035 B CN 106653035B CN 201611217367 A CN201611217367 A CN 201611217367A CN 106653035 B CN106653035 B CN 106653035B
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channel
code rate
signal
sound
weight coefficient
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CN106653035A (en
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闫建新
王磊
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GUANGSHENG DIGITAL TECHNOLOGY Co Ltd GUANGZHOU
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GUANGSHENG DIGITAL TECHNOLOGY Co Ltd GUANGZHOU
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16Vocoder architecture
    • G10L19/18Vocoders using multiple modes
    • G10L19/24Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding

Abstract

the invention relates to a method and a device for code rate allocation in digital audio coding. The method selects a group of specific adjusting coefficients to carry out self-adaptive adjustment on the masking threshold of each sub-band from low frequency to high frequency in a sound channel so as to realize self-adaptive code rate distribution in the sound channel, carries out self-adaptive code rate distribution among the sound channels according to the characteristics of each sound channel, and selects a specific weight coefficient to carry out self-adaptive adjustment on the masking threshold of a mixed high-frequency signal part so as to realize self-adaptive code rate distribution in intensity stereo coding. By the self-adaptive code rate distribution methods, the invention can obtain better subjective sound quality.

Description

Method and device for allocating code rate in digital audio coding
Technical Field
the present invention relates to digital audio coding technology, and more particularly, to a method and apparatus for allocating code rate in digital audio coding.
Background
With the development of applications such as ultra-high definition television, the demand for audio is further increased in order to obtain an immersive auditory effect. For this reason, the number of channels of the input audio signal is significantly increased (e.g., 5.1.4, 7.1.4, 22.2, etc.), even with a plurality of independent target audio signals. Given the total coding rate, how to handle the rate allocation of each channel (including the target signal) and the rate allocation in each channel will affect the overall coding quality.
the current multi-channel digital audio coding, such as DRA5.1, AAC5.1, DD (DD +), DTS, etc., all belong to perceptual audio coding techniques, and frequency spectrum coefficients are quantized and entropy-coded in a transform domain or a subband domain by a masking threshold calculated by a psychoacoustic model, and usually, the characteristics of channels are not considered in code rate allocation, and all channels are treated the same. Taking DRA multi-channel coding technology as an example (similar to other coding algorithms), for an input multi-channel PCM signal, firstly, masking threshold calculation is performed by taking a critical frequency band of human hearing as a unit through a psychoacoustic model, and meanwhile, the input multi-channel PCM signal is converted from a time domain to a frequency domain through a filter bank by using Modified Discrete Cosine Transform (MDCT), so as to obtain MDCT coefficients of a plurality of channels. Depending on the set bit rate, such as 128kbps stereo or 384kbps 5.1 surround sound, there are generally two rate allocation schemes:
The first method comprises the following steps: the code rate is equally allocated to a plurality of channels, which is a simple allocation method, and bits are allocated in each channel in a free competition manner (specifically, refer to the following second manner). For stereo 128kbps, each channel 64 kbps; for 5.1 surround sound, the low frequency effect channel typically only encodes the low frequency part below 120Hz, and may be allocated a smaller code rate, such as 24kbps, and 72kbps for each of the other 5 full band channels.
and the second method comprises the following steps: free contention mode. Firstly, calculating the total number of bits per frame, wherein for DRA coding, the number of bits per frame of stereo is 128 x 1024/48 bits, namely about 2731 bits; 5.1 surround 384kbs, 8192 bits per frame. Then, according to the masking threshold value of each scale factor band (or called as a quantization unit) of each channel, firstly, the quantization precision is increased for the coefficients in the sub-band with the largest quantization noise (i.e. the quantization noise is least easily masked), a part of bits are separated from the total bits, then, the analysis is performed on the MDCT coefficients in which sub-band of all the sub-bands of all the channels needs to be increased most in the quantization precision, then, a part of bits are further distributed from the total bits to increase the quantization precision, and so on until all the bits are consumed finally, the code rate (bits) distribution is finished.
the second allocation method is relatively complex to implement and is less adopted. The first method of equal distribution is generally used. From the above two allocation manners, although it is considered that, for the case of 5.1 channels, the frequency band actually required to be encoded by the 1 channel is only 120Hz, other full frequency bands are generally encoded to 20kHz, and non-average code rate allocation is already performed, for the full frequency band channels in stereo, 5.1, generally, the code rate is allocated averagely for each channel, and then bits (or code rate) are allocated in a free competition manner in each channel. This method does not take into account the following two points: (1) for multi-channel cases above 5.1, the contribution of each full-band channel (including the target signal) to the overall subjective sound quality is not the same; (2) under the condition of a certain code rate requirement, the influence of low-frequency distortion and high-frequency distortion of each sound channel on the total subjective sound quality is different.
when encoding Stereo or multi-channel audio signals, Intensity Stereo Coding (Intensity Stereo Coding) is usually used if the code rate is below a certain value, e.g. DRA Stereo 96kbps, 5.1 surround 256 kbps. This is because when coding below this code rate, transparent coding quality cannot be achieved, and coding strategies need to be improved. The human auditory system is more sensitive to the envelope of the high frequency part of the audio and its details are less important, based on which the intensity stereo coding technique can mix the high frequencies of a stereo signal (or the high frequency parts of 5 full band signals of 5.1 channels) into one channel while transmitting the high frequency envelopes of all channels. The coding strategy ensures that better coded subjective sound quality is obtained at a lower code rate. For example, for stereo coding, typically the left (L) and right (R) channel intensity stereo coding process is as follows:
If the frequency point of the intensity stereo coding (usually the full band channel high frequency up to 20kHz) is 8kHz, the processed L channel is configured as follows: a high-frequency part formed by mixing 0-8 kHz + of the L sound channel (8-20 kHz high frequency of the L sound channel and 8-20 kHz high frequency of the R sound channel); the processed R channels are configured as: 0-8 kHz of the R sound channel. And then, carrying out free competition code rate allocation mode processing on the reconstructed left and right channels.
it can be seen that the intensity stereo coding method has the following problem in rate allocation: since only one mixed high frequency detail is transmitted, the high frequency part of each channel is recovered by this high frequency detail and the high frequency envelope of each channel at decoding time. If the high-frequency details shared by multiple channels are coded in a conventional multi-channel code rate allocation mode (without adopting an intensity stereo mode), the shared high-frequency parts have no advantages in code rate allocation, the high-frequency part distortion of each channel is limited in each channel when the channels are coded independently, and the high-frequency part detail coding distortion of the intensity stereo coding is brought into each channel.
Disclosure of Invention
The technical problem to be solved by the present invention is to provide a method and an apparatus for allocating code rate in digital audio coding, which can obtain better subjective sound quality, aiming at the above-mentioned defects of the prior art.
in order to solve the technical problem, the invention provides a method for allocating code rate in a sound channel in digital audio coding in a first aspect, which comprises the following steps:
S1, selecting a group of specific adjusting coefficients to carry out self-adaptive adjustment on the masking threshold of each subband from low frequency to high frequency in one sound channel;
And S2, carrying out global bit allocation in the channel based on the adjusted masking threshold.
in an embodiment according to the first aspect of the present invention, the specific set of adjustment coefficients in said step S1 is selected based on the personal insensitivity to distortion.
In an embodiment according to the first aspect of the present invention, the specific set of adjustment coefficients in said step S1 is selected based on the type of the input audio signal.
in an embodiment according to the first aspect of the present invention, the step S1 further includes:
For speech signals, selecting a coefficient smaller than 1 to reduce the masking threshold of the high-frequency sub-band;
For music-like signals, an appropriate adjustment factor is selected to lower the masking threshold of the lowest frequency band and lower the masking threshold of the middle frequency band relative to the lowest frequency band.
in order to solve the technical problem, the invention provides a method for allocating code rate between sound channels in digital audio coding in a second aspect, which comprises the following steps:
s1, analyzing the channel characteristics of the input multi-channel audio signal to obtain channel configuration information;
s2, according to the sound channel configuration information, on the basis of the average distribution code rate, adjusting different weight coefficients of the code rate of each sound channel;
And S3, performing global bit allocation based on the adjusted code rate of each channel.
In an embodiment according to the second aspect of the present invention, for a5.1 channel audio signal, the adjusting of the code rate in step S2 includes: giving higher weight coefficients to the L channel and the R channel of the front sound field than to the LS channel and the RS channel of the rear sound field; for the center channel, smaller weight coefficients are given compared to the other channels when the total coding rate is high, and higher weight coefficients are given compared to the other channels when the total coding rate is low.
In an embodiment according to the second aspect of the present invention, for a 3D multi-channel audio signal, the adjusting of the code rate in step S2 includes: the weight coefficient of the middle channel is higher than that of the top channel, the weight coefficient of the top channel is higher than that of the bottom channel, and the weight coefficient of the front channel is higher than that of the rear channel.
in an embodiment according to the second aspect of the present invention, for a case where the input multi-channel audio signal includes a target signal, the step S1 further includes: analyzing the characteristics of the target signal to obtain target signal description information; the step S2 further includes: and determining the weight coefficient of the target signal code rate allocation based on the target signal description information.
in an embodiment according to the second aspect of the present invention, the determining the weight coefficients of the code rate allocation of the target signal in step S2 further includes:
when the target signal is the accompanying sound of different languages, giving the same weight coefficient as the central sound track;
and when the target signal is a directional active target signal, a weight coefficient is distributed to a code rate lower than that of the sound channel signal.
in order to solve the technical problem, the present invention provides a method for allocating a code rate in an intensity stereo coding of a digital audio in a third aspect, comprising the following steps:
S1, selecting a specific weight coefficient to carry out self-adaptive adjustment on the masking threshold of the mixed high-frequency signal part;
and S2, performing free competition code rate allocation based on the adjusted masking threshold.
in an embodiment according to the third aspect of the present invention, the weighting factor is selected based on an analysis of the mixed high frequency signal in step S1, wherein the weighting factor is higher the more high frequency components.
in order to solve the above technical problem, the present invention provides in a fourth aspect an apparatus for in-channel code rate allocation in digital audio coding, comprising:
the masking threshold adjusting module is used for selecting a group of specific adjusting coefficients to carry out self-adaptive adjustment on the masking threshold of each subband from low frequency to high frequency in one sound channel;
and the bit allocation module is used for carrying out global bit allocation in the sound channel based on the adjusted masking threshold.
in order to solve the technical problem, in a fifth aspect, the present invention provides an apparatus for allocating inter-channel code rate in digital audio coding, including:
the analysis module is used for carrying out sound channel characteristic analysis on the input multi-channel audio signal to obtain sound channel configuration information;
the code rate adjusting module is used for adjusting different weight coefficients of the code rate of each sound channel on the basis of the average allocated code rate according to the sound channel configuration information;
and the bit distribution module is used for carrying out global bit distribution based on the adjusted code rate of each sound channel.
in order to solve the technical problem, in a sixth aspect, the present invention provides an apparatus for allocating a code rate in an intensity stereo coding of a digital audio, including:
The masking threshold adjusting module is used for selecting a specific weight coefficient to carry out self-adaptive adjustment on the masking threshold of the mixed high-frequency signal part;
and the code rate allocation module is used for allocating free competition code rates based on the adjusted masking threshold.
The method and the device for allocating the code rate in the digital audio coding can adaptively process the code rate allocation among the sound channels, the code rate allocation in each sound channel and the code rate allocation when intensity stereo coding is used, thereby obtaining better subjective sound quality.
Drawings
the invention will be further described with reference to the accompanying drawings and examples, in which:
FIG. 1 is a flow chart of a method for in-channel rate allocation in digital audio coding according to an embodiment of the present invention;
FIG. 2 is a schematic diagram illustrating the method of in-channel code rate allocation in digital audio coding according to an embodiment of the present invention;
FIG. 3 is a diagram illustrating automatic adjustment coefficients of speech-like signals according to an embodiment of the present invention;
FIG. 4 is a diagram illustrating an automatic adjustment coefficient of a music-like signal according to an embodiment of the present invention;
FIG. 5 is a flowchart of a method for inter-channel rate allocation in digital audio coding according to an embodiment of the present invention;
FIG. 6 is a schematic diagram illustrating a method for inter-channel rate allocation in digital audio coding according to an embodiment of the present invention;
FIG. 7 is a flow chart of a method of rate allocation in intensity stereo coding of digital audio according to one embodiment of the present invention;
FIG. 8 is a schematic diagram illustrating a method for rate allocation in intensity stereo coding of digital audio according to an embodiment of the present invention;
FIG. 9 is a logic block diagram of an apparatus for in-channel code rate allocation in digital audio coding according to an embodiment of the present invention;
FIG. 10 is a logic diagram of an apparatus for inter-channel rate allocation in digital audio coding according to an embodiment of the present invention;
Fig. 11 is a logic block diagram of an apparatus for rate allocation in intensity stereo coding of digital audio according to an embodiment of the present invention.
Detailed Description
In order to make the objects, technical solutions and advantages of the present invention more apparent, the present invention is described in further detail below with reference to the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
the invention provides a method and a device for code rate allocation in digital audio coding, which are mainly applied to the application field of multichannel coding such as a 3D audio system of an ultra-high definition television and the like. The method and the device for allocating the code rate in the digital audio coding mainly consider the following problems: (1) under the condition of a certain code rate, the influence of low-frequency distortion and high-frequency distortion of each sound channel on the total subjective sound quality is different; (2) the contribution of each full-band channel (target signal) to the overall subjective sound quality is not the same; (3) in intensity stereo coding, the coding rate allocation of the mixed high frequency part should be increased appropriately.
Based on the above problems, the present invention firstly proposes an improvement on the way of allocating the code rate in the sound channel. FIG. 1 shows a flow diagram of a method 100 for in-channel rate allocation in digital audio coding, according to one embodiment of the invention. As shown in fig. 1, the method 100 includes the steps of:
In step S110, a group of specific adjustment coefficients is selected to perform adaptive adjustment on the masking threshold of each subband from low frequency to high frequency in one channel;
In step S120, global bit allocation in the channel is performed based on the adjusted masking threshold.
For the case that the given code rate condition is a high code rate, the code rate allocated to each channel during encoding is sufficient, and therefore, no adjustment is needed. However, for the case that the given bitrate condition is a medium-low bitrate, the subjective sound quality of the coded sound is not transparent, that is, the bitrate is not enough, and various distortions, such as quantization noise of a low frequency part, loss of a high frequency part, and the like, occur. In order to ensure that the final overall subjective sound quality is better, on one hand, in step S110, a set of adjustment coefficients may be selected according to personal insensitivity to distortion to appropriately adjust the masking threshold of each sub-band, so as to change the code rate allocation in one channel and obtain the required result. On the other hand, in step S110, a group of adjustment coefficients may be automatically selected based on the type of the input audio signal to adjust the masking threshold of each sub-band, so as to change the code rate allocation in one channel and obtain the comprehensive optimal effect.
The principle of adaptively adjusting the masking threshold based on the type of the input audio signal is shown in fig. 2, where the input audio signal is subjected to masking threshold calculation through a psychoacoustic model to obtain a masking threshold, and meanwhile, the input audio signal is subjected to signal type analysis to obtain signal type information, and then a specific adjustment coefficient is selected based on different signal types to adjust the masking threshold, and finally, global bit allocation in a channel is performed based on the adjusted masking threshold. The adjustment factor may be selected automatically internally based on the type of signal or may be set manually. For example, for speech-like signals, the masking threshold of the high frequency sub-band may be multiplied by a coefficient smaller than 1, such as 0.5, to reduce the masking threshold of the high frequency sub-band, thereby reducing the quantization accuracy of the high frequency appropriately, as shown in fig. 3. For another example, for a music-like signal, an appropriate adjustment coefficient may be selected to lower the masking threshold of the lowest frequency band and lower the masking threshold of the middle frequency band to a lesser extent relative to the lowest frequency band, so that the high frequency sub-band obtains a slightly higher code rate through free competition, and finally the high frequency distortion of the whole music-like audio is reduced, as shown in fig. 4.
The method for allocating the code rate in the sound channel in the digital audio coding of the embodiment of the invention can adaptively process the code rate allocation in the sound channel under the condition of the given code rate, thereby obtaining better subjective sound quality.
Based on the second problem, the present invention also provides an improvement in the code rate allocation between channels in encoding a multi-channel audio signal. Fig. 5 shows a flow diagram of a method 200 for inter-channel rate allocation in digital audio coding according to an embodiment of the invention. As shown in fig. 5, the method 200 includes the steps of:
in step S210, performing channel characteristic analysis on the input multi-channel audio signal to obtain channel configuration information;
in step S220, according to the channel configuration information, adjusting different weight coefficients for the code rate of each channel on the basis of evenly distributing the code rate;
in step S230, global bit allocation is performed based on the adjusted code rate of each channel.
Taking 5.1 channel audio signal as an example, first, the characteristics of the channel signals are analyzed to obtain channel configurations including 0.1 low frequency effect channel, L and R channels of the front sound field, LS and RS channels of the rear sound field, and center channel C. The low-frequency effect channel only encodes the low-frequency part below 120Hz, and the low code rate can be allocated independently. The other 5 full band channels are typically assigned the same code rate for each channel. This allocation can be used when the given bitrate condition is a sufficiently high bitrate (transparent quality). However, under the opaque quality code rate condition, considering that the human auditory system is more important for the front sound field (L & R), and the rear LS and RS are generally used to generate the rear ambient sound field, the L & R channel of the front sound field should be given a higher weight coefficient than the LS channel and the RS channel of the rear sound field. The center channel C is generally a dialogue signal (speech signal), so that the C channel may be given a smaller weight coefficient than other channels when the total coding rate is higher, and a higher weight coefficient than other channels when the total coding rate is lower, so as to ensure that the dialogue channel has a certain subjective sound quality. For example, at higher code rates (e.g., 320kpbs), L, R, C, LS has a ratio of 1.05:1.05:0.95:0.975: 0.975; when the code rate is 288kbps, the proportional relation between L, R, C, LS and RS is 1.05:1.05:1.0:0.95: 0.95; when the code rate is 256kbps, the proportional relation between L, R, C, LS and RS is 1.05:1.05:1.05:0.925: 0.925; when the code rate is 192kbps, the L, R, C, LS and RS have a proportional relation of 1.05:1.05:1.1:0.9: 0.9.
for a 3D multi-channel audio signal, such as 22.2 layer 3 channels, 5.1.4 double layer channels, etc., the adjustment principle of the code rate in step S220 is as follows: the weight coefficient of the middle channel is higher than that of the top channel, the weight coefficient of the top channel is higher than that of the bottom channel, and the weight coefficient of the front channel is higher than that of the rear channel. Taking 5.1.4 channel 3D audio as an example, in addition to the conventional 5.1 channels, there is one channel above L & R and LS & RS (named TopL, TopR, TopLs and TopRs) for generating an above sound field. At this time, if the code rate is still uniformly distributed by each channel, better subjective sound quality can not be obtained obviously. Usually, most of the sound is in the same plane as the human ear, and the upper channel merely provides ambient sound to improve the reality of the whole sound field. Therefore, the upper 4 channels (i.e., TopL, TopR, TopLs, and TopRs) may give less code rate than the average code rate, and if subdivided, the upper 2 rear channels (i.e., TopLs and TopRs) may give less code rate than the average code rate. Thus, several typical code rate configurations that can be selected are prioritized in a proportional order as follows:
(1) uniformly distributing;
(2)L&R>C>LS&RS>TopL&TopR>TopLs&TopRs;
(3)L&R>C>TopL&TopR>LS&RS>TopLs&TopRs;
(4)C>L&R>LS&RS>TopL&TopR>TopLs&TopRs;
(5)C>L&R>TopL&TopR>LS&RS>TopLs&TopRs。
For the above multiple rate allocation options of 5.1.4 channels, a certain fixed configuration, such as configuration (2), may be selected during encoding, and a rate configuration per frame may also be dynamically selected from the above several configurations according to the complexity and importance of each channel by analyzing the audio signal in real time.
for the case that the input multi-channel audio signal includes the target signal, for example, a certain target signal is added on the basis of the 5.1.4 channel signal, wherein the weight coefficient of the code rate allocation of the 5.1.4 channel can be determined according to the foregoing method, and the code rate allocation of the target signal needs to consider the characteristics of the target signal to determine the weight coefficient of the code rate allocation thereof. As shown in fig. 6, while inputting the channel configuration of the signal, the target signal contained therein needs to be analyzed to obtain the target signal description information, and then different weighting coefficients are determined based on different target signal characteristics. For example, if the target signal is an accompanying sound of a different language, the same weighting factor needs to be given as to the center channel C in the 5.1.4 channels; when the target signal is a directional moving target signal, the spatial directivity of the signal is emphasized, and the distortion of the signal itself can be relaxed appropriately, so that a lower code rate can be assigned with a weighting coefficient than that of the channel signal, for example, the weighting coefficient can be smaller than that of the middle layer channel and larger than that of the upper 4 channels.
The method for allocating the code rate among the sound channels in the digital audio coding of the embodiment of the invention can adaptively process the code rate allocation among the sound channels under the condition of the given code rate, thereby obtaining better subjective sound quality.
Based on the aforementioned third problem, the present invention also provides an improvement on the code rate allocation in the stereo coding of the multi-channel audio signal strength. Fig. 7 shows a flow diagram of a method 300 of rate allocation in intensity stereo coding of digital audio according to one embodiment of the invention. As shown in fig. 7, the method 300 includes the steps of:
In step S310, a specific weight coefficient is selected to perform adaptive adjustment on the masking threshold of the mixed high-frequency signal portion;
in step S320, free contention rate allocation is performed based on the adjusted masking threshold.
taking 5.1-channel intensity stereo coding as an example, the high frequencies of 5 full-band channels are mixed into a high-band detail signal and transmitted together with the 5-channel high-frequency envelope, at this time, the general free-competition code rate distribution mode is not reasonable, and a higher weight coefficient, such as a 1.2-time weight coefficient, needs to be given to the mixed high-frequency signal part, so that the masking curve of the high-frequency signal part can be properly reduced, and then free-competition code rate distribution is added, so that the high-frequency signal part can obtain higher code rate coding. The details of the contention-free rate allocation belong to the prior art, and are not described in detail herein. Furthermore, as shown in fig. 8, in step S310, a code rate allocation weight coefficient of the high frequency signal portion may be determined by performing signal analysis on the mixed high frequency signal portion, and then the masking threshold of the high frequency signal portion may be adaptively adjusted based on the weight coefficient. For example, if the higher the high frequency component, the higher the weight coefficient; if the high frequency component is noise-like, the weight coefficient may be appropriately reduced.
the method for allocating the code rate in the digital audio intensity stereo coding can adaptively adjust the code rate allocation of the mixed high-frequency signal part, thereby obtaining better subjective sound quality.
based on the method for allocating the code rate in the sound channel in the digital audio coding, the invention also provides a device for allocating the code rate in the sound channel in the digital audio coding. Fig. 9 shows a logic block diagram of an apparatus 400 for in-channel code rate allocation in digital audio coding according to an embodiment of the present invention. As shown in fig. 9, the apparatus 400 includes a masking threshold adjustment module 410 and a bit allocation module 420. The masking threshold adjusting module 410 is configured to select a specific set of adjusting coefficients to adaptively adjust the masking thresholds of the subbands from low frequency to high frequency in one channel; the bit allocation module 420 is configured to perform an intra-channel global bit allocation based on the adjusted masking threshold. The apparatus 400 shown in fig. 9 can be used to perform the method 100 for allocating an in-channel bitrate in the digital audio coding shown in fig. 1, and refer to the description of the method 100.
Based on the method for distributing the code rate among the sound channels in the digital audio coding, the invention also provides a device for distributing the code rate among the sound channels in the digital audio coding. Fig. 10 shows a logic block diagram of an apparatus 500 for inter-channel rate allocation in digital audio coding according to an embodiment of the present invention. As shown in fig. 10, the apparatus 500 includes an analysis module 510, a rate adjustment module 520, and a bit allocation module 530. The analysis module 510 is configured to perform channel characteristic analysis on an input multi-channel audio signal to obtain channel configuration information; the code rate adjusting module 520 is configured to adjust the code rate of each channel by using different weight coefficients on the basis of evenly allocating the code rate according to the channel configuration information; the bit allocation module 530 is configured to perform global bit allocation based on the adjusted code rate of each channel. For the case that the input multi-channel audio signal includes a target signal, the analysis module 510 further analyzes the characteristics of the target signal to obtain target signal description information; the rate adjustment module 520 also determines the weighting coefficients of the rate allocation of the target signal based on the target signal description information. The apparatus 500 shown in fig. 10 can be used to perform the method 200 for allocating inter-channel coding rate in digital audio coding shown in fig. 5, and refer to the description of the method 200.
based on the method for allocating the code rate in the intensity stereo coding of the digital audio, the invention also provides a device for allocating the code rate in the intensity stereo coding of the digital audio. Fig. 11 shows a logic block diagram of an apparatus 600 for rate allocation in intensity stereo coding of digital audio according to an embodiment of the present invention. As shown in fig. 11, the apparatus 600 includes a masking threshold adjustment module 610 and a code rate allocation module 620. The masking threshold adjusting module 610 is configured to select a specific weight coefficient to perform adaptive adjustment on the masking threshold of the mixed high-frequency signal portion; the rate allocation module 620 is configured to perform contention-free rate allocation based on the adjusted masking threshold. The apparatus 600 shown in fig. 11 may be used to perform the method 300 for rate allocation in intensity stereo coding of digital audio shown in fig. 7, and refer to the description of the method 300.
Based on the method and the device for allocating the code rate in the digital audio coding, when intensity stereo coding is adopted under the condition of certain medium code rates in a digital audio coding algorithm, self-adaptive code rate allocation can be carried out to improve subjective sound quality; under the condition of medium and low code rate, the self-adaptive code rate distribution in the sound channel can be carried out to improve the subjective sound quality; for multi-channel audio coding (e.g., above 5.1), the characteristic adaptive bitrate allocation between channels can be exploited to improve the overall subjective sound quality.
the above description is only for the purpose of illustrating the preferred embodiments of the present invention and is not to be construed as limiting the invention, and any modifications, equivalents and improvements made within the spirit and principle of the present invention are intended to be included within the scope of the present invention.

Claims (9)

1. A method for rate allocation in a channel in digital audio coding, comprising the steps of:
s1, selecting a group of specific adjusting coefficients to carry out adaptive adjustment on the masking threshold of each sub-band from low frequency to high frequency in one channel based on insensitivity of individuals to distortion or based on the type of input audio signals;
And S2, carrying out global bit allocation in the channel based on the adjusted masking threshold.
2. The method according to claim 1, wherein the selecting a specific set of adjustment coefficients based on the type of the input audio signal in step S1 further comprises:
for speech signals, selecting a coefficient smaller than 1 to reduce the masking threshold of the high-frequency sub-band;
For music-like signals, an appropriate adjustment factor is selected to lower the masking threshold of the lowest frequency band and lower the masking threshold of the middle frequency band relative to the lowest frequency band.
3. a method for inter-channel rate allocation in digital audio coding, comprising the steps of:
S1, analyzing the channel characteristics of the input multi-channel audio signal to obtain channel configuration information;
S2, according to the sound channel configuration information, on the basis of the average distribution code rate, adjusting different weight coefficients of the code rate of each sound channel;
s3, carrying out global bit distribution based on the adjusted code rate of each sound channel;
wherein, for a5.1 channel audio signal, the adjusting of the code rate in step S2 includes: giving higher weight coefficients to the L channel and the R channel of the front sound field than to the LS channel and the RS channel of the rear sound field; for the center channel, giving smaller weight coefficient compared with other channels when the total coding rate is higher, and giving higher weight coefficient compared with other channels when the total coding rate is lower;
for a 3D multi-channel audio signal, the adjusting of the code rate in step S2 includes: the weight coefficient of the middle layer channel is higher than that of the top layer channel, the weight coefficient of the top layer channel is higher than that of the bottom layer channel, and the weight coefficient of the front channel is higher than that of the rear channel;
in the case where the input multi-channel audio signal includes a target signal, the step S1 further includes: analyzing the characteristics of the target signal to obtain target signal description information; the step S2 further includes: and determining the weight coefficient of the target signal code rate allocation based on the target signal description information.
4. the method of claim 3, wherein for the case that the input multi-channel audio signal includes a target signal, the determining the weight coefficients of the code rate allocation of the target signal in step S2 further comprises:
when the target signal is the accompanying sound of different languages, giving the same weight coefficient as the central sound track;
And when the target signal is a directional active target signal, a weight coefficient is distributed to a code rate lower than that of the sound channel signal.
5. A method for allocating code rate in intensity stereo coding of digital audio, characterized by comprising the steps of:
S1, selecting a specific weight coefficient to carry out self-adaptive adjustment on the masking threshold of the mixed high-frequency signal part by carrying out signal analysis on the mixed high-frequency signal part;
and S2, performing free competition code rate allocation based on the adjusted masking threshold.
6. the method according to claim 5, wherein the weighting coefficients are selected in step S1 based on the analysis of the mixed high frequency signal, wherein the higher the high frequency components, the higher the weighting coefficients.
7. An apparatus for in-channel code rate allocation in digital audio coding, comprising:
A masking threshold adjusting module, for selecting a group of specific adjusting coefficients to perform adaptive adjustment on the masking threshold of each sub-band from low frequency to high frequency in a channel based on personal insensitivity to distortion or based on the type of the input audio signal;
And the bit allocation module is used for carrying out global bit allocation in the sound channel based on the adjusted masking threshold.
8. an apparatus for inter-channel rate allocation in digital audio coding, comprising:
the analysis module is used for carrying out sound channel characteristic analysis on the input multi-channel audio signal to obtain sound channel configuration information;
The code rate adjusting module is used for adjusting different weight coefficients of the code rate of each sound channel on the basis of the average allocated code rate according to the sound channel configuration information;
a bit allocation module for performing global bit allocation based on the adjusted code rate of each sound channel;
Wherein, for 5.1 sound channel audio signals, the code rate adjustment performed by the code rate adjustment module comprises: giving higher weight coefficients to the L channel and the R channel of the front sound field than to the LS channel and the RS channel of the rear sound field; for the center channel, giving smaller weight coefficient compared with other channels when the total coding rate is higher, and giving higher weight coefficient compared with other channels when the total coding rate is lower;
For a 3D multi-channel audio signal, the rate adjustment performed by the rate adjustment module comprises: the weight coefficient of the middle layer channel is higher than that of the top layer channel, the weight coefficient of the top layer channel is higher than that of the bottom layer channel, and the weight coefficient of the front channel is higher than that of the rear channel;
for the condition that the input multi-channel audio signal comprises a target signal, the analysis module further analyzes the characteristics of the target signal to obtain the description information of the target signal; the code rate adjustment performed by the code rate adjustment module further comprises: and determining the weight coefficient of the target signal code rate allocation based on the target signal description information.
9. an apparatus for rate allocation in intensity stereo coding of digital audio, comprising:
The masking threshold adjusting module is used for selecting a specific weight coefficient to carry out self-adaptive adjustment on the masking threshold of the mixed high-frequency signal part by carrying out signal analysis on the mixed high-frequency signal part;
and the code rate allocation module is used for allocating free competition code rates based on the adjusted masking threshold.
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