CN102446509A - Audio coding and decoding method for enhancing anti-packet loss capability and system thereof - Google Patents
Audio coding and decoding method for enhancing anti-packet loss capability and system thereof Download PDFInfo
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- CN102446509A CN102446509A CN2011103732421A CN201110373242A CN102446509A CN 102446509 A CN102446509 A CN 102446509A CN 2011103732421 A CN2011103732421 A CN 2011103732421A CN 201110373242 A CN201110373242 A CN 201110373242A CN 102446509 A CN102446509 A CN 102446509A
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Abstract
The invention relates to an audio coding and decoding method for enhancing anti-packet loss capability and a system thereof. The method comprises the following steps that: a coding terminal calculates an available space according to the maximum byte number that is allowed to be output by current frame coding and the code stream size generated by the current frame coding; the coding terminal carries out prediction on a next frame according to the available space; and when a decoding terminal detects that the next frame is lost, the decoding terminal carries out reconstruction on the next frame according to a prediction parameter predicted by the coding terminal. According to the invention, bandwidth resources are fully utilized; a spatial difference between the allowed maximum byte number corresponding to each frame and the byte number generated by actual coding is used to carry out prediction on a next frame; when decoding is carried out, a lost packet is reconstructed according to a prediction parameter, thereby improving a state of poor audio frequency due to packet loss and providing a good hearing effect for a user; meanwhile, compatibility with a corresponding standard decoder can be ensured.
Description
Technical field
The present invention relates to the audio encoding and decoding technical field, relate in particular to a kind of audio encoding and decoding method and system that strengthens anti-packet loss.
Background technology
Present main flow audio frequency or audio coder & decoder (codec) are mostly supported the coded system of on-fixed speed.Such as MP3, AAC employings such as (Advanced Audio Coding, Advanced Audio Codings) promptly is the coded system of this on-fixed speed.
Compare with the coded system of fixed rate, one of difference of on-fixed rate coding mode is: the byte number of the coding output of every frame is not fixed, and is less than the byte number of the coded system generation of fixed rate.
At present, when adopting on-fixed rate coding mode to carry out audio coding, easy packet loss in the audio code stream transmission course, thus cause audio frequency effect not good, influence the user experience effect.
Summary of the invention
Fundamental purpose of the present invention is to provide a kind of audio encoding and decoding method and system that strengthens anti-packet loss, to solve the not good problem of audio frequency effect that packet loss causes in the audio code stream transmission course.
In order to achieve the above object, the present invention proposes a kind of audio encoding and decoding method that strengthens anti-packet loss, comprising:
Coding side allows the maximum number of byte of output and the code stream size of present frame coding generation to calculate free space according to the present frame coding;
Said coding side is predicted next frame according to said free space;
When decoding end detected said next frame and loses, said decoding end was rebuild said next frame according to the Prediction Parameters of said coding side prediction.
Preferably, the step next frame predicted according to said free space of said coding side comprises:
Said coding side calculates the said Prediction Parameters relevant with next frame;
Calculate the needed space of said Prediction Parameters;
Said free space and the needed space of said Prediction Parameters are compared;
When said free space is enough, said Prediction Parameters is put into said free space; Otherwise, abort operation.
Preferably, said when decoding end detects said next frame and loses, said decoding end is according to the Prediction Parameters of said coding side prediction, and the step that said next frame is rebuild comprises:
When said decoding end detects said next frame and loses, obtain said Prediction Parameters from said free space storage;
According to said Prediction Parameters said next frame is rebuild.
Preferably, said Prediction Parameters comprises at least: present frame and next frame coding MDCT or the energy difference of adjacent frequency bands, the scale factor of next frame.
Preferably, this method also comprises:
Said decoding end judges whether packet loss according to the number of frames that the RTP head of transmission of audio code stream carries.
The present invention also proposes a kind of audio coding and decoding system that strengthens anti-packet loss, comprising: coding side and decoding end; Wherein:
Said coding side comprises:
Computing module is used for allowing the maximum number of byte of output and the code stream size of present frame coding generation to calculate free space according to the present frame coding;
Prediction module is used for according to said free space next frame being predicted;
Said decoding end is used for when detecting said next frame and lose, and according to the Prediction Parameters of said prediction module prediction, said next frame is rebuild.
Preferably, said prediction module comprises:
Computing unit is used to calculate the said Prediction Parameters relevant with next frame; And the needed space of said Prediction Parameters;
Comparing unit is used for said free space and the needed space of said Prediction Parameters are compared;
Storage unit is used for when said free space is enough, said Prediction Parameters being put into said free space; Otherwise, abort operation.
Preferably, said decoding end also is used for when detecting said next frame and lose, and obtains said Prediction Parameters from said free space storage; According to said Prediction Parameters said next frame is rebuild.
Preferably, said Prediction Parameters comprises at least: present frame and next frame coding MDCT or the energy difference of adjacent frequency bands, the scale factor of next frame.
Preferably, said decoding end also is used for the number of frames that the RTP head according to the transmission of audio code stream carries and judges whether packet loss.
A kind of audio encoding and decoding method and system that strengthens anti-packet loss that the present invention proposes; Make full use of bandwidth resources, utilize the space parallax of the byte number of corresponding permission maximum number of byte of every frame and actual coding generation, next frame is predicted; When decoding; According to Prediction Parameters packet loss is rebuild, thereby improved because the not good situation of audio frequency that packet loss causes gives the user with better auditory effect; Guaranteed compatibility simultaneously with the standard decoder of correspondence.
Description of drawings
Fig. 1 is the schematic flow sheet that the present invention strengthens audio encoding and decoding method one embodiment of anti-packet loss;
Fig. 2 is that the present invention strengthens the schematic flow sheet that coding side is predicted next frame according to free space among audio encoding and decoding method one embodiment of anti-packet loss;
Fig. 3 is the structural representation that the present invention strengthens audio coding and decoding system one embodiment of anti-packet loss;
Fig. 4 is the structural representation that the present invention strengthens coding side among audio coding and decoding system one embodiment of anti-packet loss;
Fig. 5 is the structural representation that the present invention strengthens prediction module among audio coding and decoding system one embodiment of anti-packet loss.
In order to make technical scheme of the present invention clearer, clear, will combine accompanying drawing to do further to detail below.
Embodiment
Embodiment of the invention solution mainly is: make full use of bandwidth resources; With the corresponding permission maximum number of byte of every frame and the space parallax of the byte number of actual coding generation; Next frame is predicted; When decoding, according to Prediction Parameters packet loss is rebuild, to solve the not good problem of audio frequency effect that packet loss causes in the audio code stream transmission course.
As shown in Figure 1, one embodiment of the invention proposes a kind of audio encoding and decoding method that strengthens anti-packet loss, comprising:
Step S101, coding side allows the maximum number of byte of output and the code stream size of present frame coding generation to calculate free space according to the present frame coding;
Present embodiment method running environment relates to coding side and decoding end, and in the audio code stream transmission course, coding side is at first created the buffer memory of present frame and next frame, and the present frame and the next frame that receive are encoded; Then; Allow the maximum number of byte of output and the code stream size of present frame coding generation to calculate free space according to the present frame coding; This free space is meant the space parallax of the maximum number of byte and the byte number that actual coding generates of present frame correspondence; Utilize this space parallax to predict,, thereby improve audio frequency effect so that decoding end restores packet loss as much as possible to next frame.
Step S102, coding side is predicted next frame according to free space;
Coding side is when predicting next frame; Need to calculate the relevant main Prediction Parameters of next frame; Need calculate the required space of using of above-mentioned main Prediction Parameters simultaneously; Then, the free space and the required space of using of Prediction Parameters that obtain according to aforementioned calculation judge whether free space is enough.If enough, coding side then is stored in Prediction Parameters in this free space, otherwise, abort operation.
Above-mentioned Prediction Parameters comprises: the energy difference of present frame and next frame coding MDCT (modified discrete cosine transform improves discrete cosine transform) or adjacent frequency bands, and the scale factor of next frame etc.
Step S103, when decoding end detected next frame and loses, decoding end was rebuild next frame according to the Prediction Parameters of coding side prediction.
In the transmission course of audio code stream; Decoding end is according to RTP (the Real-time Transport Protocol of transmission of audio code stream; Real time transport protocol) number of frames that carries judges whether packet loss; Decoding end detects next frame when losing, and decoding end is rebuild next frame according to Prediction Parameters that store in the free space, the coding side prediction.Though the effect of rebuilding does not reach the effect of primitive frame, the noise of but having avoided packet loss to cause has more significantly improved sound quality.
In the practical implementation process, as shown in Figure 2, above-mentioned steps S102 comprises:
Step S1021, coding side calculates the Prediction Parameters relevant with next frame;
Step S1022 calculates the needed space of Prediction Parameters;
Step S1023 compares free space and the needed space of Prediction Parameters; When free space is enough, get into step S1024; Otherwise, abort operation, process ends;
Step S1024 puts into free space with Prediction Parameters.
Compare prior art; Present embodiment make full use of bandwidth resources and the situation of the storehouse feedback mechanism of need not decoding under, utilize the space parallax of the corresponding maximum number of byte of every frame and the byte number of actual coding generation, next frame is predicted; When decoding, packet loss is rebuild according to these information of forecastings; Reach and improve Network Packet Loss and cause the not good situation of audio frequency effect, give the user with better auditory effect, especially under the situation that is not continuous packet loss; Audio frequency effect improves obvious especially, and can keep compatible with the demoder of standard.
As shown in Figure 3, one embodiment of the invention proposes a kind of audio coding and decoding system that strengthens anti-packet loss, comprising: coding side 301 and decoding end 302; Wherein:
Decoding end 302 is used for when detecting next frame and lose, according to the Prediction Parameters of coding side prediction, next frame being rebuild.
The ultimate principle that present embodiment strengthens the audio coding and decoding system of anti-packet loss is:
In the audio code stream transmission course, coding side 302 is at first created the buffer memory of present frame and next frame, and the present frame and the next frame that receive are encoded; Then; Allow the maximum number of byte of output and the code stream size of present frame coding generation to calculate free space according to the present frame coding; This free space is meant the space parallax of the maximum number of byte and the byte number that actual coding generates of present frame correspondence, and coding side 301 utilizes this space parallax to predict next frame.So that 302 pairs of packet losses of decoding end restore as much as possible, thereby improve audio frequency effect.
Above-mentioned Prediction Parameters comprises: the energy difference of present frame and next frame coding MDCT (modified discrete cosine transform improves discrete cosine transform) or adjacent frequency bands, and the scale factor of next frame etc.
In the transmission course of audio code stream; Decoding end 302 is according to RTP (the Real-time Transport Protocol of transmission of audio code stream; Real time transport protocol) number of frames that carries judges whether packet loss; Decoding end detects next frame when losing, and decoding end 302 is rebuild next frame according to Prediction Parameters that store in the free space, coding side 301 predictions.Though the effect of rebuilding does not reach the effect of primitive frame, the noise of but having avoided packet loss to cause has more significantly improved sound quality.
Particularly, as shown in Figure 4, coding side 301 comprises in the present embodiment: computing module 3011 and prediction module 3012, wherein:
As shown in Figure 5, above-mentioned prediction module 3012 comprises: computing unit 30121, comparing unit 30122 and storage unit 30123, wherein:
Comparing unit 30122 is used for free space and the needed space of Prediction Parameters are compared;
The embodiment of the invention strengthens the audio encoding and decoding method and the system of anti-packet loss, makes full use of bandwidth resources, utilizes the space parallax of the byte number of corresponding permission maximum number of byte of every frame and actual coding generation; Next frame is predicted; When decoding, according to Prediction Parameters packet loss is rebuild, thereby improved because the not good situation of audio frequency that packet loss causes; Give the user with better auditory effect, and improved bandwidth availability ratio; Simultaneously also can be only of the input of the byte number of actual coding as decoding end, can guarantee compatibility thus with the demoder of corresponding standard.
The above is merely the preferred embodiments of the present invention; Be not so limit claim of the present invention; Every equivalent structure or flow process conversion that utilizes instructions of the present invention and accompanying drawing content to be done; Or directly or indirectly be used in other relevant technical field, all in like manner be included in the scope of patent protection of the present invention.
Claims (10)
1. an audio encoding and decoding method that strengthens anti-packet loss is characterized in that, comprising:
Coding side allows the maximum number of byte of output and the code stream size of present frame coding generation to calculate free space according to the present frame coding;
Said coding side is predicted next frame according to said free space;
When decoding end detected said next frame and loses, said decoding end was rebuild said next frame according to the Prediction Parameters of said coding side prediction.
2. method according to claim 1 is characterized in that, the step that said coding side is predicted next frame according to said free space comprises:
Said coding side calculates the said Prediction Parameters relevant with next frame;
Calculate the needed space of said Prediction Parameters;
Said free space and the needed space of said Prediction Parameters are compared;
When said free space is enough, said Prediction Parameters is put into said free space; Otherwise, abort operation.
3. method according to claim 1 is characterized in that, and is said when decoding end detects said next frame and loses, and said decoding end is according to the Prediction Parameters of said coding side prediction, and the step that said next frame is rebuild comprises:
When said decoding end detects said next frame and loses, obtain said Prediction Parameters from said free space storage;
According to said Prediction Parameters said next frame is rebuild.
4. according to claim 1,2 or 3 described methods, it is characterized in that said Prediction Parameters comprises at least: present frame and next frame coding improvement discrete cosine transform MDCT or the energy difference of adjacent frequency bands, the scale factor of next frame.
5. method according to claim 4 is characterized in that, also comprises:
Said decoding end judges whether packet loss according to the number of frames that the RTP head of transmission of audio code stream carries.
6. an audio coding and decoding system that strengthens anti-packet loss is characterized in that, comprising: coding side and decoding end; Wherein:
Said coding side comprises:
Computing module is used for allowing the maximum number of byte of output and the code stream size of present frame coding generation to calculate free space according to the present frame coding;
Prediction module is used for according to said free space next frame being predicted;
Said decoding end is used for when detecting said next frame and lose, and according to the Prediction Parameters of said prediction module prediction, said next frame is rebuild.
7. system according to claim 6 is characterized in that, said prediction module comprises:
Computing unit is used to calculate the said Prediction Parameters relevant with next frame; And the needed space of said Prediction Parameters;
Comparing unit is used for said free space and the needed space of said Prediction Parameters are compared;
Storage unit is used for when said free space is enough, said Prediction Parameters being put into said free space; Otherwise, abort operation.
8. system according to claim 7 is characterized in that, said decoding end also is used for when detecting said next frame and lose, and obtains said Prediction Parameters from said free space storage; According to said Prediction Parameters said next frame is rebuild.
9. according to claim 6,7 or 8 described systems, it is characterized in that said Prediction Parameters comprises at least: present frame and next frame coding MDCT or the energy difference of adjacent frequency bands, the scale factor of next frame.
10. system according to claim 9 is characterized in that, said decoding end also is used for the number of frames that the real time transport protocol RTP head according to the transmission of audio code stream carries and judges whether packet loss.
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| WO2013189428A3 (en) * | 2013-02-22 | 2014-02-20 | 中兴通讯股份有限公司 | Anti-packet-loss real-time communication method, system and related device based on hierarchical coding |
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