CN101409808A - Method and apparatus for re-sampling audio, and digital television chip - Google Patents
Method and apparatus for re-sampling audio, and digital television chip Download PDFInfo
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Abstract
The invention provides an audio resampling method which comprises as follows: judging whether the caching length of audio data exceeds a normal range; if yes, then raising the number of the sampling points when the caching length of audio data is smaller than the lower threshold of the normal range, calculating the upper resampling parameters and skipping to an upper resampling state; when the caching length of audio data is larger than the upper threshold of the normal range, reducing the number of the sampling points, calculating the lower resampling parameters and skipping to a lower resampling state; carrying out audio resampling under the upper resampling state according to the sampling number and the resampling parameters which are raised; carrying out audio resampling under the lower resampling state according to the sampling numbers and the resampling parameters which are reduced; judging whether the difference value between the current caching length and a standard caching length achieves a normal parameter range or not, if yes, then skipping to the normal state; if not, then circulating on the current state. In the invention, a software method is used to carry out resampling treatment on the audio data which can be integrated into a terminal chip to save the power consumption of a system.
Description
Technical field
The present invention relates to the China mobile multi-media broadcasting technology field, particularly relate to a kind of audio resampling method and device of mobile multimedia broadcast system, and a kind of digital television chip that provides China Mobile multimedia broadcasting CMMB to receive.
Background technology
Mobile Multimedia Broadcasting is emerging a kind of Play Technique for Multimedia, in recent years, this technology and standard evolution are rapid, for example the digital video broadcast-handheld formula in Europe receives (DVB-H, DigitalVideo Broadcast Handheld), the single forward link (MediaFLO of the medium of the U.S., Media Forward Link Only) and the mobile multimedia broadcast system (CMMB, ChinaMobile Multimedia Broadcast) etc. of China.Mobile Multimedia Broadcasting can provide many service content for its terminal use, as programs such as advertisement, weather forecast, news, sports cast, variety, multimedia broadcasting play, films, the user can select the business of Mobile Multimedia Broadcasting according to the hobby of oneself.
Use the user of mobile multi-media broadcasting service under the situation of high-speed mobile, to watch TV by hand-held terminal with portable terminal.The user receives programme, can select the channel of oneself having the right to watch, receives the multi-medium data of selecting channel, thereby is implemented in the TV programme of seeing own selection on the portable terminal.But, in digital television broadcasting system,,, cause the buffer overflow or the underflow of receiving terminal along with the accumulation of time because transmitting terminal and receiving terminal clock frequency is asynchronous, it is discontinuous finally to cause audio frequency and video to be play.At this phenomenon, people have adopted the whole bag of tricks to handle.For video data,, therefore, adopt delay or frame-skipping to finely tune mostly and solve because human eye is insensitive to the delay and the frame-skipping broadcast of any 1 frame video data in the continuous video playback.Yet, because people's ear discontinuous more responsive comparatively speaking for sound,, can largely influence auditory effect so simply adopt the processing method of similar video to postpone or frame-skipping comes processing audio data.At present, people mainly adopt two kinds of approach to solve the problem of audio frequency output: the one, and the clock frequency of adjustment audio output interface, make the data frequency of audio frequency output adaptive transmitting terminal, this mode mainly is to adopt VCXO, phase-locked loop to carry out the adjustment of clock frequency in design, but this mode requires the adjustable precision of phase-locked loop higher, realizes that on chip difficulty is big, cost is unfavorable for being integrated in the mobile digital TV terminal chip also than higher simultaneously; The 2nd, outputting audio data is done resampling, just under the constant situation of output equipment frequency, the mode of utilization Digital Signal Processing is dynamically adjusted the sample rate of outputting audio data, with the coupling transmitting terminal.Some chip manufacturers provide professional SRC (Sample Rate Converter) chip solution at present, it is powerful, can accept the asynchronous clock input, realize the conversion of multiple sample rate, but this solution has increased hardware area and cost, and can't be integrated in the terminal chip.
In a word, need the urgent technical problem that solves of those skilled in the art to be exactly: how can effectively solve the discontinuous problem of audio frequency dateout on the terminal chip of mobile multimedia broadcast system, to improve mobile multi-media terminal user's auditory effect.
Summary of the invention
Technical problem to be solved by this invention provides a kind of audio resampling method and device of mobile multimedia broadcast system, can effectively solve the discontinuous problem of audio frequency dateout on the terminal chip of mobile multimedia broadcast system, to improve mobile multi-media terminal user's auditory effect.
Another object of the present invention provides a kind of digital television chip that receives China Mobile multimedia broadcasting CMMB, to realize the concrete application of the present invention at portable terminal.
In order to address the above problem, the embodiment of the invention provides a kind of audio resampling method of mobile multimedia broadcast system, comprising:
Whether the audio data buffer storage length exceeds normal range (NR), if exceed, then when described voice data buffer storage length during less than the lower threshold value of described normal range (NR), raises the sampled point number, calculates the parameter that upwards resamples, and jumps to upwards resampling state; When described voice data buffer storage length during greater than the upper threshold value of described normal range (NR), downward modulation sampled point number calculates downward resampling parameter, jumps to downward resampling state;
Under the resampling state that makes progress, according to sampled point number after raising and the described parameter that upwards resamples voice data is carried out audio resampling and handle; Under downward resampling state, according to sampled point number after the downward modulation and described downward resampling parameter voice data is carried out audio resampling and handle;
Whether the difference of judging current cache length and standard cache length arrives the normal parameter scope, if then jump to normal condition; If not, then circulate in current state.
Preferably, described state redirect is triggered by sync break.
Preferably, in the process of described state redirect, write down to state jump-point or the sync break number of times of upwards resampling from the normal condition jump-point to downward resampling state jump-point from the normal condition jump-point, perhaps, from initial point to state jump-point or the sync break number of times of upwards resampling from initial point to downward resampling state jump-point.
Preferably, described voice data is according to the sampled point piecemeal; At resampling state upwards or downwards under the resampling state, choose the processing that resamples of a part of audio data block by calculating.
Preferably, described resampling parameter comprises resampling piece number, and this resampling piece number is: N metadata cache deflected length doubly is divided by the sync break number of times that is write down, and the back that rounds up obtains.
Accordingly, the embodiment of the invention also provides a kind of audio resampling device of mobile multimedia broadcast system, comprising:
State redirect module, be used for the audio data buffer storage length and whether exceed normal range (NR), if exceed, then when described voice data buffer storage length during less than the lower threshold value of described normal range (NR), raise the sampled point number, calculate the parameter that upwards resamples, jump to upwards resampling state, start the module that upwards resamples; When described voice data buffer storage length during greater than the upper threshold value of described normal range (NR), downward modulation sampled point number calculates downward resampling parameter, jumps to downward resampling state, starts downward resampling module;
The module that upwards resamples is used for according to sampled point number after raising and the described parameter that upwards resamples voice data being carried out audio resampling and handles;
The resampling module is used for according to sampled point number after the downward modulation and described downward resampling parameter voice data being carried out audio resampling and handles downwards;
The recovering state module is used to judge whether the difference of current cache length and standard cache length arrives the normal parameter scope, if then notify status redirect module jumps to normal condition; If not, then circulate in current state.
Preferably, described state redirect is triggered by sync break.
Preferably, described audio resampling device also comprises:
Counting module is used at the process of described state redirect record from normal condition to resampling state upwards or the sync break number of times from normal condition to downward resampling state procedure.
Preferably, described audio resampling device also comprises:
Select the piece module, be used for writing down from the normal condition jump-point to state jump-point or the sync break number of times of upwards resampling from the normal condition jump-point to downward resampling state jump-point in the process of described state redirect, perhaps, from initial point to state jump-point or the sync break number of times of upwards resampling from initial point to downward resampling state jump-point.
Preferably, described resampling parameter comprises resampling piece number, and this resampling piece number is: N metadata cache deflected length doubly is divided by the sync break number of times that is write down, and the back that rounds up obtains.
The embodiment of the invention also provides a kind of digital television chip that receives China Mobile multimedia broadcasting CMMB, and described chip comprises the audio resampling device, and this audio resampling device comprises:
State redirect module, be used for the audio data buffer storage length and whether exceed normal range (NR), if exceed, then when described voice data buffer storage length during less than the lower threshold value of described normal range (NR), raise the sampled point number, calculate the parameter that upwards resamples, jump to upwards resampling state, start the module that upwards resamples; When described voice data buffer storage length during greater than the upper threshold value of described normal range (NR), downward modulation sampled point number calculates downward resampling parameter, jumps to downward resampling state, starts downward resampling module;
The module that upwards resamples is used for according to sampled point number after raising and the described parameter that upwards resamples voice data being carried out audio resampling and handles;
The resampling module is used for according to sampled point number after the downward modulation and described downward resampling parameter voice data being carried out audio resampling and handles downwards;
The recovering state module is used to judge whether the difference of current cache length and standard cache length arrives the normal parameter scope, if then notify status redirect module jumps to normal condition; If not, then circulate in current state.
Preferably, described audio resampling device also comprises:
Counting module, be used for writing down from the normal condition jump-point to state jump-point or the sync break number of times of upwards resampling from the normal condition jump-point to downward resampling state jump-point in the process of described state redirect, perhaps, from initial point to state jump-point or the sync break number of times of upwards resampling from initial point to downward resampling state jump-point.
Preferably, described audio resampling device also comprises:
Select the piece module, be used for, choose the processing that resamples of a part of audio data block by calculating at resampling state upwards or downwards under the resampling state; Described voice data carries out piecemeal according to sampled point.
Preferably, described resampling parameter comprises resampling piece number, and this resampling piece number is: N metadata cache deflected length doubly is divided by the sync break number of times that is write down, and the back that rounds up obtains.
Compared with prior art, the present invention has the following advantages:
At first, the present invention uses software approach to the outputting audio data processing that resamples, hardware to the mobile multimedia broadcast system terminal is changed less, do not increase its hardware area, can be integrated in easily in the terminal chip, efficiently solve that the SRC chip is big because of hardware area for a long time, at PCB (Printed Circuit Board, printed circuit board (PCB)) takies too many space on, and be not suitable for the problem of mobile multimedia broadcast system terminal;
Secondly, the present invention uses the method for piecemeal numeral filtering interpolation to carry out voice data and resamples, each audio data interpolation to a part of piece, this all carries out interpolation with traditional digital interpolative filtering to all outputting audio datas and compares, reduce the operand of filtering interpolation, saved the computing time of CPU and the resource of system;
Then, resample with hardware algorithm realization outputting audio data and to compare, the present invention uses software approach can shorten the R﹠D cycle of mobile terminal chip, and is easy to algorithm is made amendment and safeguarded, realizes simple, and cost is lower.
Description of drawings
Fig. 1 is the flow chart of steps of the audio resampling method embodiment of a kind of mobile multimedia broadcast system of the present invention;
Fig. 2 is the structured flowchart of the audio resampling device embodiment of a kind of mobile multimedia broadcast system of the present invention;
Fig. 3 is that the present invention uses the flow chart of steps that device embodiment shown in Figure 2 carries out audio resampling.
Embodiment
For above-mentioned purpose of the present invention, feature and advantage can be become apparent more, the present invention is further detailed explanation below in conjunction with the drawings and specific embodiments.
One of core idea of the embodiment of the invention is, by the audio resampling State Control, according to the processing that resamples of sampled point number and the resampling parameter voice data after to piecemeal.The present invention uses software approach to the outputting audio data processing that resamples, and does not increase the hardware area of mobile multimedia broadcast system terminal, can be integrated in easily in the terminal chip; And the present invention uses the method for piecemeal numeral filtering interpolation to carry out voice data to resample, and reduced the operand of filtering interpolation only to the audio data interpolation of a part of piece at every turn, saved system power dissipation; In addition, use the software approach R﹠D cycle short, be easy to algorithm is made amendment and safeguarded, realizes simple, cost is lower.
With reference to Fig. 1, show the flow chart of steps of the audio resampling method embodiment of a kind of mobile multimedia broadcast system of the present invention, specifically can may further comprise the steps:
Step 101: according to the sampled point number is the voice data piecemeal;
Receiving terminal at mobile multimedia broadcast system, the channel decoder receive channel passes the multiplexed frame data of coming, multiplexed frame data demultiplexing goes out Voice ﹠ Video compression bit stream and other information, and sound, video bit stream carry out sound after by sound, video decoder decodes, video data is handled.Wherein, audio bitstream is solved corresponding pulse-code modulation (PCM, Pulse CodeModulation) data by audio decoder after, can carry out piecemeal according to audio frequency frame length and resample points number.In the present embodiment, adopt 64 interpolation filter to carry out interpolation, its interpolation that makes progress can generate 65 sampled points, and interpolation can generate 63 sampled points downwards.The present embodiment voice data uses Advanced Audio Coding (AAC, Advanced Audio Coding) form, and the one frame length is 1024 PCM data, is unit with 64 sampled points, a frame voice data can be divided into 16.In actual applications, those skilled in the art can adopt other suitable digital interpolative filter to carry out interpolation and piecemeal according to the audio data format of reality, and the present invention need not this is made restriction.
After to the voice data piecemeal, when needs resample to be handled, need only be the processing that resamples of a part of audio data block wherein, and needn't all voice datas all be resampled that this greatly reduces the operand of filtering interpolation, has saved overhead.
Step 102: enter normal condition, initialization resampling parameter;
Normal condition is meant near the acceptable scope of the voice data buffer storage length standard audio buffer storage length in the audio data buffer and fluctuates that in this scope, voice data can be play glibly.Wherein, can be the initial point of voice data with standard audio buffer storage length value, with the central point of this initial point as described scope.In the present embodiment, with the beginning buffer memory one second PCM buffered data length be the standard audio buffer storage length, and with this length value as initial point.System adopts the dual track audio plays, sampled point precision 16 bits (two bytes), and sample rate is 48Kbyte/s, then one second PCM buffered data length is 48000 * 2 * 2 * 1, i.e. 192000 bytes.In addition, the channel decoder per second receives that the data of channel can produce a sync break later, and present embodiment utilizes the computing cycle of sync break as the voice data buffer storage length, and every interruption once, calculate one time the voice data buffer storage length, and the number of times of the sync break of record generation simultaneously.When the voice data buffer storage length fluctuateed near the standard audio buffer storage length, system circulated in normal condition, and waited for that sync break triggers next time.
The resampling parameter can comprise resampling piece number and resampling direction, and this step all is initialized as 0 with them.
Step 103: after sync break triggered, whether the audio data buffer storage length exceeded normal range (NR), if do not exceed, then circulates in normal condition; If exceed, then, raise the sampled point number when voice data buffer storage length during less than the lower threshold value of described normal range (NR), calculate the parameter that upwards resamples, jump to upwards resampling state, execution in step 104; When greater than the upper threshold value of described normal range (NR), downward modulation sampled point number calculates downward resampling parameter, jumps to downward resampling state, execution in step 105;
After sync break triggers, calculate the voice data buffer storage length, under the normal play situation, the voice data buffer storage length fluctuates near the standard audio buffer storage length, does not exceed normal range (NR), circulates in normal condition.If the clock frequency of the clock frequency of transmitting terminal and receiving terminal is inconsistent, be played to certain moment, the voice data buffer storage length will exceed the threshold value of normal range (NR), need carry out respective handling.When the clock frequency ratio receiving terminal clock frequency of transmitting terminal is low, be played to certain moment, the voice data buffer memory meets and exceeds the lower threshold value of normal range (NR), need jump to upwards the processing that resamples that make progress of resampling state; When the clock frequency of the clock frequency ratio receiving terminal of transmitting terminal is high, be played to certain moment, the voice data buffer memory meets and exceeds the upper threshold value of normal range (NR), then needs to jump to the processing that resamples of downward resampling state downwards.
When voice data buffer storage length during less than the lower threshold value of described normal range (NR), raise the sampled point number, raising the sampled point number in the present embodiment is 65; When greater than the upper threshold value of described normal range (NR), downward modulation sampled point number, reducing the sampled point number in the present embodiment is 63.
Simultaneously, when voice data buffer storage length overrun takes place, need to calculate the resampling parameter, carry out re-sampling operations according to the resampling parameter.Be well known that the speed of resampling should be higher than the speed of clock skew, in the present embodiment, the resampling speed of employing is 2 times of clock skew speed, and being provided with of resampling parameter is as follows:
When the lower threshold value of voice data buffer storage length less than normal range (NR),
Resampling piece number=((standard audio buffer storage length-current voice data buffer storage length) ÷ sync break number of times) * 2, wherein synchronization times is the sync break number of times from the normal condition jump-point to the state jump-point that upwards resamples, if the voice data buffer storage length is overrun first, then synchronization times is the sync break number of times from initial point to the state jump-point that upwards resamples; The resampling direction is set for making progress;
When the upper threshold value of voice data buffer storage length greater than normal range (NR),
Resampling piece number=((current voice data buffer storage length-standard audio buffer storage length) ÷ sync break number of times) * 2, wherein synchronization times is the sync break number of times from the normal condition jump-point to downward resampling state jump-point, if the voice data buffer storage length is overrun first, then synchronization times is the sync break number of times from initial point to downward resampling state jump-point; The resampling direction is set for downwards.
Need to prove, when calculating resampling piece is counted, must round up, and be limited in 1 in the scope of piecemeal piece number, in the present embodiment, promptly be limited in the scope of [1,16] the result.
Step 104:, according to the sampled point number after raising and the parameter that upwards resamples voice data is carried out audio resampling and handle execution in step 106 at the resampling state that makes progress;
When voice data buffer storage length during less than the lower threshold value of normal range (NR), jump to upwards resampling state, carry out audio resampling according to the parameter that upwards resamples that calculates in the step 104, present embodiment uses 65 sampled points, to preceding several audio data blocks of resampling piece interpolation that makes progress, get back to normal range (NR) up to the voice data buffer storage length.
Step 105:, voice data is carried out the audio resampling processing according to sampled point number after the downward modulation and downward resampling parameter at downward resampling state;
When voice data buffer storage length during greater than the upper threshold value of normal range (NR), jump to downward resampling state, carry out audio resampling according to the downward resampling parameter that calculates in the step 104, present embodiment uses 63 sampled points, preceding several audio data blocks of resampling piece are carried out downward interpolation, get back to normal range (NR) up to the voice data buffer storage length.
Step 106: sync break judges whether the difference of current cache length and standard cache length reaches normal range (NR), if then jump to normal condition after triggering; If not, then circulate in current state.
After sync break triggers, calculate current voice data buffer storage length, if the difference of current voice data buffer storage length and standard audio buffer storage length reaches normal range (NR), represent that it is near the standard audio buffer storage length, then to jump to normal condition that current voice data buffer storage length has been got back to the buffering area central point; If still do not reach, then, proceed audio resampling according to the new resampling parameter of calculating and handle, up to reaching normal range (NR) in current upwards the resampling state or the cycle of states that resamples downwards.In the present embodiment, with 4096 bytes, promptly the length of 1 frame AAC voice data is normal range (NR), when current voice data buffer storage length-192000|＜4096, jumps to normal condition.Certainly, those skilled in the art can set other parameter area that adapts with actual conditions in application, and the present invention need not this is made restriction.
Need to prove, for above-mentioned method embodiment, for simple description, so it all is expressed as a series of combination of actions, but those skilled in the art should know, the present invention is not subjected to the restriction of described sequence of movement, because according to the present invention, some step can adopt other orders or carry out simultaneously.Secondly, those skilled in the art also should know, the embodiment described in the specification all belongs to preferred embodiment, and related action and module might not be that the present invention is necessary.
With reference to Fig. 2, show the structured flowchart of the audio resampling device embodiment of a kind of mobile multimedia broadcast system of the present invention, can comprise with lower module:
Wherein, described state redirect is triggered by sync break.
The module 202 that upwards resamples is used for according to the sampled point number after raising and the parameter that upwards resamples voice data being carried out the audio resampling processing;
Wherein, the described parameter that upwards resamples comprises resampling piece number and resampling direction.Described resampling piece number is: N metadata cache deflected length doubly is divided by the sync break number of times that is write down, and the back that rounds up obtains.Described resampling direction is for upwards resampling and resampling downwards.
Wherein, described downward resampling parameter comprises resampling piece number and resampling direction, and described resampling piece number is identical with the module 202 that upwards resamples with the setting of resampling direction.
Recovering state module 204 is used to judge whether the difference of current cache length and standard cache length arrives normal range (NR), if then notify status redirect module jumps to normal condition; If not, then circulate in current state.
Wherein, described state redirect is triggered by sync break.
Preferably, the device of present embodiment can also comprise:
Counting module, be used for writing down from the normal condition jump-point to state jump-point or the sync break number of times of upwards resampling from the normal condition jump-point to downward resampling state jump-point in the process of described state redirect, perhaps, from initial point to state jump-point or the sync break number of times of upwards resampling from initial point to downward resampling state jump-point.
Preferably, the device of present embodiment can also comprise:
Select the piece module, be used for, choose the processing that resamples of a part of audio data block by calculating at resampling state upwards or downwards under the resampling state; Described voice data carries out piecemeal according to sampled point.
With reference to Fig. 3, show the present invention and use the flow chart of steps that device embodiment shown in Figure 2 carries out audio resampling, specifically can may further comprise the steps:
Step 301: according to the sampled point number is the voice data piecemeal;
Present embodiment adopts 64 interpolation filter that AAC format audio data are carried out interpolation, and a frame voice data is divided into 16.
Step 302: enter normal condition, initialization resampling parameter;
System enters normal condition, and promptly resample piece number and resampling direction of initialization resampling parameter is 0, and waits for that sync break triggers.
Step 303: after sync break triggered, whether state redirect module audio data buffer storage length exceeded normal range (NR); If do not exceed, then circulate in normal condition; If exceed, then, raise the sampled point number when described voice data buffer storage length during less than the lower threshold value of described normal range (NR), calculate the parameter that upwards resamples, jump to upwards resampling state, start the module that upwards resamples; When described voice data buffer storage length during greater than the upper threshold value of described normal range (NR), downward modulation sampled point number calculates downward resampling parameter, jumps to downward resampling state, starts downward resampling module.
In this step, when voice data buffer storage length during, raise the sampled point number less than the lower threshold value of described normal range (NR), raising the sampled point number in the present embodiment is 65; When greater than the upper threshold value of described normal range (NR), downward modulation sampled point number, reducing the sampled point number in the present embodiment is 63.
The resampling parameter is provided with as follows:
When the lower threshold value of voice data buffer storage length less than normal range (NR),
Resampling piece number=((standard audio buffer storage length-current voice data buffer storage length) ÷ sync break number of times) * 2, wherein synchronization times is the sync break number of times from the normal condition jump-point to the state jump-point that upwards resamples, if the voice data buffer storage length is overrun first, then synchronization times is the sync break number of times from initial point to the state jump-point that upwards resamples; The resampling direction is set for making progress;
When the upper threshold value of voice data buffer storage length greater than normal range (NR),
Resampling piece number=((current voice data buffer storage length-standard audio buffer storage length) ÷ sync break number of times) * 2, wherein synchronization times is the sync break number of times from the normal condition jump-point to downward resampling state jump-point, if the voice data buffer storage length is overrun first, then synchronization times is the sync break number of times from initial point to downward resampling state jump-point; The resampling direction is set for downwards.
Need to prove, when calculating resampling piece is counted, must round up, and be limited in 1 in the scope of piecemeal piece number, in the present embodiment, promptly be limited in the scope of [1,16] the result.In resampling CALCULATION OF PARAMETERS process, called counting module, so that the sync break number of times to be provided; The resampling parameter is counted result of calculation and is transferred to and select the piece resume module.
Step 304: the module that upwards resamples when upwards resampling state, is carried out audio resampling processing, execution in step 306 according to the sampled point number after raising and the parameter that upwards resamples to voice data at the Audio Processing state;
Present embodiment uses 65 sampled points, to preceding several audio data blocks of resampling piece that select the piece module to select interpolation that makes progress, gets back to normal range (NR) up to the voice data buffer storage length.
Step 305: when the module that resamples downwards is downward resampling state at the Audio Processing state, according to the sampled point number after the downward modulation and downwards the resampling parameter voice data carried out audio resampling handle;
Present embodiment uses 63 sampled points, and preceding several audio data blocks of resampling piece that select the piece module to select are carried out downward interpolation, gets back to normal range (NR) up to the voice data buffer storage length.
Step 306: after sync break triggered, the recovering state module judged whether the difference of current cache length and standard cache length reaches normal range (NR), if then jump to normal condition; If not, then circulate in current state.
In the present embodiment, when | current voice data buffer storage length-192000|＜4096, jump to normal condition; Otherwise,, proceed audio resampling according to the new resampling parameter of calculating and handle, up to reaching normal range (NR) in current upwards the resampling state or the cycle of states that resamples downwards.
Because embodiment shown in Figure 3 can correspondence be applicable among the aforesaid method embodiment that so description is comparatively simple, not detailed part can be referring to the description of this specification front appropriate section.
According to embodiments of the invention, the digital television chip that a kind of CMMB of providing receives also is provided, in this digital television chip, comprise audio resampling device described above.Because the audio resampling device has been done detailed description, at this, comprises that the digital television chip of this audio resampling device repeats no more.
More than the audio resampling method and the device of a kind of mobile multimedia broadcast system provided by the present invention is described in detail, used specific case herein core idea of the present invention and execution mode are set forth, the explanation of above embodiment just is used for helping to understand method of the present invention and core concept thereof; Simultaneously, for one of ordinary skill in the art, according to thought of the present invention, the part that all can change in specific embodiments and applications, in sum, this description should not be construed as limitation of the present invention.
Claims (14)
1, a kind of audio resampling method of mobile multimedia broadcast system is characterized in that, may further comprise the steps:
Whether the audio data buffer storage length exceeds normal range (NR), if exceed, then when described voice data buffer storage length during less than the lower threshold value of described normal range (NR), raises the sampled point number, calculates the parameter that upwards resamples, and jumps to upwards resampling state; When described voice data buffer storage length during greater than the upper threshold value of described normal range (NR), downward modulation sampled point number calculates downward resampling parameter, jumps to downward resampling state;
Under the resampling state that makes progress, according to sampled point number after raising and the described parameter that upwards resamples voice data is carried out audio resampling and handle; Under downward resampling state, according to sampled point number after the downward modulation and described downward resampling parameter voice data is carried out audio resampling and handle;
Whether the difference of judging current cache length and standard cache length arrives normal range (NR), if then jump to normal condition; If not, then circulate in current state.
2, method according to claim 1 is characterized in that, described state redirect is triggered by sync break.
3, method according to claim 2 is characterized in that, also comprises: record in the process of described state redirect
From the normal condition jump-point to state jump-point or the sync break number of times of upwards resampling from the normal condition jump-point to downward resampling state jump-point,
Perhaps, from initial point to state jump-point or the sync break number of times of upwards resampling from initial point to downward resampling state jump-point.
4, according to claim 1,2 or 3 described methods, it is characterized in that, also comprise:
Described voice data is according to the sampled point piecemeal;
At resampling state upwards or downwards under the resampling state, choose the processing that resamples of a part of audio data block by calculating.
5, method according to claim 4 is characterized in that, described resampling parameter comprises resampling piece number, and this resampling piece number is: N metadata cache deflected length doubly is divided by the sync break number of times that is write down, and the back that rounds up obtains.
6, a kind of audio resampling device of mobile multimedia broadcast system is characterized in that, comprising:
State redirect module, be used for the audio data buffer storage length and whether exceed normal range (NR), if exceed, then when described voice data buffer storage length during less than the lower threshold value of described normal range (NR), raise the sampled point number, calculate the parameter that upwards resamples, jump to upwards resampling state, start the module that upwards resamples; When described voice data buffer storage length during greater than the upper threshold value of described normal range (NR), downward modulation sampled point number calculates downward resampling parameter, jumps to downward resampling state, starts downward resampling module;
The module that upwards resamples is used for according to sampled point number after raising and the described parameter that upwards resamples voice data being carried out audio resampling and handles;
The resampling module is used for according to sampled point number after the downward modulation and described downward resampling parameter voice data being carried out audio resampling and handles downwards;
The recovering state module is used to judge whether the difference of current cache length and standard cache length arrives the normal parameter scope, if then notify status redirect module jumps to normal condition; If not, then circulate in current state.
7, device according to claim 6 is characterized in that, described state redirect is triggered by sync break.
8, device according to claim 7 is characterized in that, also comprises:
Counting module is used for the process record in described state redirect
From the normal condition jump-point to state jump-point or the sync break number of times of upwards resampling from the normal condition jump-point to downward resampling state jump-point,
Perhaps, from initial point to state jump-point or the sync break number of times of upwards resampling from initial point to downward resampling state jump-point.
9, according to claim 6,7 or 8 described devices, it is characterized in that, also comprise:
Select the piece module, be used for, choose the processing that resamples of a part of audio data block by calculating at resampling state upwards or downwards under the resampling state; Described voice data carries out piecemeal according to sampled point.
10, device according to claim 9 is characterized in that, described resampling parameter comprises resampling piece number, and this resampling piece number is: N metadata cache deflected length doubly is divided by the sync break number of times that is write down, and the back that rounds up obtains.
11, a kind of digital television chip that provides China Mobile multimedia broadcasting CMMB to receive is characterized in that described chip comprises the audio resampling device, and described audio resampling device comprises:
State redirect module, be used for the audio data buffer storage length and whether exceed normal range (NR), if exceed, then when described voice data buffer storage length during less than the lower threshold value of described normal range (NR), raise the sampled point number, calculate the parameter that upwards resamples, jump to upwards resampling state, start the module that upwards resamples; When described voice data buffer storage length during greater than the upper threshold value of described normal range (NR), downward modulation sampled point number calculates downward resampling parameter, jumps to downward resampling state, starts downward resampling module;
The module that upwards resamples is used for according to sampled point number after raising and the described parameter that upwards resamples voice data being carried out audio resampling and handles;
The resampling module is used for according to sampled point number after the downward modulation and described downward resampling parameter voice data being carried out audio resampling and handles downwards;
The recovering state module is used to judge whether the difference of current cache length and standard cache length arrives the normal parameter scope, if then notify status redirect module jumps to normal condition; If not, then circulate in current state.
12, digital television chip according to claim 11 is characterized in that, described audio resampling device also comprises:
Counting module is used for the process record in described state redirect
From the normal condition jump-point to state jump-point or the sync break number of times of upwards resampling from the normal condition jump-point to downward resampling state jump-point,
Perhaps, from initial point to state jump-point or the sync break number of times of upwards resampling from initial point to downward resampling state jump-point.
According to claim 11 or 12 described digital television chips, it is characterized in that 13, described audio resampling device also comprises:
Select the piece module, be used for, choose the processing that resamples of a part of audio data block by calculating at resampling state upwards or downwards under the resampling state; Described voice data carries out piecemeal according to sampled point.
14, digital television chip according to claim 13 is characterized in that, described resampling parameter comprises resampling piece number, and this resampling piece number is: N metadata cache deflected length doubly is divided by the sync break number of times that is write down, and the back that rounds up obtains.
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| CN104123943A (en) * | 2013-04-28 | 2014-10-29 | 安凯(广州)微电子技术有限公司 | Audio signal resampling method and apparatus |
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