CN109862442B - Input stream processing method and processing device based on IP transmission - Google Patents
Input stream processing method and processing device based on IP transmission Download PDFInfo
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Abstract
The invention relates to the technical field of IP transmission, and particularly discloses an input stream processing method and device based on IP transmission, which comprises the steps of adjusting the length of a time slice of an input stream to a time slice with a preset length, and obtaining the average code rate delta Ri of the time slice with the preset length; the time length of a time slice with a preset length is reduced gradually, and the fluctuation of the average code rate Delta Ri of the time slice with the preset length is monitored at the same time; if the fluctuation of the average code rate delta Ri of the preset length time slice of the current time length exceeds a preset threshold, stopping reducing the time length of the preset length time slice, and setting the preset length time slice of the previous time length as a minimum stable time slice; the channel cache space is adjusted according to the basic characteristics of the input stream and the minimum stable time slice, the invention can provide a channel code rate allocation mechanism matched with compensation, reasonable space allocation is realized in a limited space, the code rate of an encoder is well restored, and the decoding end can be ensured to perform audio and video decoding stably and reliably.
Description
Technical Field
The invention relates to the technical field of IP transmission, in particular to an input stream processing method and device based on IP transmission.
Background
With the development of scientific technology, the application of the IP transmission mode is more and more extensive, for video and audio communication based on the IP transmission, the encoding is usually performed by an encoder and the encoded video and audio communication is transmitted to a decoding end for decoding, however, due to uncertain delay and jitter existing in the IP transmission, the decoding end cannot perform stable and reliable audio and video decoding, and the IP transmission cannot perform TR101290 analysis on PCR jitter and PCR drift on an IP-based data stream normally through an analyzer, and cannot support a system for decoding for clock recovery well.
Disclosure of Invention
The invention provides an input stream processing method and device based on IP transmission, which can well restore the code rate of an encoder and ensure that a decoding end can stably and reliably decode audio and video.
The specific scheme provided by the invention is as follows:
an input stream processing method based on IP transmission comprises the following steps:
adjusting the time slice length of the input stream to a time slice with a preset length to obtain the average code rate delta Ri of the time slice with the preset length;
successively reducing the time length of the time slices with the preset length, and simultaneously monitoring the fluctuation of the average code rate Delta Ri of the time slices with the preset length;
if the fluctuation of the average code rate delta Ri of the preset length time slice of the current time length exceeds a preset threshold, stopping reducing the time length of the preset length time slice, and setting the preset length time slice of the previous time length as a minimum stable time slice;
the channel buffer space is adjusted according to the basic characteristics of the input stream and the minimum stable time slice.
Optionally, the calculating and obtaining the average code rate Δ Ri of the time slice with the preset length specifically includes:
averagely cutting the preset length time slice into a plurality of preset length time slices to be counted, calculating and obtaining code rates of a plurality of target time slices to be counted, and calculating and obtaining the average code rate of the preset length time slices according to the plurality of code rates.
Optionally, the adjusting the channel buffer space according to the basic characteristic of the input stream and the minimum stable time slice specifically includes:
obtaining a cache coefficient N according to the basic characteristics of the input stream, wherein the cache coefficient N is a fixed parameter;
and adjusting the channel cache space by combining the cache coefficient N and the minimum stable time slice.
Optionally, the average code rate of the minimum stable time slice is used as the pre-recovery buffer output code rate Δ Ro.
Optionally, the method further includes:
calculating and obtaining the difference value of the output code rate Delta Ro of the pre-recovery cache and the average code rate Delta Ri of the time slices with the preset length;
obtaining cache position information according to the position of the central line of the cache space;
and calculating to obtain a cache adjustment compensation value according to the difference value and the cache position information.
Optionally, the method further includes:
acquiring PCR information, checking PCR interval and precision, and calculating to obtain a drift compensation value based on the TR101290 standard.
Optionally, the method further includes:
setting channel code rate sample information, burst sample information and channel change rate sample information;
calling channel code rate sample information and matching the code rate of the current channel;
calling burst sample information and an input sample for matching;
calling channel change rate information and an input code rate change rate for matching;
and selecting a corresponding time slice and issuing time slice parameters according to the matching result.
Optionally, the method includes:
the device comprises an IP code rate pre-recovery module and a drift removal control module;
the IP code rate pre-recovery module is used for acquiring the output code rate Delta Ro of the pre-recovery cache and dividing the cache space for each IP channel; and
and the drift removal control module judges the input stream and adopts a cache adjustment compensation method or a drift compensation method according to the judgment result.
Optionally, the apparatus further includes a comprehensive threshold determination module and a fast recovery module;
the comprehensive threshold value judging module is used for updating channel code rate sample information, burst sample information and channel change rate sample information; and
and the quick recovery module is used for selecting a corresponding time slice and issuing time slice parameters according to the matching results of the channel code rate sample information and the code rate of the current channel, the burst sample information and the input sample, and the channel change rate sample information and the input code rate change rate.
Optionally, the IP code rate pre-recovery module is further configured to generate a burst early warning signal, and send the burst early warning information to the comprehensive threshold determination module.
The invention has the beneficial effects that: the invention can provide a channel code rate distribution mechanism of matching compensation, reasonable space distribution is realized in a limited space, the code rate of an encoder is well restored, and a decoding end can be ensured to stably and reliably decode audio and video.
Drawings
Fig. 1 is a flowchart of an input stream processing method based on IP transmission according to an embodiment of the present invention;
FIG. 2 is a flow chart of two other methods for processing input streams based on IP transmission according to embodiments of the present invention;
fig. 3 is a flow chart of burst early warning recovery provided in the embodiment of the present invention;
fig. 4 is a diagram of an application scenario of an IP transmission-based input stream processing apparatus provided in an embodiment of the present invention;
fig. 5 is an application scenario diagram of the comprehensive threshold determination module and the fast recovery module provided in the embodiment of the present invention.
Detailed Description
In order to explain the technical solution of the present invention in detail, the technical solution of the embodiment of the present invention will be clearly and completely described below with reference to the accompanying drawings of the embodiment of the present invention. It is to be understood that the embodiments described are only a few embodiments of the present invention, and not all embodiments. All other embodiments, which can be derived by a person skilled in the art from the described embodiments of the invention without any inventive step, are within the scope of protection of the invention.
Unless defined otherwise, all technical and scientific terms used herein have the same meaning as commonly understood by one of ordinary skill in the art to which this invention belongs. The terminology used herein is for the purpose of describing particular embodiments only and is not intended to be limiting of the invention. As used herein, the term "and/or" includes any and all combinations of one or more of the associated listed items.
Referring to FIG. 1, a flow chart of the method of the present application is shown.
The input stream processing method based on IP transmission comprises the following steps:
101. and adjusting the time slice length of the input stream to a time slice with a preset length to obtain the average code rate delta Ri of the time slice with the preset length.
Specifically, after the channel is started, the data receiving state starts to operate, the length of the time slice is adjusted to a preset length time slice, in this embodiment, the time slice is adjusted to a maximum value, for example, 2 seconds, the code rate of each preset length time slice is recorded and compared, and the characteristics of the input stream are analyzed at the beginning stage to determine whether the data stream of the channel conforms to the CBR mode.
102. The time length of the time slice with the preset length is reduced gradually, and the fluctuation of the average code rate Delta Ri of the time slice with the preset length is monitored at the same time.
Specifically, since the length of the time slice is adjusted to the maximum value, the error resilience of the maximum time slice to the channels is obvious, but there is a big problem that the larger the time slice is, which means that it needs more buffers for data storage, when the channels grow exponentially, the code rate that each channel can allocate is very limited, for example, on a 4Gb DDR, if the channels increase to 1024, the maximum space that each channel can allocate is only 4Mb, if there are channels with larger actual code rates, for example, 100Mb/s, the buffer space of 4Mb appears quite limited, so this embodiment provides a time slice-based buffer setting method, whose formula is buffer _ range = N × T Δ Ri, where N represents the buffer coefficient, and T is the minimum stable time slice length at the stable code rate, and delta Ri is the average code rate of the time slices with the preset length, and the minimum stable time slice is obtained, the time slices with the preset length are required to be gradually shortened, and whether the average code rate delta Ri of the time slices with the preset length fluctuates greatly or not is observed.
103. And if the fluctuation of the average code rate Delta Ri of the preset length time slice of the current time length exceeds a preset threshold, stopping reducing the time length of the preset length time slice, and setting the preset length time slice of the previous time length as the minimum stable time slice.
Before the minimum stable time slice is obtained, setting a threshold value of the average code rate Delta Ri of the time slices with preset length, stopping trying when the fluctuation of the value of the average code rate Delta Ri of the time slices with preset length exceeds the preset threshold value, and setting the time slices with preset length of the last time length as the minimum stable time slice.
104. The channel buffer space is adjusted according to the basic characteristics of the input stream and the minimum stable time slice.
In this embodiment, a time slice cutting mode is adopted, for example, a preset length time slice of 2 seconds is cut into 8 parts, each time interval is 125 milliseconds, and since the code rates of each time slice may be different, the code rate change rate in the channel time slice is obtained by selecting the maximum value and the minimum value, and the channel cache space is adjusted according to the code rate change rate, the code rate size and the minimum stable time slice.
In the embodiment, a proper matching parameter is determined to receive the current channel input stream, and good support is provided for the subsequent de-drift processing part.
In some embodiments, 101, adjusting the time slice length of the input stream to a time slice with a preset length to obtain an average code rate Δ Ri of the time slice with the preset length specifically includes:
the method comprises the steps of averagely cutting a preset length time slice into a plurality of preset length time slices to be counted, calculating and obtaining code rates of a plurality of target time slices to be counted, and calculating and obtaining the average code rate delta Ri of the preset length time slices according to the plurality of code rates, wherein if a preset length time slice of 2 seconds is cut into 8 time slices with each time interval of 125 milliseconds, the average code rate delta Ri of the preset length time slices is obtained according to the code rate fluctuation condition of the preset length time slices to be counted in 8 time slices to be counted because the code rates of the preset length time slices to be counted are possibly different.
In some embodiments, the adjusting 104 the channel buffer space according to the basic characteristics of the input stream and the minimum stable time slice specifically includes:
obtaining a cache coefficient N according to the basic characteristics of the input stream, wherein the cache coefficient N is a fixed parameter, adjusting the channel cache space by combining the cache coefficient N and the minimum stable time slice, and the setting of N needs to consider a plurality of aspects and provides a parameter most suitable for the current channel.
According to the embodiment, the division of the buffer space is obtained by obtaining the basic characteristics of the current input stream and selecting the proper minimum stable time slice, so that the buffer problem of the jitter and the drift of the elimination code rate in the input scene based on multi-channel and multi-program multiplexing at present is solved, a channel code rate distribution mechanism matched with smooth compensation is provided, and reasonable space distribution is realized in a limited space.
In some embodiments, the average code rate of the minimum stable time slice is used as the pre-recovery buffer output code rate Δ Ro, and after the minimum stable time slice is selected, the average code rate corresponding to the minimum stable time slice needs to be used as the pre-recovery buffer output code rate Δ Ro and used as the reference synchronization code rate, where the code rate has a code rate R that is similar to that of the sending end before transmission through the IP.
In this embodiment, the output code rate Δ Ro of the pre-recovery buffer after the buffer smoothing has higher accuracy and wider jitter resistance.
In some embodiments, as shown in fig. 2, the method for processing an input stream based on IP transmission further includes:
201. and calculating and obtaining the difference value of the output code rate Delta Ro of the pre-recovery cache and the average code rate Delta Ri of the time slices with the preset length.
Specifically, when the average code rate Δ Ri of a preset-length time slice is greater than the pre-recovery buffer output code rate Δ Ro, the buffer data is increased, which means that the direction of compensation needs to be controlled incrementally, otherwise, the difference between the average code rate Δ Ri of the preset-length time slice and the pre-recovery buffer output code rate Δ Ro needs to be selected as accurately as possible, and therefore, a progressive difference extraction is provided in this embodiment, which is specifically implemented by using a larger time window, for example, a 2-second time window in this embodiment, the code rate Ri first counts the average code rate Δ Ri of the preset-length time slice through the 2-second time window, and then updates the average code rate Δ Ri of the preset-length time slice every 125 milliseconds, so that the average code rate Δ Ri of the preset-length time slice has a jitter removal function of 2 seconds, and the pre-recovery buffer output code rate Δ Ro is obtained in a similar manner, an important benefit of this approach is that the jitter introduced by the input is greatly reduced, reducing the complexity of drift compensation.
202. And obtaining cache position information according to the position of the central line of the cache space.
Specifically, after the system starts to receive the data stream, the start read pointer is set at the position of the middle line of the buffer space, so that the buffer overflows upwards and overflows downwards at relatively close time, and the jitter and the delay are periodic, so that the method can better adopt the buffer to absorb the jitter, the buffer position provides relatively direct information, and the average code rate Δ Ri of the preset-length time slice and the output code rate Δ Ro of the pre-recovery buffer are caused by difference.
203. And calculating to obtain a cache adjustment compensation value according to the difference value and the cache position information.
Because there is a case of multiplexing output at the encoding end, in this embodiment, a method for adjusting compensation based on buffering is provided, and a specific formula thereof is as follows: when Δ Ro >. DELTA Ri, comp = ([ Delta ] Ro- [ Delta ] Ri)/([ Delta buffer _ level) ]; when Δ Ri >. DELTA.Ro, comp = ([ Delta ] Ri-DELTA.Ro)/(buffer _ range- Δ buffer _ level).
In this embodiment, a trend compensation manner is adopted to correct the current pre-recovery cache output code rate Δ Ro, and the specific method is as follows: in this embodiment, the delta buffer _ level is stored once every 125 milliseconds, and the newly counted delta buffer _ level is compared with the last time, and if there is an expansion trend, compensation is started.
In some embodiments, as shown in fig. 2, the method for processing an input stream based on IP transmission further includes: acquiring PCR information, checking PCR interval and precision, and calculating to obtain a drift compensation value based on a TR101290 standard, wherein the method specifically comprises the following steps:
204. and acquiring PCR information.
Specifically, the PID of the PCR of a certain program is locked, and continuously monitored, and the PCR interval and PCR precision information are obtained and checked, where the checking mode in this embodiment is to use a local nominal clock 27Mhz to perform TR101290 analysis on the input stream, so as to obtain the required information.
205. The PCR spacing and accuracy were checked.
Specifically, in the network transmission process, because uncertainty exists and the stability of data is damaged, in this embodiment, the following checking method is adopted by using the formula: Ri/Ti is approximately equal to R (i +1)/T (i +1), wherein R is the code rate between adjacent PCRs, T is the time between adjacent PCRs, and if the ratio of the code rate information between adjacent PCRs to the PCR value is consistent, the current PCR information is reliable and effective.
206. A drift compensation value is calculated.
Specifically, according to the PCR drift determination described in TR101290, a drift compensation value range in unit time is obtained, and this is used as a threshold to correct Ro, and the calculation procedure of this embodiment is as follows: according to the requirement that | dr _ range | < 10ppm/hour, a reference base _ dr and max _ dr when 10ppm drifts are calculated according to TR101290, a driftable difference value is obtained according to dr _ diff = | base _ dr-max _ dr |, the value is used as a control threshold of compensation and is used for filtering an overshoot phenomenon caused by jitter, comp = dr _ diff or comp > dr _ diff is determined, and which compensation mode is more reliable is determined according to the check result of 205.
According to the current cache change trend, the difference value between the average code rate delta Ri and the delta Ro of the current time slice with the preset length is combined, whether the PCR is locked or not is judged, and high-precision compensation values are finally obtained according to information such as whether drift compensation is started or not, and the high-precision compensation values are used for ensuring consistency of the output code rate delta Ro of the pre-recovery cache and the code rate R of the encoder.
In some embodiments, as shown in fig. 3, the method further includes a burst pre-alarm recovery function, such as:
301. and setting channel code rate sample information, burst sample information and channel change rate sample information.
The embodiment is dynamically controlled by obtaining the basic characteristics of the input stream, and because the adaptation of the input stream exists in the control process, in order to quickly recover in an abnormal state, the embodiment provides a quick locking method of a feature table, and the feature table is a set of matching parameters formed in long-term operation according to the current parameters.
302. And calling channel code rate sample information and matching the code rate of the current channel.
And matching the channel code rate sample information with the code rate of the current channel, wherein the input code rate is 1M, the matched sample information is in an interval of 0.5-3M, and the parameter information obtained in the interval is input with the minimum code rate.
303. And calling the burst sample information and the input sample for matching.
And matching the burst sample information with the input sample, for example, no burst interval exists at present, and the parameter information obtained in the interval is the stable code rate.
304. And calling channel change rate information and the input code rate change rate for matching.
And matching the channel change rate information with the input code rate change rate, wherein the current change rate is less than 1%, the code rate fluctuation is small, and the channel state is good.
305. And selecting a corresponding time slice and issuing time slice parameters according to the matching result.
According to the matching result, the minimum stable time slice is matched, for example 125 milliseconds, a proper N-corresponding buffer is allocated to the channel, the units at the bottom layer can be quickly locked and start a correction function after matching information is obtained, and the design can be quickly synchronized when a signal is in short abnormity or interruption.
Referring to fig. 4, a device application scenario diagram of the present application is shown.
The input stream processing device based on IP transmission comprises an IP code rate pre-recovery module and a drift removal control module.
The IP code rate pre-recovery module is used for acquiring the output code rate Delta Ro of the pre-recovery cache and dividing the cache space for each IP channel; setting an output code rate reference value for the average code rate Delta Ri of each time slice according to the code rate statistics based on the time slices; allocating memory according to the code rate, and allocating a cache space based on time slice and code rate matching for each IP channel; and locking the code rate Ri based on the time slice according to a code rate pre-recovery function, and setting a closer pre-recovery cache output code rate Delta Ro according to the cache position of the data.
The drift removal control module judges the input stream, adopts a buffer adjustment compensation method or a drift compensation method according to a judgment result, and starts the drift removal control module to judge according to the input condition because the code rate of IP input is always in change in a certain fixed time period according to the change rate of the input code rate and the condition of available buffer of a current channel.
In some embodiments, as shown in fig. 5, the IP transmission-based input stream processing apparatus further includes a composite threshold determination module and a fast recovery module.
The comprehensive threshold value judging module is used for updating the channel code rate sample information, the burst sample information and the channel change rate sample information.
And the quick recovery module is used for selecting the corresponding time slice and issuing the time slice parameters according to the matching results of the channel code rate sample information and the code rate of the current channel, the burst sample information and the input sample, and the channel change rate sample information and the input code rate change rate.
In this embodiment, for jitter generated by an IP network and jitter generated by an encoder in a burst mode, a receiving code rate is recovered according to an adjusted time slice length, a proper time slice length is selected to monitor a channel code rate, an excessively large time slice easily causes a waste of cache resources, an excessively small time slice has poor anti-jitter performance, a cache space suitable for a current input characteristic can be obtained according to periodicity of the code rate at an encoding end, buffer _ range = N × T Δ Ri, N represents a cache coefficient, T is a minimum stable time slice length under a stable code rate, Ri Δ is an average code rate of a time slice with a preset length, and on the basis, a burst early warning function can analyze whether a relative relationship between Δ Ri and T which change in real time at present is changed or not, so that the cache space needs to be adjusted again.
According to the above explanation, a pre-recovery buffer output code rate Δ Ro can be obtained, which has the characteristic of an average code rate corresponding to the length of an approximate minimum stable time slice, where Ri (t) = R (t) ± d (t) exists in the average code rate Δ Ri of the time slice with the preset length and the code rate R of the encoder, and d (t) is a nonlinear time-varying function, but has the characteristic of a certain range convergence.
Because the pre-recovery buffer output code rate Δ Ro is determined, it can be considered that the current system receiving code rate is close to the coding rate, and then the de-drift control module needs to be started to compensate the code rate, so we need to select a proper way to calculate the compensation value according to the channel program condition.
The compensation of the pre-recovery buffer output code rate Δ Ro needs to refer to several aspects, such as the current buffer data position, the change rate of Δ Ri, the change rate of Δ Ro, and the change rate and direction of PCR precision and PCR drift.
In this embodiment, two adjustment methods for high-precision compensation are provided, one is an adjustment method based on the relative relationship between the buffer location and the rate of change of the input and output within the corresponding time slice, so as to generate the following equation: when Δ Ro >. DELTA Ri, comp = ([ Delta Ro- ] DELTA Ri)/[ Delta (buffer _ level); when the code rate is equal to or greater than the preset threshold value, the adjustment formula is comp = ([ Delta ] Ri-Delta ] Ro)/(buffer _ range- [ Delta ] buffer _ level), PCR is not referred to by the adjustment formula, a linear compensation mode is adopted, dynamic code rate adjustment is carried out on the [ Delta ] Ro under the condition that overflow is not generated in a buffer memory, and the mode has better drift control capability when the multiplexing coded data input at the front end is processed.
Another way to adjust the method is to know, through the TR101290 protocol, that the drift range of the decoding clock needs to satisfy: therefore, according to the information obtained by PCR, the code rate drift value max _ dr when the code rate is 10ppm/hour is obtained through analysis and calculation, so that the largest dr _ diff value which can be adjusted in unit time can be obtained, and the adjustment mode of the compensation value is as follows: when Δ Ro >. DELTA.Ri or comp < dr _ diff, comp = ([ Delta ] Ro- [ Delta ] Ri)/([ Delta buffer _ level); when the data stream is transmitted through the IP, the problem of decoding clock drift caused by jitter after the data stream is transmitted through the IP can be solved better, and the effect is better when a single program channel is used.
In some embodiments, the IP code rate pre-recovery module is further configured to generate a burst warning signal, and send the burst warning signal to the comprehensive threshold determination module.
Specifically, the IP code rate pre-recovery module counts input data according to an adjustable time slice, the length of the time slice is controlled by the comprehensive threshold module, and the purpose is that the fluctuation of the code rate is large in different environments, in a severe environment, in order to predict the change condition of the current channel in advance, a small time slice is required to perform continuous statistics on the code rate, for example, 10 milliseconds is used as a unit, and when the code rate in a certain time slice far exceeds the currently received average code rate, an emergency warning signal is generated and transmitted to the comprehensive threshold module for processing.
The technical features of the embodiments described above may be arbitrarily combined, and for the sake of brevity, all possible combinations of the technical features in the embodiments described above are not described, but should be considered as being within the scope of the present specification as long as there is no contradiction between the combinations of the technical features.
The above-mentioned embodiments only express several embodiments of the present invention, and the description thereof is more specific and detailed, but not to be understood as limiting the scope of the present invention. It should be noted that, for a person skilled in the art, several variations and modifications can be made without departing from the spirit of the invention, which falls within the scope of the invention, and therefore the scope of the invention is to be determined by the appended claims.
Claims (10)
1. An input stream processing method based on IP transmission is characterized by comprising the following steps:
adjusting the time slice length of the input stream to a time slice with a preset length to obtain the average code rate delta Ri of the time slice with the preset length;
the time length of a time slice with a preset length is reduced gradually, and the fluctuation of the average code rate Delta Ri of the time slice with the preset length is monitored at the same time;
if the fluctuation of the average code rate delta Ri of the preset length time slice of the current time length exceeds a preset threshold, stopping reducing the time length of the preset length time slice, and setting the preset length time slice of the previous time length as a minimum stable time slice;
the channel buffer space is adjusted according to the basic characteristics of the input stream and the minimum stable time slice.
2. The method according to claim 1, wherein the obtaining an average code rate Δ Ri for a time slice with a preset length specifically comprises:
averagely cutting the preset length time slice into a plurality of preset length time slices to be counted, calculating and obtaining code rates of a plurality of target time slices to be counted, and calculating and obtaining the average code rate of the preset length time slices according to the plurality of code rates.
3. The method according to claim 1, wherein the adjusting the channel buffer space according to the basic characteristics and the minimum stable time slice of the input stream specifically comprises:
obtaining a cache coefficient N according to the basic characteristics of the input stream, wherein the cache coefficient N is a fixed parameter;
and adjusting the channel cache space by combining the cache coefficient N and the minimum stable time slice.
4. The method of claim 1, wherein the average code rate of the minimum stable time slice is used as a pre-recovery buffer output code rate Δ Ro.
5. The method for processing the input stream based on the IP transmission according to the claim 4, wherein the method further comprises:
calculating and obtaining the difference value between the output code rate Delta Ro of the pre-recovery cache and the average code rate Delta Ri of the time slices with the preset length;
obtaining cache position information according to the position of the central line of the cache space;
and calculating to obtain a cache adjustment compensation value according to the difference value and the cache position information.
6. The method for processing the input stream based on the IP transmission according to claim 1, further comprising:
acquiring PCR information, checking PCR interval and precision, and calculating to obtain a drift compensation value based on the TR101290 standard.
7. The method for processing input stream based on IP transport according to claim 5 or 6, characterized in that it further comprises:
setting channel code rate sample information, burst sample information and channel change rate sample information;
calling channel code rate sample information and matching the code rate of the current channel;
calling burst sample information and an input sample for matching;
calling channel change rate information and an input code rate change rate for matching;
and selecting a corresponding time slice and issuing time slice parameters according to all the matching results.
8. An input stream processing device based on IP transmission, applied to the processing method of claim 7, characterized by comprising:
the device comprises an IP code rate pre-recovery module and a drift removal control module;
the IP code rate pre-recovery module is used for acquiring the output code rate Delta Ro of the pre-recovery cache and dividing the cache space for each IP channel; and
and the drift removal control module judges the input stream and adopts a cache adjustment compensation method or a drift compensation method according to the judgment result.
9. The IP-transport based input stream processing apparatus according to claim 8, further comprising a composite threshold decision module and a fast restoration module;
the comprehensive threshold value judging module is used for updating channel code rate sample information, burst sample information and channel change rate sample information; and
and the quick recovery module is used for selecting a corresponding time slice and issuing time slice parameters according to the matching results of the channel code rate sample information and the code rate of the current channel, the burst sample information and the input sample, and the channel change rate sample information and the input code rate change rate.
10. The IP transmission-based input stream processing apparatus of claim 8, wherein the IP code rate pre-recovery module is further configured to generate a burst warning signal and send the burst warning signal to a comprehensive threshold determination module.
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