EP4381600A1 - Réception et conversion des fréquences d'échantillonnage de données audio et vidéo transmises de façon asynchrone - Google Patents

Réception et conversion des fréquences d'échantillonnage de données audio et vidéo transmises de façon asynchrone

Info

Publication number
EP4381600A1
EP4381600A1 EP22760905.4A EP22760905A EP4381600A1 EP 4381600 A1 EP4381600 A1 EP 4381600A1 EP 22760905 A EP22760905 A EP 22760905A EP 4381600 A1 EP4381600 A1 EP 4381600A1
Authority
EP
European Patent Office
Prior art keywords
data
buffer
sample rate
filter
input data
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Pending
Application number
EP22760905.4A
Other languages
German (de)
English (en)
Inventor
Marc Brunke
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Individual
Original Assignee
Individual
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Individual filed Critical Individual
Publication of EP4381600A1 publication Critical patent/EP4381600A1/fr
Pending legal-status Critical Current

Links

Classifications

    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
    • H03H17/00—Networks using digital techniques
    • H03H17/02—Frequency selective networks
    • H03H17/04—Recursive filters
    • H03H17/0416—Recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing
    • H03H17/0422—Recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing the input and output signals being derived from two separate clocks, i.e. asynchronous sample rate conversion
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
    • H03H17/00—Networks using digital techniques
    • H03H17/02—Frequency selective networks
    • H03H17/04—Recursive filters
    • H03H17/0416—Recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing
    • H03H17/0427—Recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing characterized by the ratio between the input-sampling and output-delivery frequencies
    • H03H17/0455—Recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing characterized by the ratio between the input-sampling and output-delivery frequencies the ratio being rational
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04J—MULTIPLEX COMMUNICATION
    • H04J3/00—Time-division multiplex systems
    • H04J3/02—Details
    • H04J3/06—Synchronising arrangements
    • H04J3/062—Synchronisation of signals having the same nominal but fluctuating bit rates, e.g. using buffers
    • H04J3/0632—Synchronisation of packets and cells, e.g. transmission of voice via a packet network, circuit emulation service [CES]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L7/00—Arrangements for synchronising receiver with transmitter
    • H04L7/0016—Arrangements for synchronising receiver with transmitter correction of synchronization errors
    • H04L7/002—Arrangements for synchronising receiver with transmitter correction of synchronization errors correction by interpolation
    • H04L7/0029—Arrangements for synchronising receiver with transmitter correction of synchronization errors correction by interpolation interpolation of received data signal
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L7/00—Arrangements for synchronising receiver with transmitter
    • H04L7/0016—Arrangements for synchronising receiver with transmitter correction of synchronization errors
    • H04L7/005—Correction by an elastic buffer
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/439—Processing of audio elementary streams
    • H04N21/4392—Processing of audio elementary streams involving audio buffer management
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/44—Processing of video elementary streams, e.g. splicing a video clip retrieved from local storage with an incoming video stream or rendering scenes according to encoded video stream scene graphs
    • H04N21/44004—Processing of video elementary streams, e.g. splicing a video clip retrieved from local storage with an incoming video stream or rendering scenes according to encoded video stream scene graphs involving video buffer management, e.g. video decoder buffer or video display buffer
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N21/00—Selective content distribution, e.g. interactive television or video on demand [VOD]
    • H04N21/40—Client devices specifically adapted for the reception of or interaction with content, e.g. set-top-box [STB]; Operations thereof
    • H04N21/43—Processing of content or additional data, e.g. demultiplexing additional data from a digital video stream; Elementary client operations, e.g. monitoring of home network or synchronising decoder's clock; Client middleware
    • H04N21/44—Processing of video elementary streams, e.g. splicing a video clip retrieved from local storage with an incoming video stream or rendering scenes according to encoded video stream scene graphs
    • H04N21/4402—Processing of video elementary streams, e.g. splicing a video clip retrieved from local storage with an incoming video stream or rendering scenes according to encoded video stream scene graphs involving reformatting operations of video signals for household redistribution, storage or real-time display
    • H04N21/440263—Processing of video elementary streams, e.g. splicing a video clip retrieved from local storage with an incoming video stream or rendering scenes according to encoded video stream scene graphs involving reformatting operations of video signals for household redistribution, storage or real-time display by altering the spatial resolution, e.g. for displaying on a connected PDA
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04N—PICTORIAL COMMUNICATION, e.g. TELEVISION
    • H04N7/00—Television systems
    • H04N7/01—Conversion of standards, e.g. involving analogue television standards or digital television standards processed at pixel level
    • H04N7/0102—Conversion of standards, e.g. involving analogue television standards or digital television standards processed at pixel level involving the resampling of the incoming video signal

Definitions

  • the invention relates to a method and a device for converting data and a system for transmitting data.
  • the information is sent wired or wirelessly.
  • a transmitter or source z. B. microphone, camera, DVD player or the like and as a receiver or sink speaker, screen or the like.
  • the data is often sent directly using known standards (e.g. DVI, HDMI, SPDIF, AES/EBU) or using a point-to-point connection.
  • networks e.g. Ethernet, Internet
  • the receiver processes the received data using an SRC (sample rate converter, sampling rate converter, frame rate converter).
  • SRC sample rate converter, sampling rate converter, frame rate converter
  • the SRC converts a received video signal with a first resolution into a different screen resolution (e.g. source is 800x600 pixels, screen is 1920x1200 pixels) or in the case of an audio signal into a different sample rate (e.g. source with 48 kHz, loudspeakers at 44.1 kHz or 48 kHz not synchronized with the source).
  • An SRC is often implemented with low-pass filters which, based on the input signal, calculate which pixels or samples are best suited for the new resolutions, so that the viewer or listener can see or hear a result that is as true to the original as possible compared to the source.
  • synchronous point-to-point connections e.g. DVI, SPDIF
  • a synchronization signal or clock signal is also transmitted.
  • the sampling rates are in a known, fixed ratio relative to one another, which means that conversion can be carried out using relatively simple means.
  • Conventional SRCs work on the basis of this principle.
  • the method mentioned at the outset for converting data comprises at least the following steps.
  • the data packets include input data with a first sample rate.
  • the input data is assigned to positions in a combined input and filter buffer, hereinafter referred to simply as the "filter buffer", based on the first sample rate.
  • the input data are combined based on their respective position in the filter buffer as part of a low-pass filtering to output data with a defined second sample rate.
  • the input data in the filter buffer advances position-by-position and data-driven.
  • conversion means a conversion of data.
  • the conversion of the data relates on the one hand to the conversion of the data arriving in packets into a serial and preferably synchronizable or synchronized data stream and/or on the other hand to a change in the sampling rate or sample rate on which the data is based.
  • the data in particular digital data, can in principle include any data that is based on a sample rate. They therefore generally have a number of measuring points with the associated measured values, the so-called samples.
  • the data can be in the form of digital audio and/or video data, as will be explained in more detail later.
  • a sample rate refers to a temporal or spatial resolution on which the data is based. i.e. the sample rate is inversely proportional to a temporal or spatial distance between individual measurement points.
  • the data is thus recorded or measured with a corresponding sample rate and can also be stored, buffered, output and/or played back with this sample rate or with a different sample rate after conversion.
  • the incoming data is received as a number of data packets, for example by means of an input interface.
  • a data packet includes in particular at least one sample.
  • a data packet preferably includes a number of samples.
  • a data packet is therefore a so-called burst or data burst.
  • a plurality or a series, ie several, data packets are received.
  • the input data is therefore asynchronous data arriving in packets.
  • the incoming data or data packets are not synchronous, ie they are not coordinated in terms of time and usually arrive at irregular times. distance one. However, this packetized data is still based on a sample rate, namely the first sample rate.
  • the data packets can also contain other data such as e.g. B. include header data.
  • these are data packets that are transmitted in a network with a corresponding transmission protocol or network protocol.
  • the network protocol can, for example, be a protocol from the group of TCP/IP protocols, in particular the Ethernet protocol or another standard Internet protocol (IP) such as Sonet, ATM, 5G or the like.
  • IP Internet protocol
  • the data packets can take different routes in the network and therefore also have different transmission times.
  • a filter tap is assigned to a position in the filter buffer. It influences the result of the combination of the data in the filter buffer with the output data. This is done e.g. B. by means of a specifically defined weighting of the position or the filter taps within the framework of the low-pass filtering known in principle.
  • the low-pass filtering can be implemented in different ways, as will be explained in more detail later.
  • Newly arriving data - e.g. B. from the network - are assigned or allocated to an initial position in the filter buffer, as described below.
  • Data-driven means that when new data arrives, the data already in the filter buffer advances position by position. That is, if no new data arrives, the data remain at their previous position in the filter buffer. The data therefore advance in a burst-driven manner when a new data packet arrives.
  • the filter buffer is not operated with a fixed or defined clock, which was previously based on the first sample rate that was necessarily determined for this purpose, because according to the invention, the positions of the data in the filter buffer only change when new data arrives, using the invention it is therefore basically no longer necessary to determine the first sample rate.
  • the data advances in sync.
  • the advancement of the data takes place e.g. B. by shifting the actual position of the samples in the buffer, a new assignment of the samples to virtual positions or taps using memory addresses, pointers (pointers) or in a similar way.
  • Changing the position of a sample also changes the effect, for example the weighting of the low-pass filtering, of this sample when combined with the output data.
  • a sample has passed the intended positions in the filter buffer, it is deleted. This creates space for the following samples or data.
  • the filter buffer is a basically readable and writable, in particular random, memory.
  • the memory does not have to be read out sequentially or in blocks.
  • the addressing is preferably not carried out via the individual cells, but via words, ie z. B. in the form of blocks with the size of a sample. As described, it is filled with the input data and is read out at least as part of the low-pass filtering. In particular, it is implemented as an integrated circuit. But he can z. B. can also be composed of several memory modules.
  • the entire filter buffer is used for filtering. This is based on the idea that more data used in the filtering also leads to a better result. This also means in particular that preferably no areas in the filter buffer remain unused.
  • a value for a sample of the output data is available. i.e. using low-pass filtering, a value is interpolated for each sample of the output data.
  • the result of the combination or the low-pass filtering is read out or output with or in time with the second sample rate.
  • the second sample rate can fundamentally be different from or the same as the first sample rate.
  • the second sample rate preferably differs from the first sample rate.
  • the second sample rate is defined or specified. i.e. it can be specified, for example, when designing the process or the conversion device.
  • the second sample rate is preferably adjustable. i.e. she can e.g. B. by means of a user input, in accordance with other devices connected to the conversion device or by means of a default setting.
  • a preferably large configuration of the filter buffer means that missing data packets and/or an incorrect sequence of the incoming data packets can be animals are advantageously compensated in the context of the conversion of the data.
  • the method according to the invention takes place without the reconstruction of a synchronization signal.
  • the conversion device mentioned at the outset for converting data comprises an input interface.
  • the input interface is designed to receive a number of asynchronously arriving data packets.
  • the data packets include input data with a first sample rate.
  • the conversion device also has a filter buffer.
  • the filter buffer is designed in such a way that the input data is assigned to positions therein on the basis of the first sample rate.
  • the input data in the filter buffer advances position-by-position and data-driven.
  • a low-pass filter is included in the conversion device. The low-pass filter combines the input data based on their respective position in the filter buffer into output data with the defined second sample rate.
  • the conversion device according to the invention is therefore particularly suitable for carrying out a method according to the invention for converting data.
  • the input interface is e.g. B. designed as a common interface for the network standard used.
  • the conversion device preferably also includes an output interface for outputting the output data.
  • the output interface is e.g. B. designed as a standard interface for a desired audio standard or video standard.
  • the term input interface or output interface also includes non-standardised, ie e.g. B. proprietary, interface configurations.
  • the method according to the invention and the conversion device according to the invention can basically work without an input buffer or with a very small-sized pre-buffer.
  • the filter buffer alone is already sufficient to convert the data according to the invention, so that no further buffer is required for the basic invention.
  • the system mentioned at the outset for the transmission of data comprises an asynchronous data network and a conversion device according to the invention, the conversion device receiving and converting data packets from the data network.
  • the term "system” here generally describes the interaction of the explicitly mentioned components.
  • the system can also include several other components such.
  • the conversion device therefore acts in particular as an interface between an asynchronous network and a synchronous network and/or an asynchronous or synchronous terminal device.
  • a data-driven filter buffer is used in a conversion device for converting sample rates.
  • a large part of the aforementioned components of the conversion device, in particular the low-pass filter, can be implemented entirely or partially in the form of software modules in a processor of a corresponding control device.
  • the object is also achieved by a corresponding computer program product with a computer program which can be loaded directly into a memory device of a control device of a conversion device, with program sections in order to carry out all or at least some of the steps of the method according to the invention when the program is executed in the control device .
  • Such a computer program product can, in addition to the computer program, optionally contain additional components such as e.g.
  • documentation and/or additional components also include hardware components, such as hardware keys (dongles, etc.) for using the software.
  • a computer-readable medium for example a memory stick, a hard disk or another transportable or permanently installed data medium, on which the program sections of the computer program that can be read and executed by a computer unit of the control device are stored, can be used for transport to the control device and/or for storage on or in the control device are.
  • the computer unit can, for example, have one or more microprocessors or the like working together.
  • the input data preferably includes samples according to the first sample rate.
  • the samples of the input data are then preferably assigned to a respective position in the filter buffer on the basis of the first sample rate.
  • the input data are preferably combined as a function of the respective position of their samples in the filter buffer.
  • a sample is a signal digitized from an analog signal, i.e. a measured value recorded and digitized at a measurement time.
  • the sample has a bit depth that depends on how the measurements were taken.
  • the input data and the output data preferably include audio data and/or video data.
  • audio data there are typically audio samples which have an audio level with a bit depth of 8 bits, 16 bits, 20 bits, 32 bits or preferably 24 bits.
  • the audio data is therefore in particular audio data from the professional audio sector.
  • a video sample exists for each video frame and for each video level or color channel and the pixel associated with that video level.
  • a video frame of an HD signal with 1080 lines and a color resolution of 4:2:2 consists of 1920 samples for the luma signal and 960 samples each for the two color difference signals per line, i.e. a total of 3840 samples per line.
  • the conversion according to the invention can be applied to any spatial resolutions or color resolutions as well as temporal resolutions.
  • the color depth is z.
  • 24-bit true color, 8 bits each for R, G and B
  • 32-bit true color + 8-bit alpha channel, 8-bit each for R, G, B and alpha
  • 30-bit deep color / HDR video, 10 bits each for Y, U and V
  • 36 bits Deep Color / HDR10+ / Dolby Vision, 12 bits each for R, G and B.
  • the first sample rate of a video signal thus has several components.
  • a first component is the temporal resolution in the individual video frames, i.e. the frame rate.
  • One or more second components of the first sample rate of the video signal are the spatial resolutions of the individual color channels. This applies analogously to the second sample rate.
  • the second sample rate can be the same as the first sample rate, so that the conversion of packet-wise se incoming data takes place in a serial data stream. However, it can also differ from the first sample rate in one or more components, so that these components are converted.
  • the conversion method according to the invention can correspondingly be applied to one or more components of the signal in the case of video data. So it can be used to convert the frame rate and/or spatial resolution of each color channel. i.e. in particular that the principle of low-pass filtering can be applied both to a time sequence of samples that are assigned to a pixel and to video levels, e.g. B. spatially adjacent pixels of a video frame, for example, to interpolate intermediate pixels for a higher spatial resolution. With a suitable memory in the filter buffer and sufficient computing power, a simultaneous conversion of the frame rate and the spatial resolution is also possible.
  • the audio data preferably have sample rates of 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz or 192 kHz.
  • the video data preferably have temporal sample rates or frame rates of 25 Hz, 50 Hz, 59.94 Hz, 100 Hz or 119.88 Hz.
  • the positions of the filter buffer are preferably assigned filter coefficients for low-pass filtering, which are particularly preferably adjustable.
  • the output data is therefore generated from a linear combination of the data stored at the positions as part of the low-pass filtering, with the filter coefficients serving as weighting factors.
  • the fact that the filter coefficients are "adjustable" means that they can be changed as required. If e.g. If, for example, the first sample rate or the second sample rate with which the output data is output changes, the filter coefficients can be adjusted accordingly.
  • the low-pass filtering preferably takes place using an 11R filter or particularly preferably using an FIR filter.
  • the operating principles of the corresponding filters are known in principle to those skilled in the art.
  • the data stored in the filter buffer can particularly preferably be modified by means of feedback loops.
  • a feedback loop is preferably assigned to each position in the filter buffer or each filter tap. If, for example, it is determined when reading out the data that the data is not in one or more of the positions change, for example because no new input data arrives, the respective feedback loop regulates that the data stored at the position asymptotically approach the value zero. In other words, in this case, the feedback loops ensure that the filter decays.
  • the output data are preferably generated as a serial data stream by repeating at least some of the steps of the method according to the invention. For example, as part of the steps, the combination can be repeated, each time generating a new sample of output data.
  • the combination can also be repeated if no new input data is received. If the filter buffer is set up in such a way that the data remain unchanged in their position when there is no new input data, a time-constant value can be output as output data when the combination is repeated, for example.
  • the previously described decay can also be implemented by means of feedback loops.
  • the filter buffer is therefore preferably dimensioned to buffer at least 5 ms, particularly preferably at least 10 ms, very particularly preferably at least 15 ms, most preferably at least 20 ms of the input data.
  • the first sample rate is particularly preferably estimated by assigning the rate of incoming samples to one of the standard sample rates, in particular one of the standard sample rates specified above.
  • this estimation does not correspond to a reconstruction of a sync signal or clock signal, as would be required for a prior art SRC.
  • the input data are preferably pre-buffered and particularly preferably pre-sorted in a pre-buffer before the filter buffer.
  • the prebuffer is a relative to the Filter buffer small memory.
  • the pre-buffer is preferably many times smaller than the filter buffer. i.e. it has, for example, a memory size of at most one third, preferably at most one quarter, particularly preferably at most one fifth, very particularly preferably at most one tenth, of the memory size of the filter buffer.
  • the prebuffer can e.g. B. be designed as a FIFO memory.
  • the pre-buffer has logic that sends the data packets z. B. based on header data, as is common in many network protocols, sorted.
  • the pre-buffer should therefore only temporarily store a relatively small amount of data, for example in order to intercept bursts of data, ie one or more data packets arriving at short intervals. Transmission pauses occur between such bursts. They arise e.g. B. by combining multiple samples within the source to send data packets more efficiently.
  • the filter buffer is thus filled more evenly or in a more orderly manner, which at the same time improves the result of the following low-pass filtering.
  • the pre-buffer is particularly preferably connected to logic, a circuit or an electronic component which estimates the first sample rate as already described above.
  • the pre-buffer preferably forwards the individual samples at a higher rate than the estimated sample rate.
  • the rate is preferably at least 1% higher, more preferably at least 2% higher, most preferably at least 4% higher than the estimated sample rate. This advantageously prevents the pre-buffer from filling up.
  • the filter buffer is preferably designed in such a way that it can be read out and written to at the same time.
  • an arbitration logic regulates collisions, the z. B. arise with simultaneous accesses. So he can z. B. be designed as a dual-port RAM.
  • This refinement of the filter buffer is particularly advantageous since the two otherwise separate systems can work with the common data when filling and reading out the filter buffer without restricting each other's access speed.
  • the filter buffer and the low-pass filter form an interpolator.
  • the signal present at the output of the low-pass filter has a high frequency, e.g. B. is in the range of 50 MHz, updated
  • the conversion device preferably also includes a decimator, which is connected downstream of the interpolator and outputs the intermediate results generated by the interpolator with the desired second sample rate, i.e. a lower frequency or the new video resolution or desired audio sample rate .
  • the input data is preferably converted into output data in the form of a serial data stream.
  • the serial data stream of the output data is particularly preferably synchronous or can be synchronized with a number of devices connected to the conversion device.
  • the output data is then preferably transmitted via an output interface to a number of other devices with which the conversion device is synchronized. So it can be z. B. be just one or more synchronously connected to the conversion device facilities.
  • the conversion device is particularly preferably connected synchronously to a synchronously operating audio interface and/or video interface, which is followed by a number of other devices.
  • the quality of the (low-pass) filter of the interpolator of the sample rate converter is thus improved by the significantly increased number of TAPs by means of the invention. Furthermore, due to the increased low-pass effect, an increased tolerance for unstable or large networks is achieved.
  • the three electronic components described at the outset are implemented in just one component.
  • FIG. 1 shows a block diagram of a sequence of an exemplary embodiment of a method according to the invention for converting data
  • FIG. 2 is a block diagram of an embodiment of a conversion device according to the invention for converting data and
  • FIG. 3 shows a roughly schematic diagram of an embodiment of a system according to the invention for the transmission of data.
  • FIG. 1 shows a block diagram of a sequence of a method according to the invention for converting data by way of example and in a roughly schematic manner. It is explained below together with the exemplary embodiment of a conversion device 20 according to the invention for converting data, which is shown in FIG. 2 as an example and schematically as a block diagram.
  • a first step i input data ED arriving asynchronously are received in the form of data packets DP by means of an input interface 21 .
  • data packets DP For the sake of simplicity, three data packets PO, P1, P2 are shown as an example. However, it is usually a very much larger number of data packets DP.
  • the data packets arrive asynchronously because they B. are received over a network and their transit time through the network consequently differs due to differently arranged packets or differently routed paths through the network.
  • the input data ED include a large number of samples S01, S02, . . . , S23, S24, etc., which are based on a first sample rate SR1.
  • the reference number for a sample includes the respective package number as the front digit after the "S" and the number of a sample in the respective package as the back digit.
  • the input data ED represent z. B. a digitized audio signal with the first sample rate SR1 of z. B. 96 kHz and a bit depth of 24 bits was sampled.
  • Each sample S01, S02, . . . , S23, S24 thus has a sample value that characterizes the audio signal.
  • the input data ED are optionally pre-buffered in a pre-buffer 28 in order to compensate for irregular arrival of the data packets DP, and also optionally, e.g. B. pre-sorted based on their header data to sorted input data ED *.
  • the buffer 28 optionally includes sorting logic (not shown). So if e.g. B. the data packet P1 arrived at the input interface 21 before the data packet PO, they could be rearranged in the prebuffer 28 so that they are again arranged in the correct order shown.
  • the prebuffer can B. be designed as a FIFO buffer.
  • the pre-buffered or sorted input data ED* are transmitted to a filter buffer 22, preferably sample-by-sample at a higher rate than the estimated sample rate. i.e. each sample S01 , S02, ... , S23, S24 etc. is stored at a position in the filter buffer 22 or in a filter tap. For the sake of simplicity, the positions are shown here in a consecutive row. Even if such an implementation is possible in principle, it is clear to the person skilled in the art that the position in a memory, e.g. B. is described by memory addresses or pointers (pointers). Accordingly, the samples S01, S02, ...
  • S23, S24 do not usually have to be stored in storage locations that are actually lined up one after the other, but the ordered series of samples S01 , S02, ... , S23, S24 can be assigned to the samples S01 , S02, ..., S23, S24 assigned a series of memory addresses or pointers are understood.
  • the samples S01, S02, . i.e. the position of the samples S01 , S02, ... , S23, S24 is shifted by the number of newly arriving samples. As described above, this can basically be done by physically moving the sample to a new storage location.
  • the series of memory addresses or pointers is preferably simply changed accordingly.
  • a weighting element g1, g2, . . . gn of a low-pass filter 23 is linked to a position or a filter tap in the filter buffer 22. i.e. as part of a low-pass filtering that takes place in step ii of the method, the values of the samples S01, S02, ..., S23, S24 are each used by means of the weighting element or by means of the filter coefficient g1, g2, gn weighted, which is associated with them based on their position in the filter buffer 22 respectively.
  • the correspondingly weighted values are all subsequently added by means of a summation element 27 .
  • the setting of the filter coefficients can be fixed for specific parameters, for example during production, or can also be electronically set later.
  • the result of this linear combination is output from the low-pass filter 23 as combined data KD.
  • a sync signal CLK is provided for synchronization by means of a sync source 26 .
  • the sync source 26 can—as shown here—be arranged within the conversion device 20 so that other devices connected to the conversion device are synchronized to this sync signal CLK.
  • the sync source 26 can also be designed as an interface to another device connected to the conversion device or as a network interface.
  • the sample rate-converted data SKD are output via an output interface 24 .
  • the output interface 24 z. B. other devices or a network, preferably synchronized connected.
  • the filter buffer 22 is designed to be much larger than the pre-buffer 28. As can already be seen from the area interruptions in FIG. 2, not all positions of the pre-buffer 28 and the filter buffer 22 are shown. Likewise, the relationship between the pre-buffer 28 and the filter buffer is not to scale, but the filter buffer 22 is preferred many times larger than the prebuffer 28. It is dimensioned in particular to buffer at least 5 ms, preferably at least 10 ms, particularly preferably at least 15 ms, very particularly preferably at least 20 ms of the input data ED. This size of the filter buffer 22 advantageously makes it easier to compensate for interchanged and/or missing data packets DP—in particular if there is no prebuffer 28 or if the prebuffer 28 alone is not sufficient for this.
  • the samples S01, S02, S03 and S04 of the first data packet PO had failed, the samples S11, S12, S13 and S14 of the second data packet P1 would directly follow the preceding data. This would eventually cause a break or crackle in the audio signal when the signal was readily output.
  • the absence of the first data packet PO no longer has any significant impact on normal human hearing. The same applies analogously in the event that z. B. the first data packet PO and the second data packet P1 arrive in the filter buffer 22 in reversed order.
  • the first sample rate SR1 is 96 kHz.
  • the parameters such as e.g. B. the filter coefficients g1, g2, ..., gn are modified in order to output the signal as faithfully as possible to the original with the second sample rate SR2.
  • the weighting factors of the weighting elements g1, g2, . . . , gn can preferably be set. So you can e.g. B. be determined depending on the (estimated) first sample rate SR1, the second sample rate SR2 and / or other relevant parameters.
  • FIG. 3 shows a diagram of an exemplary embodiment of a system 30 according to the invention for the transmission of data in a roughly schematic manner.
  • the system 30 includes a microphone 31 as an audio source, the audio signal AS of which is preamplified and digitized by means of an A/D converter, i. H. is sampled with the first sample rate SR1 and a defined bit depth.
  • the sampled signal is transmitted to a network device 32, where it is packed into data packets DP and transmitted using a defined network protocol, e.g. B. Ethernet or other standard Internet protocols (IP) such as Sonet, ATM, 5G, sent via a data network 33.
  • IP Internet protocols
  • the data packets DP can be routed differently in the data network 33 and are therefore received asynchronously by the conversion device 20 as input data ED.
  • the conversion device 20 converts the input data ED, in particular the conversion into synchronized data and/or sample rate-converted data SKD takes place.
  • a second sample rate SR2 to be output can be specified by means of an input interface 25 for a conversion of the sample rate.
  • the specification can e.g. B. by a user, by a connected audio device or via the data network 33 or via an audio network synchronously connected to the conversion device 20.
  • the synchronous and sample rate-converted data SKD are converted back into an analog audio signal by means of a suitable loudspeaker 34 and reproduced.
  • a suitable loudspeaker 34 In addition to the microphone 31 and the loudspeaker 34, the system can also have a large number of other audio sources (microphones, line-in, etc.), playback devices, processing devices (such as mixers), or network devices (such as routers, repeaters ) and/or similar.
  • the conversion device 20 thus serves as an interface between an asynchronous data network 33 and synchronized audio devices such. B. the speaker 34, and / or synchronous data networks, in particular synchronous audio networks and / or video networks.
  • indefinite article “a” or “an” does not rule out the possibility that the characteristics in question can also be present more than once.
  • the terms “device”, “unit” and “system” do not rule out the component in question consisting of a number of interacting subcomponents, which may also be spatially distributed.

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Multimedia (AREA)
  • Computer Hardware Design (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Physics & Mathematics (AREA)
  • Mathematical Physics (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)
  • Communication Control (AREA)

Abstract

L'invention concerne un procédé de conversion de données comprenant au moins les étapes énoncées ci-après. Un certain nombre de paquets de données (P0, P1,..., P4) entrant de manière asynchrone sont reçus, les paquets de données (DP) comprenant des données d'entrée (ED) ayant une première fréquence d'échantillonnage. Des positions sont attribuées aux données d'entrée (ED) dans un tampon filtre (22) sur la base de la première fréquence d'échantillonnage. Les données d'entrée (ED) sont combinées sur la base de leur position respective dans le tampon filtre (22) pour former des données de sortie (SKD) présentant une deuxième fréquence d'échantillonnage (SR2) définie dans le cadre d'un filtrage passe-bas (23). Selon l'invention, les données d'entrée sont déplacées dans le tampon filtre (22) en termes de position et en fonction des données. Cette invention concerne en outre un dispositif de conversion (20) et un système (30) pour la transmission de données ainsi que l'utilisation d'un tampon filtre (22) dans un dispositif de conversion (20) pour la conversion de fréquences d'échantillonnage (SR1, SR2).
EP22760905.4A 2021-08-03 2022-08-01 Réception et conversion des fréquences d'échantillonnage de données audio et vidéo transmises de façon asynchrone Pending EP4381600A1 (fr)

Applications Claiming Priority (2)

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DE102021120204.3A DE102021120204A1 (de) 2021-08-03 2021-08-03 Empfang von Daten
PCT/EP2022/071594 WO2023012126A1 (fr) 2021-08-03 2022-08-01 Réception et conversion des fréquences d'échantillonnage de données audio et vidéo transmises de façon asynchrone

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EP4381600A1 true EP4381600A1 (fr) 2024-06-12

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EP (1) EP4381600A1 (fr)
CA (1) CA3226734A1 (fr)
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WO (1) WO2023012126A1 (fr)

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US6061410A (en) 1997-02-27 2000-05-09 Advanced Micro Devices Frequency ratio estimation arrangement and method thereof
US7023868B2 (en) * 1999-04-13 2006-04-04 Broadcom Corporation Voice gateway with downstream voice synchronization
US7738613B1 (en) * 2003-03-21 2010-06-15 D2Audio Corporation Streaming multi-channel audio as packetized data or parallel data with a separate input frame sync
JP2009162918A (ja) * 2007-12-28 2009-07-23 Toshiba Microelectronics Corp 復号再生装置及び方法並びに受信装置
DE102009008092B4 (de) 2009-02-09 2014-10-30 Atlas Elektronik Gmbh Verfahren und Vorrichtung zum Kompensieren von Abtastratenschwankungen
EP2768246B1 (fr) * 2013-02-13 2019-06-12 Sennheiser Communications A/S Procédé de fonctionnement d'un dispositif auditif et dispositif auditif

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DE102021120204A1 (de) 2023-02-09
WO2023012126A1 (fr) 2023-02-09

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