TWI394398B

TWI394398B - Apparatus and method for transmitting a sequence of data packets and decoder and apparatus for decoding a sequence of data packets

Info

Publication number: TWI394398B
Application number: TW97109732A
Authority: TW
Inventors: Sperschneider Ralph; Lutzky Manfred; Gayer Marc; Lohwasser Markus; Schnell Markus; Schuldt Michael
Original assignee: Fraunhofer Ges Forschung
Priority date: 2007-03-20
Filing date: 2008-03-19
Publication date: 2013-04-21
Also published as: TW200845644A

Description

用於傳輸資料分組序列的設備和方法以及用於對資料分組序列進行解碼的解碼器和設備Apparatus and method for transmitting a sequence of data packets and for solving a sequence of data packets Code decoder and device

本發明涉及資料通信應用，具體涉及經由面向分組的網路的音頻信號的即時通信。The present invention relates to data communication applications, and more particularly to instant communication of audio signals via a packet oriented network.

在經由面向分組的網路的即時通信中，例如IP語音(VoIP)中，通常不能保證所有的分組均在所需要的時間內到達接收機。其原因在於，當以面向分組的方式(例如經由網際網路)傳輸資料時，不同的分組採取經由資料網路的不同路徑，並且該不同的分組可以採用的、從該分組的發射機至該分組的接收機的、經由資料網路的路徑取決於當前的網路狀況。In instant messaging via packet-oriented networks, such as voice over IP (VoIP), it is generally not guaranteed that all packets arrive at the receiver within the required time. The reason for this is that when transmitting data in a packet-oriented manner (for example via the Internet), different packets take different paths via the data network, and the different packets can be used from the transmitter of the packet to the The path of the packetized receiver via the data network depends on the current network conditions.

即使以產生分組的順序來發送分組，到達接收機的順序也很有可能不同。發現了有利路徑的分組甚至可能“追上”較早時間發送但卻採取從發射機至接收機的較長路徑的分組。Even if the packets are transmitted in the order in which the packets are generated, the order of arrival at the receiver is likely to be different. A packet that finds a favorable path may even "catch up" a packet that was sent earlier but takes a longer path from the transmitter to the receiver.

由於接收機緩衝器在分組序列中的所有分組到達之前將簡單地進行緩衝，因此對於不需要即時操作的應用或者允許相對大的延遲的即時應用而言，上述不成問題。可以示例性地通過分組處的分組號或者分組序列指示來確定分組在序列中的位置，然後接收機將在複製分組或傳遞分組之前以正確的順序對分組進行分類。Since the receiver buffer will simply buffer before all packets in the sequence of packets arrive, this is not a problem for applications that do not require immediate operation or instant applications that allow for relatively large delays. The location of the packet in the sequence can be exemplarily determined by the packet number or packet sequence indication at the packet, and then the receiver will classify the packet in the correct order before copying the packet or delivering the packet.

然而，選擇的緩衝器越小，或者分組在從發射機向接收機傳輸時可以具有的可允許延遲越小，則分組失敗率越大。不僅在分組確實已丟失時會導致分組丟失，而且在分組從發射機到達接收機需要過長的時間時也會導致分組丟失。另一存在問題的情況是當分組在從發射機至接收機的過程中經歷了資料損壞時，即分組實際上產生錯誤時。However, the smaller the selected buffer, or the smaller the allowable delay that the packet can have when transmitting from the transmitter to the receiver, the more the packet failure rate Big. Packet loss can result not only when the packet is indeed lost, but also when the packet takes too long from the transmitter to reach the receiver. Another problematic situation is when a packet experiences data corruption in the process from the transmitter to the receiver, ie when the packet actually generates an error.

這種對延遲要求苛刻的應用在網際網路電話(IP語音)中出現，在這種應用中，為了使呼叫不被中斷，從發射機至接收機的分組必須滿足的延遲需求在順序方面相對嚴格。具體地，當在發射機側佈置了音頻編碼器並在接收機處佈置了解碼器，並且在接收機處不再有要編碼的資料時，也即當接收機側的解碼器由於缺少資料呈現而“崩潰”時，結果將是呼叫中斷。This demanding application for delays arises in Internet telephony (IP voice), in which the delay requirements that must be met from the transmitter to the receiver in order to keep the call uninterrupted in terms of order strict. Specifically, when an audio encoder is arranged on the transmitter side and a decoder is arranged at the receiver, and there is no more data to be encoded at the receiver, that is, when the decoder on the receiver side is presented due to lack of data In the case of "crash", the result will be a call interruption.

本發明的目的是提供一種儘管要求苛刻也能提供良好的通信品質的用於傳送資料分組的思想。It is an object of the present invention to provide an idea for transmitting data packets that provides good communication quality despite being demanding.

本發明的目的是通過一種依據申請專利範圍第1項的用於傳輸資料分組序列的設備、一種依據申請專利範圍第19項的用於傳輸資料分組序列的方法、一種依據申請專利範圍第20項的用於對資料分組序列進行解碼的解碼器、一種依據申請專利範圍第24項的用於對資料分組序列進行解碼的方法、或者一種依據申請專利範圍第25項的電腦程式來實現的。The object of the present invention is to provide an apparatus for transmitting a data packet sequence according to item 1 of the patent application scope, a method for transmitting a data packet sequence according to claim 19 of the patent application scope, and a claim 20 according to the patent application scope. A decoder for decoding a sequence of data packets, a method for decoding a sequence of data packets according to claim 24 of the patent application, or a computer program according to claim 25 of the patent application.

為了避免解碼器側的崩潰和/或避免解碼器側的用戶可聽見的偽像，根據本發明，執行對分組序列中的分組是否丟失或錯誤的檢測。如果確定了這種分組丟失或分組差錯，則將提供替代分組，該替代分組是關於分組語法的有效分組，然而，其中該替代分組的音頻內容具有特定的內容特性。根據本發明的用於傳輸資料分組序列的設備輸出未擾亂的分組序列，然而，其中錯誤分組或未接收到的分組已被替代分組所替代，因此該傳輸設備所的輸出序列包括至少一個接收到的分組以及一個替代分組。在一個實現中，替代分組中的分組或幀的內容和/或內容特性獨立於音頻信號，即不取決於在先或後續的分組或幀。然而，如果對該分組採取差錯隱藏措施，則合成的音頻內容將取決於在先或後續的幀，即將不再為預定的或獨立於信號。In order to avoid a crash on the decoder side and/or to avoid audible artifacts on the decoder side, according to the invention, whether to perform a packet in the sequence of packets is performed Lost or wrong detection. If such a packet loss or packet error is determined, then an alternate packet will be provided, which is a valid packet with respect to the packet syntax, however, where the audio content of the alternate packet has a particular content characteristic. The apparatus for transmitting a sequence of data packets according to the present invention outputs an undisturbed sequence of packets, however, wherein an erroneous packet or an unreceived packet has been replaced by a substitute packet, so that the output sequence of the transmission device includes at least one received Grouping and an alternative grouping. In one implementation, the content and/or content characteristics of the packets or frames in the alternate packet are independent of the audio signal, ie, not dependent on prior or subsequent packets or frames. However, if error concealment measures are taken for the packet, the synthesized audio content will depend on the previous or subsequent frame, and will no longer be predetermined or independent of the signal.

此外，該替代分組在有效載荷區域中提供了對該分組是替代分組的事實的指示，其中該指示是可忽略的或者可由基本解碼器來解釋的，以使得根據預定的內容特性如同有效分組一樣地對該替代分組進行解碼，並且其中該指示可由擴展解碼器解釋，該擴展解碼器與該基本解碼器相比具有用於執行產生替代分組的內容的差錯隱藏措施的擴展功能，該替代分組具有與該預定的內容特性不同的內容特性。該替代分組可以是純有效載荷分組或者可以是包括有效載荷部分和報頭部分的分組，該指示不存在於報頭部分，而是優選地存在於有效載荷部分。Furthermore, the substitute packet provides an indication in the payload area of the fact that the packet is a substitute packet, wherein the indication is negligible or can be interpreted by the base decoder such that the predetermined content characteristics are as valid as the valid packet Decoding the alternate packet, and wherein the indication is interpretable by an extended decoder having an extended function for performing error concealment measures for generating content of the substitute packet, the substitute packet having A content characteristic different from the predetermined content characteristics. The alternate packet may be a pure payload packet or may be a packet comprising a payload portion and a header portion, the indication not present in the header portion, but preferably present in the payload portion.

資料分組序列的接收機接收未擾亂的資料分組流，該資料分組均具有有效的資料語法。該接收機將能夠容易地對資料分組序列進行解碼。在該接收機作為基本接收機時，該接收機將因此容易地對替代分組進行解碼並再現預定的音頻內容。然而，由於該音頻內容是預定的，因此這將導致品質損失，並因此將不能很好地適合在先的分組或幀以及後續的分組或幀。然而，與由於不存在分組以及因此中斷了整個通信連接而導致的解碼器完全崩潰的情況相比，這種短時品質損失並不成問題。The receiver of the data packet sequence receives the undisturbed data packet stream, which has a valid data syntax. The receiver will be able to easily decode the sequence of data packets. In the receiver as a basic receiver At this time, the receiver will thus easily decode the substitute packet and reproduce the predetermined audio content. However, since the audio content is predetermined, this will result in a loss of quality and therefore will not fit well into prior packets or frames and subsequent packets or frames. However, such short-term quality loss is not a problem compared to the case where the decoder is completely collapsed due to the absence of packets and thus the entire communication connection is interrupted.

相反地，擴展解碼器能夠使用替代分組中的、指出了該分組是替代分組而非普通分組的指示來識別上述事實，並且不是簡單地處理替代分組，而是在接收到替代分組時發起差錯隱藏措施。Conversely, the extended decoder can use the indication in the alternate packet that indicates that the packet is a substitute packet instead of a normal packet, and does not simply process the substitute packet, but initiates error concealment when the substitute packet is received. Measures.

除幀/分組重複之外，差錯隱藏措施示例性地包括已存於記憶體中的先前分組或後續分組在先前分組和後續分組之間的外插。這種外插或內插具體包括在差錯隱藏情況中對音頻信號的短時頻譜進行分頻段能量測量和合成，該音頻信號的頻譜值是以大概隨機的方式產生的，然而，其中該隨機產生的音頻信號的分頻段能量取決於已經通過差錯隱藏措施而產生的、先前和/或後續的正確接收的一個或更多分組的能量。In addition to frame/packet repetition, error concealment measures illustratively include extrapolation of previous or subsequent packets already in memory between previous and subsequent packets. The extrapolation or interpolation specifically includes sub-band energy measurement and synthesis of the short-time spectrum of the audio signal in the case of error concealment, the spectral value of the audio signal is generated in a roughly random manner, however, the random generation The sub-band energy of the audio signal depends on the energy of one or more packets that have been previously received and/or subsequently received by the error concealment measure.

在一個實施例中，替代分組所具有的預定內容特性是零頻譜。這裏的結果是對預定的替代分組進行解碼的基本解碼器執行“雜訊抑制”。備選地，該內容特性可以是其音頻內容和/或頻譜值與絕對聽音臨界值相關的音頻信號，並且是以該音頻內容小於該絕對聽音臨界值的兩倍的方式來示例性地定義的，因此，在所有頻段中包括特定但少量的雜訊，該雜訊在特定情況下可能比簡單的“雜訊抑制”在主觀上聽起來更好。In one embodiment, the substitute packet has a predetermined content characteristic that is a zero spectrum. The result here is that the basic decoder that decodes the predetermined alternate packet performs "noise suppression." Alternatively, the content characteristic may be an audio signal whose audio content and/or spectral value is related to an absolute listening threshold, and is exemplarily provided in such a manner that the audio content is less than twice the absolute listening threshold Defined, therefore, includes specific but small amounts in all frequency bands The noise, which may sound subjectively better than simple "noise suppression" in certain situations.

下面將參照附圖詳細地描述本發明的實施例。Embodiments of the present invention will be described in detail below with reference to the accompanying drawings.

第一圖示出了一種用於傳輸表示音頻信號的資料分組序列的設備。在第一圖中示例性地實現為基站10的傳輸設備包括用於接收該序列中的分組的裝置11，分組語法是針對分組指定的。接收裝置11示例性地連接至面向分組的傳輸網路，例如網際網路12。此外，基站10包括用於檢測分組序列中的分組是否丟失或發生錯誤的裝置13。此外，還提供了用於提供替代分組的裝置14，以替代錯誤分組或丟失的分組。替代分組是關於分組語法的有效分組，然而替代分組的音頻內容具有預定的內容特性。此外，該基站包括用於輸出分組序列的裝置15，該分組序列包括至少一個接收的分組以及一個替代分組。利用基站的示例，輸出裝置15是耦合至天線16的HF前端，用於根據預定規範(例如NG DECT規範)來將資料分組序列傳輸至下面將參照第二圖進行更詳細討論的移動單元。The first figure shows an apparatus for transmitting a sequence of data packets representing an audio signal. The transmission device exemplarily implemented in the first figure as base station 10 comprises means 11 for receiving packets in the sequence, the packet syntax being specified for the packets. The receiving device 11 is exemplarily connected to a packet-oriented transmission network, such as the Internet 12. Furthermore, the base station 10 comprises means 13 for detecting whether a packet in a sequence of packets is missing or an error has occurred. In addition, means 14 for providing alternate packets are provided to replace erroneous packets or lost packets. The alternate packet is a valid packet with respect to the packet syntax, however the audio content of the alternate packet has a predetermined content characteristic. Furthermore, the base station comprises means 15 for outputting a sequence of packets comprising at least one received packet and a substitute packet. Using an example of a base station, output device 15 is an HF front end coupled to antenna 16 for transmitting a sequence of data packets to a mobile unit as will be discussed in greater detail below with respect to the second figure in accordance with a predetermined specification (e.g., NG DECT specification).

第一圖中的裝置11經由分組線路17耦合至輸出裝置15，以將接收到的普通(也即無錯的)分組準時傳輸至裝置15。此外，該用於提供替代分組的裝置14經由替代分組線路18與輸出裝置15相連。優選地，用於提供替代分組的裝置14包括在其中儲存替代分組的記憶體。只要識別到了丟失的或錯誤的分組，裝置14都啟用記憶體訪問以從該記憶體中取回替代分組，並經由線路18將該替代分組饋入該輸出裝置15。The device 11 in the first figure is coupled to the output device 15 via a packet line 17 to transmit the received normal (i.e., error free) packets to the device 15 on time. Furthermore, the means 14 for providing an alternate packet is connected to the output device 15 via an alternate packet line 18. Preferably, the means 14 for providing a substitute packet includes a memory in which the substitute packet is stored. As long as it is recognized With the missing or erroneous packet, device 14 enables memory access to retrieve the alternate packet from the memory and feed the alternate packet to output device 15 via line 18.

在一個實施例中，控制檢測裝置13以檢測分組丟失，並在超過最大延遲的時間內未接收到分組序列中的分組時，啟用用於提供替代分組的裝置。在一個實施例中，該最大延遲示例性地可經由控制線路19來控制。可以經由控制線路19將在VoIP應用中示例性地包括最大延遲的QoS(服務品質)請求饋入該檢測裝置13。對於其他面向分組的應用中，除VoIP之外，還可能存在經由控制線路19饋入該檢測裝置13的不同QoS請求。備選地，該檢測裝置13還可以具有固定地設置的準則，根據該準則來檢測分組差錯或分組丟失並啟用用於提供替代分組的裝置14。In one embodiment, the detecting means 13 is controlled to detect packet loss and to enable means for providing a substitute packet when the packet in the sequence of packets is not received within a time exceeding the maximum delay. In one embodiment, this maximum delay is illustratively controllable via control line 19. A QoS (Quality of Service) request, which typically includes a maximum delay in the VoIP application, can be fed to the detection device 13 via the control line 19. For other packet-oriented applications, in addition to VoIP, there may be different QoS requests that are fed into the detection device 13 via the control line 19. Alternatively, the detection device 13 may also have a fixedly set criterion according to which a packet error or packet loss is detected and the means 14 for providing a substitute packet is enabled.

在一個實施例中，經由替代分組線路18提供給輸出裝置15的替代分組不僅具有預定的內容特性，而且具有對該分組是替代分組的事實的指示。在一個實施例中，該指示使得接收該替代分組的基本解碼器忽略該指示並根據預定的內容特性如同有效分組一樣對該分組進行解碼，並使得與基本解碼器相比具有擴展功能的擴展解碼器解釋該指示，以執行針對替代分組產生與預定的內容特性不同的內容的差錯隱藏措施。In one embodiment, the alternate packet provided to output device 15 via alternate packet line 18 has not only a predetermined content characteristic, but also an indication of the fact that the packet is a substitute packet. In one embodiment, the indication causes the base decoder receiving the substitute packet to ignore the indication and decode the packet as a valid packet according to a predetermined content characteristic and to enable extended decoding with extended functionality compared to the base decoder The instruction interprets the indication to perform error concealment measures that produce content that is different from the predetermined content characteristics for the alternate packet.

第二圖示出了可示例性地位於移動單元20中的用於對分組序列進行解碼的解碼器。該解碼器包括用於接收分組語法針對其指定的分組序列的接收機21，該序列包括至少The second figure shows a decoder that can be exemplarily located in mobile unit 20 for decoding a sequence of packets. The decoder includes a receiver 21 for receiving a sequence of packets for which a packet syntax is specified, the sequence including at least

一個資料分組以及至少一個替代分組，該替代分組是關於該分組語法的有效分組，並且該替代分組的音頻內容包括預定的內容特性。此外，該替代分組包括對該分組是替代分組的事實的指示。然而，只能通過第二圖所示的擴展編碼器來解釋該指示，而不能通過基本解碼器來解釋該指示。A data packet and at least one replacement packet, the replacement packet being a valid packet with respect to the packet syntax, and the audio content of the replacement packet including predetermined content characteristics. Moreover, the alternate packet includes an indication of the fact that the packet is a substitute packet. However, the indication can only be interpreted by the extended encoder shown in the second figure, and the indication cannot be interpreted by the base decoder.

由檢測器22完成對該替代指示的解釋，該檢測器22被實現為檢測分組是否包含該指示並因此是替代分組。第二圖中的擴展解碼器還包括差錯隱藏裝置23，該差錯隱藏裝置23用於對包括與包含在替代分組中的預定內容特性不同的內容特性的合成音頻內容進行合成。此外，該擴展解碼器包括音頻呈現裝置24，該音頻呈現裝置24用於在分組並非替代分組時呈現分組中的音頻內容，以及用於在分組是替代分組時呈現該合成音頻內容。該音頻呈現裝置24與輸出單元25相耦合，該輸出單元25示例性地包括D/A轉換器、放大器和揚聲器。The interpretation of the alternate indication is done by detector 22, which is implemented to detect if the packet contains the indication and is therefore a substitute packet. The spread decoder in the second figure further includes error concealing means 23 for synthesizing the synthesized audio content including the content characteristics different from the predetermined content characteristics contained in the substitute packet. Moreover, the extended decoder includes an audio rendering device 24 for presenting audio content in the packet when the packet is not a substitute packet, and for presenting the synthesized audio content when the packet is a substitute packet. The audio presentation device 24 is coupled to an output unit 25 that illustratively includes a D/A converter, an amplifier, and a speaker.

具體地，該音頻呈現裝置經由傳輸常規分組的分組線路26與接收機21相耦合。此外，該音頻呈現裝置24經由替代分組線路27與差錯隱藏裝置23相連，通過該替代分組線路27將合成音頻內容從差錯隱藏裝置23傳輸至音頻呈現裝置24。In particular, the audio rendering device is coupled to the receiver 21 via a packet line 26 that transmits regular packets. Furthermore, the audio presentation means 24 is connected to the error concealing means 23 via the alternative packet line 27, via which the synthesized audio content is transmitted from the error concealing means 23 to the audio presentation means 24.

差錯隱藏裝置23可以以不同的方式來執行差錯隱藏。簡單的差錯隱藏方法是簡單地重複先前的幀和/或先前分組的音頻內容或後續分組的音頻內容。這種差錯隱藏方法被稱為“幀重複”。備選地，可以實施該差錯隱藏方法以執行外插或內插。該外插或內插可以相對於頻譜值或頻段來執行。在頻譜值外插的情況下，可以基於一個或多個具有與先前幀相等的頻率的頻譜值來形成替代幀的頻譜值。備選地，還可以關於多個頻段來執行差錯隱藏，在該多個頻段中示例性地由緩衝器作為亂數發生器產生或以大概確定的方式產生一個頻段中的頻譜值，然後對該頻譜值進行加權以使得該頻譜值所表示的能量等於目標能量，該目標能量源自一個或多個先前的幀和/或一個或多個後續的幀。該先前和/或後續的幀可以是有效的接收分組，或者可以是在發生分組丟失並且不僅丟失單個分組而且丟失多個連續分組時由差錯隱藏裝置所產生的分組和/或幀。The error concealing means 23 can perform error concealment in different ways. A simple error concealment method is to simply repeat the audio content of the previous frame and/or the previously grouped audio content or subsequent packets. This error concealment method is called "frame repetition." Alternatively, the error concealment method can be implemented to perform Extrapolation or interpolation. This extrapolation or interpolation can be performed with respect to spectral values or frequency bands. In the case of spectral value extrapolation, the spectral values of the substitute frame may be formed based on one or more spectral values having frequencies equal to the previous frame. Alternatively, error concealment may also be performed with respect to a plurality of frequency bands in which the spectral values in one frequency band are generated by the buffer as a random number generator or in a roughly determined manner, and then The spectral values are weighted such that the energy represented by the spectral values is equal to the target energy derived from one or more previous frames and/or one or more subsequent frames. The previous and/or subsequent frames may be valid received packets, or may be packets and/or frames generated by the error concealment device when packet loss occurs and not only a single packet is lost but multiple consecutive packets are lost.

在本發明的一個實施例中，產生分組序列的音頻編碼器是基於變化的音頻編碼器。這種基於變換的音頻編碼器包括時頻轉換級30，通過時頻轉換級30將時域音頻信號轉換為短時頻譜序列。將每個短時頻譜饋入執行量化的量化器31，量化器31由心理聲學模型32控制，以便執行該量化以使得量化雜訊不干擾主觀的音頻印象。該量化器的下游是熵編碼器33，該熵編碼器33示例性地可以是Huffman編碼器。該熵編碼器提供了與由量化器31以比例因數的形式示例性地提供的、並由熵編碼器33以所使用的編碼表的形式示例性地提供的輔助資訊有關的位元序列，並且該位元序列形成了要提供給在輸出側輸出資料分組序列的分組打包器34的資料。除了該分組打包器之外，如同示例性地在關鍵字MP3(MPEG-1層3)或AAC (MPEG-4)或AC-3 等中已知的，第二圖中所示的音頻編碼器表示了典型的基於變換的編碼器。應當指出，根據需求將分組打包器34實現為針對每個音頻幀(即每個短時頻譜)產生一個分組，或者產生多於一個的音頻幀，即，將多個編碼的短時頻譜引入單個分組中。In one embodiment of the invention, the audio encoder that produces the sequence of packets is a variable audio encoder. This transform-based audio encoder includes a time-frequency conversion stage 30 that converts the time-domain audio signal into a short-term spectral sequence by the time-frequency conversion stage 30. Each short-term spectrum is fed into a quantizer 31 that performs quantization, which is controlled by a psychoacoustic model 32 to perform the quantization so that the quantization noise does not interfere with the subjective audio impression. Downstream of the quantizer is an entropy encoder 33, which may illustratively be a Huffman encoder. The entropy coder provides a sequence of bits related to auxiliary information exemplarily provided by the quantizer 31 in the form of a scaling factor and exemplarily provided by the entropy encoder 33 in the form of a coding table used, and This sequence of bits forms the data to be supplied to the packet packer 34 which outputs the sequence of data packets on the output side. In addition to the packet packer, as exemplarily in the keyword MP3 (MPEG-1 Layer 3) or AAC (MPEG-4) or AC-3 As is known in the art, the audio encoder shown in the second figure represents a typical transform-based encoder. It should be noted that the packet packetizer 34 is implemented as needed to generate one packet for each audio frame (i.e., each short time spectrum) or to generate more than one audio frame, i.e., to introduce multiple encoded short time spectra into a single In the group.

第四圖示出了第二圖中的音頻呈現裝置24的更詳細的表示，具體地是示出了音頻呈現裝置24與差錯隱藏裝置23的協作。在輸入側，音頻呈現裝置包括對分組進行拆包以便從“主資訊”中分離出輔助資訊的分組拆包器40。The fourth figure shows a more detailed representation of the audio rendering device 24 in the second figure, in particular showing the cooperation of the audio rendering device 24 with the error concealing device 23. On the input side, the audio rendering device includes a packet unpacker 40 that unpacks the packets to separate the auxiliary information from the "master information."

將主資訊(即由位元序列表示的短時頻譜)饋入熵編碼器41，熵編碼器41提供了量化索引，將該量化索引饋入反量化器42，反量化器42在輸出端處提供量化後又反量化的頻譜值，然後經框43中的頻時轉換之後使用該頻譜值產生輸出音頻信號。熵解碼器41和反量化器42均可通過輔助資訊來控制，該熵解碼器典型地接收碼表索引，而該反量化器42接收用於執行正確的反量化的比例因數。The main information (i.e., the short-term spectrum represented by the bit sequence) is fed to an entropy coder 41, which provides a quantization index, which is fed to an inverse quantizer 42, which is at the output. The quantized and inversely quantized spectral values are provided and then used to generate an output audio signal after the frequency binning in block 43. Both the entropy decoder 41 and the inverse quantizer 42 can be controlled by auxiliary information, which typically receives a code table index, and the inverse quantizer 42 receives a scaling factor for performing the correct inverse quantization.

然後在分組拆包器40包括如第二圖的檢測器22中存在的檢測特性時，該分組拆包器40能夠向隱藏裝置22發送替代分組指示，以使得該隱藏裝置23可以識別當前的分組並非第三圖中的編碼器產生的分組，而是基站產生的替代分組。在這種情況下，該隱藏裝置將向熵解碼器提供位元序列，或者向反量化器提供量化索引序列，或者向頻時轉換裝置提供頻譜值序列，以便在任何位置處將合成音頻內容饋入解碼器功能鏈。優選地，在該鏈的末端處(即在頻時轉換級43處)饋入合成的頻譜音頻內容。The packet unpacker 40 can then send a substitute packet indication to the hiding device 22 when the packet unpacker 40 includes a detection characteristic as present in the detector 22 of the second figure such that the hiding device 23 can identify the current packet. It is not the packet generated by the encoder in the third figure, but the substitute packet generated by the base station. In this case, the hiding device will provide a sequence of bits to the entropy decoder, or provide a sequence of quantized indices to the inverse quantizer, or provide a sequence of spectral values to the time-frequency conversion device to feed the synthesized audio content at any location. Into the decoder function chain. Preferably at the end of the chain (ie at The frequency-time conversion stage 43) feeds the synthesized spectral audio content.

這些頻譜內容優選地取決於已正確接收的先前的頻譜，或者取決於後續頻譜，該後續頻譜可能已存在，並且可能包括與頻譜值、頻段或二者(即與頻譜值和頻段)相關的差錯隱藏裝置，以用於合成該音頻內容。These spectral contents preferably depend on the previous spectrum that has been correctly received, or on the subsequent spectrum, which may already be present and may include errors associated with spectral values, frequency bands or both (ie with spectral values and frequency bands) A device is hidden for synthesizing the audio content.

如同使用第一圖和第二圖已示出的，在基站和移動單元中利用本發明的實現，其中假設基站可以以不正確的順序來接收分組，例如當基站與網際網路相耦合時，而移動單元取決於以正確的順序來接收分組序列。例如，通過DECT標準定義基站與移動單元之間的這種通信連接。As has been shown using the first and second figures, an implementation of the present invention is utilized in a base station and a mobile unit, wherein it is assumed that the base station can receive packets in an incorrect order, such as when the base station is coupled to the Internet, The mobile unit depends on receiving the sequence of packets in the correct order. For example, such a communication connection between a base station and a mobile unit is defined by the DECT standard.

在經由面向分組的網路的即時通信(例如VoIP)中，不能保證所有的分組都在所需要的時間內到達接收機。在特定的(非常有限的)時間之後，必須將尚未到達的分組歸類為丟失的分組。在IP分組丟失的情況下，產生替代音頻幀並通過基站進行傳輸。In instant messaging (e.g., VoIP) via a packet-oriented network, there is no guarantee that all packets will arrive at the receiver within the required time. After a certain (very limited) time, the packets that have not arrived must be classified as missing packets. In the event that an IP packet is lost, an alternate audio frame is generated and transmitted through the base station.

這種替代將由下一代(NG)DECT系統中的基站來執行。從基站向移動單元傳輸該替代的幀，以代替原始的(而非接收到的)幀。NG DECT規範特別地通過寬頻和超寬頻音頻編解碼器和IP終端來表示當前的DECT規範的擴展。This replacement will be performed by a base station in a next generation (NG) DECT system. The alternate frame is transmitted from the base station to the mobile unit in place of the original (rather than received) frame. The NG DECT specification specifically represents an extension of the current DECT specification through broadband and ultra-wideband audio codecs and IP terminals.

NG DECT站包括一個或多個從NG DECT基站接收呼叫的無線電話。因此，可以使得VoIP呼叫直接經過NG DECT電話。在理想情況下，可以將VoIP語音分組從基站傳輸至移動單元，而無需在基站中進行再次編碼。The NG DECT station includes one or more radiotelephones that receive calls from the NG DECT base station. Therefore, it is possible to make a VoIP call directly through the NG DECT phone. Ideally, VoIP voice packets can be transmitted from the base station to the mobile unit without re-encoding in the base station.

例如，當使用其常規語法未提供用信號通知幀丟失的特殊方式的音頻編解碼器時，產生替代分組。這種替代音頻幀應當可由符合標準的解碼器進行解碼，但在一個實施例中，應當同時向擴展的解碼器提供明確地識別該幀是替代幀的方式，以使得該擴展的解碼器可以啟用相應的對策，例如差錯隱藏。應當在沒有較大計算複雜度的情況下另外完成替代幀的引入，具體地是在沒有基站對先前音頻資料的特定估計的情況下完成的，以使得基站可以如同純中繼站(即提供資料傳輸而不再進行解碼和編碼的站)一樣地操作。因此，基站應當僅執行非常少量的分組拆包(如果有的話)，該分組拆包可示例性地僅被執行用於恢復分組序列資訊，該分組序列資訊指示資料分組被佈置在該序列中的何處，因此可以對由該資料分組序列所表示的音頻內容進行正確解碼。For example, when using its regular syntax, it is not provided to signal the loss of the frame. When a special way of audio codec is generated, an alternate packet is generated. Such alternate audio frames should be decodable by a standard compliant decoder, but in one embodiment, the extended decoder should be simultaneously provided with a way to explicitly identify that the frame is a substitute frame so that the extended decoder can be enabled Corresponding countermeasures, such as error concealment. The introduction of a substitute frame should be done without significant computational complexity, in particular without the base station's specific estimation of previous audio material, so that the base station can act as a pure relay (ie providing data transmission) The station that does not decode and encode is operated the same. Therefore, the base station should only perform a very small number of packet unpacking (if any), which can be exemplarily performed only for recovering packet sequence information, the packet sequence information indicating that the data packet is arranged in the sequence Wherever it is, the audio content represented by the sequence of data packets can be correctly decoded.

在一個實施例中，基於變換的編碼器使用特定的用戶專用資料區域來向包括擴展功能的解碼器(即擴展編碼器)提供信號通知，以指示相應的幀是替代幀，即使底層的位元流語法標準尚未提供這種信號通知。In one embodiment, the transform-based encoder uses a particular user-specific material area to signal a decoder (ie, an extended encoder) that includes an extended function to indicate that the corresponding frame is a substitute frame, even if the underlying bit stream This signaling has not been provided by the grammar standard.

當幀或分組攜帶報頭和有效載荷部分時，其中該有效載荷部分包含有用資料，由於在後續的音頻處理中無論如何都不會考慮報頭，因而該指示將丟失並因此無法再進行差錯隱藏，因此優選地將該指示容納在該有效載荷部分中。When a frame or packet carries a header and payload portion, where the payload portion contains useful data, since the header is not considered anyway in subsequent audio processing, the indication is lost and thus error concealment is no longer possible, thus The indication is preferably accommodated in the payload portion.

當分組僅具有有效載荷部分而不具有報頭時，僅將該指示容納在有效載荷部分，從而能夠實現本實施例。優選地，將該指示容納在音頻資料和/或音頻資料部分中。When the packet has only the payload portion without the header, only the indication is accommodated in the payload portion, so that the present embodiment can be implemented. Preferably, the indication is contained in an audio material and/or audio material portion.

在一個實現中，在未提供用信號通知幀丟失的顯式方式的情況下，替代音頻幀和/或替代分組滿足該替代幀和/或替代分組可由符合標準的解碼器進行解碼的第一準則。第二準則是替代幀應當向擴展解碼器提供明確識別該幀是替代幀的方式，以使得該擴展解碼器可以啟用差錯隱藏。In one implementation, where an explicit manner of signaling frame loss is not provided, the substitute audio frame and/or the substitute packet satisfy a first criterion that the substitute frame and/or the substitute packet can be decoded by a conforming decoder . The second criterion is that the substitute frame should provide the extension decoder with a way to explicitly identify that the frame is a substitute frame so that the extended decoder can enable error concealment.

如果替代幀是根據預定的標準化的資料流程語法和/或分組語法或幀語法的有效幀，則符合標準的傳統的解碼器將能夠對該替代幀進行解碼。在一個實現中，優選地通過不具有音頻內容的幀來替代丟失的幀，即，執行所謂的雜訊抑制。雜訊抑制意味著整個頻譜被設置為零。具體地，通過使用AAC標準(MPEG-4-音頻)，該變化優選地用於產生零頻譜，在零頻譜中將用於傳輸頻譜值的最高的比例因數頻段設置為零(max_sfb=0)。備選地，還可以傳輸針對再次為零的比例因數頻段的頻譜值。這原理上可以使用所提供的不同碼本中的任何可用的Huffman碼本來實現，其中當使用碼本"ZERO_HCB"(Zero Huffman碼表)時，顯然不必傳輸這些譜線。If the substitute frame is a valid frame according to a predetermined standardized data flow syntax and/or a packet syntax or a frame syntax, a standard-compliant conventional decoder will be able to decode the substitute frame. In one implementation, the lost frame is preferably replaced by a frame that does not have audio content, ie, so-called noise suppression is performed. Noise suppression means that the entire spectrum is set to zero. Specifically, by using the AAC standard (MPEG-4-Audio), the change is preferably used to generate a zero spectrum in which the highest scale factor band for transmitting spectral values is set to zero (max_sfb = 0). Alternatively, it is also possible to transmit spectral values for a scale factor band that is again zero. This can in principle be implemented using any of the available Huffman codebooks provided, wherein when using the codebook "ZERO_HCB" (Zero Huffman code table), it is obviously not necessary to transmit these spectral lines.

應當指出，檢測錯誤幀或丟失幀的基站可以已執行差錯隱藏措施。然而，根據本發明，由於差錯隱藏措施在計算方面較為複雜，並且額外地需要估計先前的(以及可能為將來的)音頻信號，因此基站優選地不執行這種差錯隱藏措施。具體地，當DECT基站為多個移動單元“提供服務”時，為了能夠執行複雜的差錯隱藏，其結果將是基站再次對所有音頻內容連續地進行解碼和編碼。除了對於較高的處理器和記憶體資源的相關需求之外，特別是在使用有損編碼器時，結果將是由於串列編碼效應而導致的附加的品質惡化。此外，延遲將大大增加。It should be noted that base stations that detect erroneous frames or lost frames may have performed error concealment measures. However, according to the present invention, since the error concealment measures are computationally complicated and additionally require estimation of the previous (and possibly future) audio signals, the base station preferably does not perform such error concealment measures. In particular, when a DECT base station "serves" a plurality of mobile units, in order to be able to perform complex error concealment, the result will be that the base station continuously decodes and encodes all of the audio content again. Except for In addition to the high processor and memory resource requirements, especially when using lossy encoders, the result will be additional quality degradation due to the tandem encoding effect. In addition, the delay will increase greatly.

由於在AAC標準中未提供用信號通知幀丟失的顯式方式，在一個實現中，使用常規的解碼器所忽略的信號通知方式。這裏保持資料語法和/或分組語法。另一方面，由於這種類型的幀也可以在常規操作中出現(例如在解碼器輸入端處沒有信號時)，因此簡單地將幀的頻譜設置為零不足以提供關於這是替代幀和/或替代分組的事實的安全的指示。Since an explicit way of signaling frame loss is not provided in the AAC standard, in one implementation, the signaling method that is ignored by conventional decoders is used. Keep the data syntax and/or group syntax here. On the other hand, since this type of frame can also occur in normal operation (eg, when there is no signal at the decoder input), simply setting the spectrum of the frame to zero is not sufficient to provide a replacement frame and/or Or a safe indication of the fact of substituting a group.

對於幀是替代幀和/或替代分組的事實的指示向擴展解碼器提供了以下資訊：當前幀並非頻譜實際為零的幀，而是由於傳輸差錯而被引入到基站中以避免解碼器故障和/或語音鏈路故障的幀。The indication of the fact that the frame is a substitute frame and/or a substitute packet provides the extended decoder with the information that the current frame is not a frame whose spectrum is actually zero, but is introduced into the base station due to transmission errors to avoid decoder failure and / or frame of the voice link failure.

音頻編碼標準典型地提供了允許附加有效載荷傳輸的用戶專用資料區域，然而，其中該有效載荷被傳統的解碼器(即不具有擴展功能的基本解碼器)所忽略。在AAC標準中，如第六圖A中所定義的，這種用戶定義的有效載荷是所謂的"extension_payload"。如第六圖B所示，根據"extension_type"變數的值，該標準提供了不同的目的。第六圖A和第六圖B摘錄自標準ISO/IEC14496-3:2005(E)。在該標準中出於以下目的而提供了其中所述的填充單元(FILL)的使用。當針對所有音頻資料以及所有附加資料的位元總數小於該幀中用於實現目標位元率的所允許的最小位元數時，必須將填充單元添加至位元流中。當編碼器想要引入這種DRC資訊時，添加動態範圍控制位元(DRC位元)。如同標準所述，在正常情況下，避免填充位元並使用自由位元(free bit)來填滿位元儲存和/或位元保存庫。只有在位元儲存已滿時，才寫入填充位元。允許任何數目的填充位元。The audio coding standard typically provides a user-specific data area that allows for additional payload transmission, however, where the payload is ignored by conventional decoders (i.e., base decoders that do not have extended functionality). In the AAC standard, as defined in Figure 6A, this user-defined payload is the so-called "extension_payload". As shown in Figure 6B, the standard provides different purposes depending on the value of the "extension_type" variable. Figure 6 and Figure 6B are extracted from the standard ISO/IEC 14496-3:2005 (E). The use of the filling unit (FILL) described therein is provided in this standard for the following purposes. When the total number of bits for all audio data and all additional data is less than the allowed for the target bit rate in the frame The padding unit must be added to the bit stream when the minimum number of bits. Dynamic range control bits (DRC bits) are added when the encoder wants to introduce such DRC information. As stated by the standard, under normal circumstances, padding bits are avoided and free bits are used to fill the bit store and/or bit store. The padding bit is written only when the bit store is full. Allow any number of padding bits.

在本發明的一個實現中，如第七圖在70處所示，將"extension_type"設置為"0000"，以便與填充位元的標準使用相反地將對幀丟失的指示寫入"other_bits"欄位。In one implementation of the invention, as shown in Figure 7, at 70, "extension_type" is set to "0000" to write an indication of frame loss to the "other_bits" column as opposed to the standard use of padding bits. Bit.

在用於填充位元的標準中所提供的用戶專用資料區域用於提供對替代幀的信號通知，即用於容納對於替代幀的指示。然而，根據實現，可以根據"extension_type"值的不同設置來使用其他"extension_payload()"。由於優選地(通過將max_sfb設置為零)有效地傳輸零頻譜或者使用Zero Huffman碼本，因此存在可用於各個exteusion_payload()的足夠的位元。The user-specific material area provided in the standard for padding bits is used to provide a signalling of the substitute frame, ie for accommodating an indication of the substitute frame. However, depending on the implementation, other "extension_payload()" can be used depending on the different settings of the "extension_type" value. Since the zero spectrum is preferably transmitted efficiently (by setting max_sfb to zero) or using the Zero Huffman codebook, there are enough bits available for each extent_payload().

應當指出，在71處示出了針對替代分組的典型的遵從MPEG-4的資料流程和/或分組語法和/或幀語法，其中，如72處所示，使用變數"max_sfb=0"。優選地，還將獲得有效分組語法所需要的所有其他資料設置為零。然而，應當指出，該資料自身並非對於替代幀的可靠的指示。由於正常的編碼器不會寫入具有零頻譜的幀而是寫入特定的extension_payload，因此只有extension_payload 70將產生可靠的指示。It should be noted that a typical MPEG-4 compliant data flow and/or packet syntax and/or frame syntax for alternate packets is shown at 71, where the variable "max_sfb=0" is used as shown at 72. Preferably, all other data required to obtain a valid packet syntax is also set to zero. However, it should be noted that the material itself is not a reliable indication of a substitute frame. Since the normal encoder does not write a frame with a zero spectrum but writes a specific extension_payload, only the extension_payload 70 will produce a reliable indication.

第八圖示出了針對採樣速率為48kHZ、單聲道信號、位元速率為64kBit/s的新一代DECT的示例性替代幀。應當指出，由於位元流中的extension_payload並非“按位元組對齊”，因此在第八圖中不容易看出如第七圖在70處所示的extension_payload。The eighth figure shows an exemplary alternative frame for a new generation DECT with a sampling rate of 48 kHz, a mono signal, and a bit rate of 64 kBit/s. It should be noted that since the extension_payload in the bit stream is not "aligned by bit", the extension_payload as shown at 70 in the seventh figure is not easily seen in the eighth figure.

此外，應當指出，"extension_payload"中的位元組合包括也被稱為“差錯模式”的位元模式，該位元模式與"FRAME_LOSS"的ASCII碼相對應。通過該位元模式來保證不產生與extension_payload的其他用戶的衝突，這是因為另外的用戶幾乎不可能使用“FRAME_LOSS"的ASCII碼來用信號通知與"FRAME_LOSS"無關的事。Furthermore, it should be noted that the bit combination in "extension_payload" includes a bit pattern, also referred to as "error mode", which corresponds to the ASCII code of "FRAME_LOSS". This bit pattern is used to ensure that no conflict with other users of extension_payload occurs, because it is almost impossible for another user to use the ASCII code of "FRAME_LOSS" to signal something unrelated to "FRAME_LOSS".

下面將參照第五圖來討論可能在從分組的發射機至解碼器和/或至解碼器中的音頻呈現裝置的傳輸場景中的不同點處出現的分組或幀的不同順序。The different sequences of packets or frames that may occur at different points in the transmission scene from the transmitter of the packet to the decoder and/or to the audio presentation device in the decoder will be discussed below with reference to the fifth diagram.

第五圖A示出了編號為(i-1)、i、(i+1)、(i+2)的分組的序列。這種正確的分組或幀序列在面向分組的傳輸網路(例如網際網路)的輸出端處出現。The fifth diagram A shows a sequence of packets numbered (i-1), i, (i+1), (i+2). This correct sequence of packets or frames occurs at the output of a packet-oriented transport network, such as the Internet.

第五圖B示出了在基站的輸入端處的分組序列，其中可以看出，在第五圖B所考慮的時域部分中分組i尚未到達，但是分組順序已經變得混淆。其原因在於，分組i已經完全丟失或者獲得從發射機至接收機的非常長的路徑。另一方面，分組(i+2)已經獲得了非常有利的路徑，因此該分組在從發射機(即編碼器)至基站輸入端處的途中“追上”了分組(i+1)。Figure 5B shows a sequence of packets at the input of the base station, where it can be seen that packet i has not arrived in the time domain portion considered in Figure 5B, but the order of packets has become confusing. The reason for this is that the packet i has been completely lost or obtained a very long path from the transmitter to the receiver. On the other hand, the packet (i+2) has obtained a very advantageous path, so the packet "chasses up" the packet (i+1) on the way from the transmitter (i.e. encoder) to the base station input.

在第五圖B中，第一圖中的接收裝置11將在分組到達時以正確的順序再次對分組進行分類。此外，解碼裝置13將發現具有編號i的分組尚未出現或者發生錯誤。因此，如第五圖C所示，將產生針對編號i的替代分組。因此第五圖C示出了第一圖中的輸出裝置15所輸出的分組序列。在從根據第一圖的基站至根據第二圖的移動單元的路徑上，分組順序沒有改變。然而，如第二圖所示，擴展解碼器將識別編號為i的分組是替代分組。與第五圖D所示的產生音頻內容的正常呈現的其他分組相反地，針對替代分組產生返回至差錯隱藏措施的合成音頻內容。In the fifth diagram B, the receiving device 11 in the first figure will classify the packets again in the correct order when the packets arrive. Furthermore, the decoding device 13 will find that the packet with the number i has not yet appeared or an error has occurred. Therefore, as shown in the fifth figure C, an alternative packet for number i will be generated. The fifth figure C thus shows the sequence of packets output by the output device 15 in the first figure. On the path from the base station according to the first figure to the mobile unit according to the second figure, the order of the packets is not changed. However, as shown in the second figure, the extended decoder will recognize that the packet numbered i is a substitute packet. Contrary to the other packets that produce the normal presentation of the audio content shown in Figure D, the synthesized audio content returned to the error concealment measure is generated for the alternate packet.

本發明的方法可根據情況以硬體或軟體來實現。其實現可以基於數位儲存介質，具體是基於具有可被電子地讀出的控制信號的磁片或CD，該磁片或CD可與可編程電腦系統協作，以便執行相應的方法。總體上，本發明還可在電腦程式產品中實現，該電腦程式產品包括儲存在機器可讀載體上的程式碼，用於在該電腦程式產品在電腦上運行時執行該方法。換言之，本發明因此可以被實現為具有程式碼的電腦程式，用於在該電腦程式在電腦上運行時執行該方法。The method of the present invention can be implemented in a hardware or a soft body depending on the situation. The implementation may be based on a digital storage medium, in particular based on a magnetic disk or CD having a control signal that can be electronically read out, which may cooperate with a programmable computer system to perform a corresponding method. In general, the present invention can also be implemented in a computer program product comprising a program code stored on a machine readable carrier for performing the method when the computer program product is run on a computer. In other words, the invention can thus be implemented as a computer program with a code for performing the method while the computer program is running on the computer.

10‧‧‧基站10‧‧‧ base station

11‧‧‧接收裝置11‧‧‧ Receiving device

12‧‧‧網際網路12‧‧‧Internet

13‧‧‧檢測裝置13‧‧‧Detection device

14‧‧‧用於提供替代分組的裝置14‧‧‧Devices for providing alternative groupings

15‧‧‧輸出裝置15‧‧‧Output device

16‧‧‧天線16‧‧‧Antenna

17‧‧‧分組線路17‧‧‧Packet line

18‧‧‧替代分組線路18‧‧‧Alternative packet line

19‧‧‧控制線路19‧‧‧Control lines

20‧‧‧移動單元20‧‧‧Mobile unit

21‧‧‧接收機21‧‧‧ Receiver

22‧‧‧檢測器22‧‧‧Detector

23‧‧‧差錯隱藏裝置23‧‧‧Error concealing device

24‧‧‧音頻呈現裝置24‧‧‧Audio presentation device

25‧‧‧輸出單元25‧‧‧Output unit

26‧‧‧分組線路26‧‧‧Packet line

27‧‧‧替代分組線路27‧‧‧Alternative packet line

30‧‧‧時頻轉換級30‧‧‧Time-frequency conversion stage

31‧‧‧量化器31‧‧‧Quantifier

32‧‧‧心理聲學模型32‧‧‧ psychoacoustic model

33‧‧‧熵編碼器33‧‧‧Entropy encoder

34‧‧‧分組打包器34‧‧‧Packer

40‧‧‧分組拆包器40‧‧‧Group unpacker

41‧‧‧熵編碼器41‧‧‧Entropy encoder

42‧‧‧反量化器42‧‧‧Reverse Quantizer

43‧‧‧頻時轉換43‧‧‧frequency conversion

第一圖示出了用於傳輸資料分組序列的設備的實現的電路框圖；第二圖是用於對資料分組序列進行解碼的解碼器的電路框圖；第三圖是用於產生資料分組序列的音頻解碼器的電路框圖；第四圖特別示出了第二圖的音頻呈現裝置的更特定的實現；第五圖A示出了編碼器輸出的資料分組序列；第五圖B示出了基站接收到的資料分組序列；第五圖C示出了由基站輸出並由移動單元接收的、已***替代分組的資料分組序列；第五圖D示出了音頻呈現裝置中所產生的音頻內容的序列；第六圖A示出了根據ISO/IEC 14496-3:2005 (E)MPEG4的extension_payload的語法；第六圖B示出了用於例證extension_type欄位的值的表；第七圖示出了用於基於變換的音頻編碼器/解碼器的示例性分組語法；第八圖示出了包括有效分組語法和預定內容特性的替代分組的示例。The first figure shows a block circuit diagram of an implementation of a device for transmitting a sequence of data packets; the second figure is a circuit block diagram of a decoder for decoding a sequence of data packets; the third figure is for generating data packets A block diagram of a sequence of audio decoders; a fourth diagram particularly showing a more specific implementation of the audio rendering apparatus of the second diagram; a fifth diagram A showing a sequence of data packets output by the encoder; a sequence of data packets received by the base station; a fifth diagram C shows a sequence of data packets of the inserted substitute packets output by the base station and received by the mobile unit; and FIG. 5D shows the generation of the audio presentation device a sequence of audio content; a sixth diagram A shows the syntax of extension_payload according to ISO/IEC 14496-3:2005 (E) MPEG4; a sixth diagram B shows a table for illustrating the value of the extension_type field; The figure shows an exemplary packet syntax for a transform-based audio encoder/decoder; the eighth figure shows an example of an alternate packet including a valid packet syntax and predetermined content characteristics.

Claims

一種用於將表示音頻信號的資料分組序列傳輸到基本解碼器的設備，包括：接收裝置(11)，用於接收所述序列中的分組，分組語法是針對所述分組指定的；檢測裝置(13)，用於檢測所述分組序列中的分組是否丟失或發生錯誤；提供裝置(14)，用於提供替代分組以替代發生錯誤的分組或丟失的分組，所述替代分組是關於所述分組語法的有效分組，所述替代分組的音頻內容是所述所述基本解碼器能夠再現的預定的音頻內容並且具有預定的內容特性，並且所述替代分組包含針對所述分組是替代分組的事實的指示，其中，所述指示被實現為使得所述指示是可被所述基本解碼器所忽略或解釋的，以使得可以根據所述預定的內容特性如同有效分組一樣地對所述替代分組進行解碼；以及輸出裝置(15)，用於輸出所述分組序列，所述分組序列包括至少一個接收到的分組以及至少一個替代分組。 An apparatus for transmitting a sequence of data packets representing an audio signal to a base decoder, comprising: receiving means (11) for receiving packets in said sequence, packet syntax is specified for said packet; detecting means ( 13) for detecting whether a packet in the sequence of packets is lost or an error occurs; providing means (14) for providing a substitute packet to replace the erroneous packet or the lost packet, the substitute packet being related to the packet An efficient grouping of grammars, the audio content of the alternate packet being predetermined audio content that the base decoder is capable of reproducing and having predetermined content characteristics, and the substitute packet containing the fact that the packet is a substitute packet An indication, wherein the indication is implemented such that the indication is ignorable or interpretable by the base decoder such that the substitute packet can be decoded as a valid packet according to the predetermined content characteristic And an output device (15) for outputting the sequence of packets, the sequence of packets comprising at least one received packet And at least one alternative grouping.

依據申請專利範圍第1項所述的設備，其中，可由接收裝置(11)接收的序列中的分組來自基於變換的音頻編碼器，並包括所述音頻信號的時域部分中的短時頻譜，以及其中，所述預定的內容特性是所有的頻譜值等於零或者共同表示小於心理聲學靜止聽音臨界值所表示的能量的兩倍的能量。 The apparatus of claim 1, wherein the packets in the sequence receivable by the receiving device (11) are from a transform-based audio encoder and include a short-term spectrum in a time domain portion of the audio signal, And wherein the predetermined content characteristic is that all of the spectral values are equal to zero or collectively represent less than twice the energy represented by the psychoacoustic still listening threshold.

依據申請專利範圍第1項所述的設備，其中，所述提供裝置(14)包括：記憶體，用於儲存所述替代分組；記憶體讀取器，用於只要所述檢測裝置(13)檢測到發生錯誤的分組或丟失的分組時，就從所述記憶體中讀取所述替代分組。 The device according to claim 1, wherein the providing device (14) comprises: a memory for storing the substitute packet; and a memory reader for the detecting device (13) When an erroneous packet or a lost packet is detected, the substitute packet is read from the memory.

依據申請專利範圍第1項所述的設備，其中，所述接收裝置(11)是電話基站的輸入介面，並且可連接至被實現用於基於分組的資料傳輸的網路。 The device of claim 1, wherein the receiving device (11) is an input interface of a telephone base station and is connectable to a network implemented for packet based data transmission.

依據申請專利範圍第1項所述的設備，其中，所述替代分組具有所述指示所位於的有效載荷區域。 The device of claim 1, wherein the substitute packet has a payload area in which the indication is located.

依據申請專利範圍第1項所述的設備，其中，所述分組語法被實現為定義擴展有效載荷欄位，以及所述指示是由所述擴展有效載荷欄位中的資料形成的。 The device of claim 1, wherein the packet syntax is implemented to define an extended payload field, and the indication is formed by data in the extended payload field.

依據申請專利範圍第6項所述的設備，其中，所述資料是根據字母代碼產生的，所述資料具有指示資料丟失的含義。 The device of claim 6, wherein the material is generated based on an alphabetic code, the material having a meaning indicating loss of data.

依據申請專利範圍第7項所述的設備，其中，所述資料表示了表現“FRAME_LOSS”或“資料丟失”。 The device of claim 7, wherein the material represents a performance "FRAME_LOSS" or "data loss."

依據申請專利範圍第7項所述的設備，其中，所述字母代碼是ASCII碼。 The device of claim 7, wherein the letter code is ASCII code.

依據申請專利範圍第1項所述的設備，其中，所述分組序列由音頻編碼器根據MPEG-1層3或MPEG-4 AAC產生，位元儲存功能被禁用。 The device according to claim 1, wherein the packet sequence is generated by an audio encoder according to MPEG-1 Layer 3 or MPEG-4 AAC, and the bit storage function is disabled.

依據申請專利範圍第1項所述的設備，所述設備被實現為基站。 According to the device of claim 1, the device is implemented as a base station.

依據申請專利範圍第1項所述的設備，其中，所述分組語法包括填充資料欄位，在對所述音頻信號的一部分進行編碼不需要為所述幀所提供的位元組的最小數目時，由基本編碼器填充所述填充資料欄位，以及其中，所述指示由所述填充資料欄位中的預定的位元組合來表示。 The device of claim 1, wherein the packet syntax comprises a padding data field, when encoding a portion of the audio signal without requiring a minimum number of bytes provided for the frame Filling the padding data field with a base encoder, and wherein the indication is represented by a predetermined combination of bit bits in the padding data field.

依據申請專利範圍第1項所述的設備，其中，所述檢測裝置(13)被實現用於在預定持續時間內等待具有序列位置指示的資料分組，以及用於在經過了所述預定持續時間而未檢測到所述資料分組時，通過信號向所述提供裝置(14)通知分組丟失。 The apparatus of claim 1, wherein the detecting means (13) is implemented to wait for a data packet having a sequence position indication for a predetermined duration and for elapse of the predetermined duration When the data packet is not detected, the packet loss is notified to the providing device (14) by a signal.

依據申請專利範圍第13項所述的設備，其中，所尋找的序列位置資訊是由在先或後續的有效分組的序列位置指示設置的。 The device of claim 13, wherein the sought sequence position information is set by a sequence position indication of a prior or subsequent valid packet.

依據申請專利範圍第13項所述的設備，其中，所述預定持續時間可通過QoS請求(19)來設置和預定。 The device of claim 13, wherein the predetermined duration is settable and predetermined by a QoS request (19).

依據申請專利範圍第15項所述的設備，其中，當所述QoS請求具有較小的延遲時，所述預定持續時間較小，並且當所述QoS請求允許較大的延遲時，所述預定持續時間較大。 The device of claim 15, wherein the predetermined duration is small when the QoS request has a small delay, and the predetermined time when the QoS request allows a larger delay The duration is large.

依據申請專利範圍第1項所述的設備，其中，所述輸出裝置(15)被實現用於將所述分組序列作為未中斷的分組序列以及表示完整的連續序列的替代分組而輸出。 The device of claim 1, wherein the output device (15) is implemented to use the sequence of packets as uninterrupted The sequence of packets and the alternative packets representing the complete contiguous sequence are output.

一種用於將表示音頻信號的資料分組序列傳輸到擴展解碼器的設備，其中所述擴展解碼器與基本解碼器相比具有擴展功能，包括：接收裝置(11)，用於接收所述序列中的分組，分組語法是針對所述分組指定的；檢測裝置(13)，用於檢測所述分組序列中的分組是否丟失或發生錯誤；提供裝置(14)，用於提供替代分組以替代發生錯誤的分組或丟失的分組，所述替代分組是關於所述分組語法的有效分組，所述替代分組的音頻內容是所述基本解碼器能夠再現的預定的音頻內容並且具有預定的內容特性，並且所述替代分組包含針對所述分組是替代分組的事實的指示，其中，所述指示被實現為使得所述指示是可被所述基本解碼器所忽略或解釋的，以使得可以根據所述預定的內容特性如同有效分組一樣地對所述替代分組進行解碼，以及，所述指示是可由所述擴展解碼器解釋的，以執行產生所述替代分組的內容的差錯隱藏措施，所述替代分組的內容具有與所述預定的內容特性不同的內容特性；以及輸出裝置(15)，用於輸出所述分組序列，所述分組序列包括至少一個接收到的分組以及至少一個替代分組。 An apparatus for transmitting a sequence of data packets representing an audio signal to an extended decoder, wherein the extended decoder has an extended function compared to the base decoder, comprising: receiving means (11) for receiving in the sequence a packet, a packet grammar is specified for the packet; a detecting means (13) for detecting whether a packet in the sequence of packets is missing or an error occurs; providing means (14) for providing a substitute packet instead of generating an error a packet or a lost packet, the substitute packet being a valid packet with respect to the packet syntax, the audio content of the substitute packet being predetermined audio content that the base decoder is capable of reproducing and having predetermined content characteristics, and The substitute packet includes an indication of the fact that the packet is a substitute packet, wherein the indication is implemented such that the indication is ignorable or interpretable by the base decoder such that the predetermined The content characteristic decodes the substitute packet as a valid packet, and the indication is decodable by the extension Explained to perform error concealment measures for generating content of the replacement packet, the content of the replacement packet having content characteristics different from the predetermined content characteristics; and output means (15) for outputting the sequence of packets The sequence of packets includes at least one received packet and at least one substitute packet.

根據申請專利範圍第18項所述的設備，其中，可由接收裝置(11)接收的序列中的分組來自基於變換的音頻編碼器，並包括所述音頻信號的時域部分中的短時頻譜，以及其中，所述預定的內容特性是所有的頻譜值等於零或者共同表示小於心理聲學靜止聽音臨界值所表示的能量的兩倍的能量。 The device of claim 18, wherein the packets in the sequence receivable by the receiving device (11) are from a transform-based audio encoder and include a short-term spectrum in a time domain portion of the audio signal, And wherein the predetermined content characteristic is that all of the spectral values are equal to zero or collectively represent less than twice the energy represented by the psychoacoustic still listening threshold.

根據申請專利範圍第18項所述的設備，其中，所述提供裝置(14)包括：記憶體，用於存儲所述替代分組；記憶體讀取器，用於只要所述檢測裝置(13)檢測到發生錯誤的分組或丟失的分組時，就從所述記憶體中讀取所述替代分組。 The device according to claim 18, wherein the providing device (14) comprises: a memory for storing the substitute packet; a memory reader for as long as the detecting device (13) When an erroneous packet or a lost packet is detected, the substitute packet is read from the memory.

根據申請專利範圍第18項所述的設備，其中，所述接收裝置(11)是電話基站的輸入介面，並且可連接至被實現用於基於分組的資料傳輸的網路。 The device of claim 18, wherein the receiving device (11) is an input interface of a telephone base station and is connectable to a network implemented for packet based data transmission.

根據申請專利範圍第18項所述的設備，其中，所述替代分組具有所述指示所位於的有效載荷區域。 The device of claim 18, wherein the substitute packet has a payload area in which the indication is located.

根據申請專利範圍第18項所述的設備，其中，所述分組語法被實現為定義擴展有效載荷欄位，以及所述指示是由所述擴展有效載荷欄位中的資料形成的。 The device of claim 18, wherein the packet syntax is implemented to define an extended payload field, and the indication is formed by data in the extended payload field.

根據申請專利範圍第23項所述的設備，其中，所述資料是根據字母代碼產生的，所述資料具有指示資料丟失的含義。 The device of claim 23, wherein the material is generated based on an alphabetic code, the material having a meaning indicating loss of data.

根據申請專利範圍第24項所述的設備，其中，所述資料表示了表現“FRAME_LOSS”或“資料丟失”。 The device of claim 24, wherein the material represents a performance "FRAME_LOSS" or "data loss."

根據申請專利範圍第24項所述的設備，其中，所述字母代碼是ASCII碼。 According to the device of claim 24, wherein The letter code is ASCII code.

根據申請專利範圍第18項所述的設備，其中，所述分組序列由音頻編碼器根據MPEG-1層3或MPEG-4 AAC產生，位元儲存功能被禁用。 The device according to claim 18, wherein the packet sequence is generated by an audio encoder according to MPEG-1 Layer 3 or MPEG-4 AAC, and the bit storage function is disabled.

根據申請專利範圍第18項所述的設備，所述設備被實現為基站。 According to the device of claim 18, the device is implemented as a base station.

根據申請專利範圍第18項所述的設備，其中，所述分組語法包括填充資料欄位，在對所述音頻信號的一部分進行編碼不需要為所述幀所提供的位元組的最小數目時，由基本編碼器填充所述填充資料欄位，以及其中，所述指示由所述填充資料欄位中的預定的位元組合來表示。 The device of claim 18, wherein the packet syntax comprises a padding data field, when encoding a portion of the audio signal without requiring a minimum number of bytes provided for the frame Filling the padding data field with a base encoder, and wherein the indication is represented by a predetermined combination of bit bits in the padding data field.

根據申請專利範圍第18項所述的設備，其中，所述檢測裝置(13)被實現用於在預定持續時間內等待具有序列位置指示的資料分組，以及用於在經過了所述預定持續時間而未檢測到所述資料分組時，通過信號向所述提供裝置(14)通知分組丟失。 The device of claim 18, wherein the detecting means (13) is implemented to wait for a data packet having a sequence position indication for a predetermined duration and for elapse of the predetermined duration When the data packet is not detected, the packet loss is notified to the providing device (14) by a signal.

根據申請專利範圍第30項所述的設備，其中，所尋找的序列位置資訊是由在先或後續的有效分組的序列位置指示設置的。 The device of claim 30, wherein the sought sequence position information is set by a sequence position indication of a prior or subsequent valid packet.

根據申請專利範圍第30項所述的設備，其中，所述預定持續時間可通過QoS請求(19)來設置和預定。 The device of claim 30, wherein the predetermined duration is settable and predetermined by a QoS request (19).

根據申請專利範圍第32項所述的設備，其中，當所述QoS請求具有較小的延遲時，所述預定持續時間較小，並且當所述QoS請求允許較大的延遲時，所述預定持續時間較大。 The device of claim 32, wherein the predetermined duration is greater when the QoS request has a smaller delay Small, and when the QoS request allows for a large delay, the predetermined duration is large.

根據申請專利範圍第18項所述的設備，其中，所述輸出裝置(15)被實現用於將所述分組序列作為未中斷的分組序列以及表示完整的連續序列的替代分組而輸出。 The device of claim 18, wherein the output device (15) is implemented to output the sequence of packets as an uninterrupted sequence of packets and an alternate packet representing a complete contiguous sequence.

一種用於將表示音頻信號的資料分組序列傳輸到基本解碼器的方法，包括：接收(11)所述序列中的分組，分組語法是針對所述分組指定的；檢測(13)所述分組序列中的分組是否丟失或發生錯誤；提供(14)替代分組以替代發生錯誤的分組或丟失的分組，所述替代分組是關於所述分組語法的有效分組，所述替代分組的音頻內容是所述基本解碼器能夠再現的預定的音頻內容並且具有預定的內容特性，並且所述替代分組包含針對所述分組是替代分組的事實的指示，其中，所述指示被實現為使得所述指示是可被基本解碼器忽略或解釋的，以使得根據所述預定的內容特性如同有效分組一樣地對所述替代分組進行解碼；以及輸出(15)所述分組序列，所述分組序列包括至少一個接收到的分組以及至少一個替代分組。 A method for transmitting a sequence of data packets representing an audio signal to a base decoder, comprising: receiving (11) a packet in the sequence, a packet syntax is specified for the packet; detecting (13) the sequence of packets Whether the packet in the packet is lost or an error occurs; providing (14) a substitute packet in place of the erroneous packet or the lost packet, the substitute packet being a valid packet with respect to the packet grammar, the audio content of the substitute packet being The predetermined audio content that the base decoder is capable of reproducing and having predetermined content characteristics, and the substitute packet includes an indication of the fact that the packet is a substitute packet, wherein the indication is implemented such that the indication is The basic decoder ignores or interprets such that the substitute packet is decoded as a valid packet according to the predetermined content characteristic; and outputs (15) the sequence of packets, the sequence of packets including at least one received Grouping and at least one alternative grouping.

一種用於將表示音頻信號的資料分組序列傳輸到擴展解碼器的方法，其中所述擴展解碼器與基本解碼器相比具有擴展功能，包括：接收(11)所述序列中的分組，分組語法是針對所述分組指定的；檢測(13)所述分組序列中的分組是否丟失或發生錯誤；提供(14)替代分組以替代發生錯誤的分組或丟失的分組，所述替代分組是關於所述分組語法的有效分組，所述替代分組的音頻內容是所述基本解碼器能夠再現的預定的音頻內容並且具有預定的內容特性，並且所述替代分組包含針對所述分組是替代分組的事實的指示，其中，所述指示被實現為使得所述指示是可被基本解碼器忽略或解釋的，以使得根據所述預定的內容特性如同有效分組一樣地對所述替代分組進行解碼，以及，所述指示是可由所述擴展解碼器解釋的，以執行產生所述替代分組的內容的差錯隱藏措施，所述替代分組的內容具有與所述預定的內容特性不同的內容特性；以及輸出(15)所述分組序列，所述分組序列包括至少一個接收到的分組以及至少一個替代分組。 A method for transmitting a sequence of data packets representing an audio signal to an extended decoder, wherein the extended decoder has extended functionality compared to the base decoder, including: Receiving (11) a packet in the sequence, the packet syntax is specified for the packet; detecting (13) whether a packet in the sequence of packets is missing or an error occurs; providing (14) a substitute packet to replace the packet in which the error occurred Or a lost packet, the substitute packet being a valid packet with respect to the packet syntax, the audio content of the substitute packet being predetermined audio content that the base decoder is capable of reproducing and having predetermined content characteristics, and the replacement The packet contains an indication of the fact that the packet is a substitute packet, wherein the indication is implemented such that the indication is ignorable or interpretable by the base decoder such that the predetermined content characteristics are as valid as the valid packet Decoding the alternate packet, and the indication is interpretable by the extended decoder to perform error concealment measures for generating content of the substitute packet, the content of the substitute packet having the predetermined a content characteristic having different content characteristics; and outputting (15) the sequence of packets, the sequence of packets including at least one received Grouping and at least one alternative grouping.

一種用於對分組序列進行解碼的擴展解碼器，其中所述擴展解碼器與基本解碼器相比具有擴展功能，包括：接收機(21)，用於接收所述序列中的分組，分組語法是針對所述分組指定的，所述序列包括至少一個資料分組和至少一個替代分組，所述替代分組是關於所述分組語法的有效分組，所述替代分組的音頻內容是所述基本解碼器能夠再現的預定音頻內容並且具有預定的內容特性，並且所述替代分組包含針對所述分組是替代分組的事實的指示，其中，所述指示使得所述替代分組能夠被所述基本解碼器接收並且所述替代分組能夠根據所述預定的內容特性如同有效分組一樣地解碼，並使得所述擴展解碼器被實現為解釋所述指示以執行產生所述替代分組的內容的差錯隱藏措施，所述替代分組的內容具有與所述預定的內容特性不同的內容特性；檢測器(22)，用於檢測分組是否包含所述指示並因此是替代分組；差錯隱藏裝置(23)，利用所述差錯隱藏裝置合成所述替代分組的音頻內容，所述音頻內容的內容特性不同於所述預定的內容特性；以及音頻呈現裝置(24)，用於在所述分組並非替代分組時呈現分組的所述音頻內容，以及用於在所述分組是替代分組時呈現所述合成音頻內容。 An extended decoder for decoding a sequence of packets, wherein the extended decoder has an extended function compared to a base decoder, comprising: a receiver (21) for receiving packets in the sequence, the packet syntax is The sequence is specified for the packet, the sequence comprising at least one data packet and at least one substitute packet, the substitute packet being a valid packet with respect to the packet syntax, the audio content of the substitute packet being capable of being reproduced by the base decoder Predetermined audio content and has predetermined content characteristics, and The substitute packet includes an indication of a fact that the packet is a substitute packet, wherein the indication enables the substitute packet to be received by the base decoder and the substitute packet is capable of being valid according to the predetermined content characteristic The packet is decoded identically, and the extended decoder is implemented to interpret the indication to perform error concealment measures for generating content of the replacement packet, the content of the replacement packet having content different from the predetermined content characteristic a detector (22) for detecting whether the packet contains the indication and is therefore a substitute packet; an error concealing means (23) synthesizing the audio content of the substitute packet with the error concealing means, the audio content The content characteristic is different from the predetermined content characteristic; and an audio rendering device (24) for presenting the audio content of the packet when the packet is not a substitute packet, and for presenting when the packet is a substitute packet Synthesize audio content.

依據申請專利範圍第37項所述的擴展解碼器，其中，所述預定的內容特性是零頻譜，所述分組序列中的接收到的分組由基於變換的音頻編碼器產生，以及所述音頻呈現裝置(24)包括基於變換的音頻解碼器(41，42，43)。 The extended decoder of claim 37, wherein the predetermined content characteristic is a zero spectrum, the received packet in the sequence of packets is generated by a transform-based audio encoder, and the audio presentation The device (24) includes a transform based audio decoder (41, 42, 43).

依據申請專利範圍第38項所述的擴展解碼器，其中，所述差錯隱藏裝置(23)被實現用於產生合成頻譜值，以及其中，所述音頻呈現裝置被實現用於將所述合成頻譜值轉換為時域表示(43)。 An extended decoder according to claim 38, wherein the error concealing means (23) is implemented to generate a synthesized spectral value, and Wherein the audio rendering device is implemented to convert the synthesized spectral values into a time domain representation (43).

依據申請專利範圍第37項所述的擴展解碼器，其中，所述差錯隱藏裝置(23)被實現用於通過對來自先前或後續的完整或隱藏的音頻分組的音頻內容的外插來產生所述合成音頻內容，或者被實現用於通過對先前的完整或隱藏的分組以及後續的完整或隱藏的分組的音頻內容的內插來產生所述合成音頻內容。 The spread decoder of claim 37, wherein the error concealing means (23) is implemented for generating an extrapolation of audio content from previous or subsequent complete or hidden audio packets. The synthesized audio content is either implemented for generating the synthesized audio content by interpolation of previous complete or hidden packets and subsequent complete or hidden grouped audio content.

一種用於在擴展解碼器中對分組序列進行解碼的方法，其中所述擴展解碼器與基本解碼器相比具有擴展功能，包括：接收(21)所述序列中的分組，分組語法是針對所述分組指定的，所述序列包括至少一個資料分組和至少一個替代分組，所述替代分組是關於所述分組語法的有效分組，所述替代分組的音頻內容是所述基本解碼器能夠再現的預定音頻內容並且具有預定的內容特性，並且所述替代分組包含針對所述分組是替代分組的事實的指示，其中，所述指示使得所述替代分組能夠被所述基本解碼器接收並且所述替代分組能夠根據所述預定的內容特性如同有效分組一樣地解碼，並使得所述擴展解碼器能夠解釋所述指示以執行產生所述替代分組的內容的差錯隱藏措施，所述替代分組的內容具有與所述預定的內容特性不同的內容特性；檢測(22)所述分組是否包含所述指示並因此是替代分組；針對所述替代分組，通過所述差錯隱藏措施來合成(23)音頻內容，所述音頻內容的內容特性不同於所述預定的內容特性；以及在所述分組並非替代分組時呈現(24)分組的所述音頻內容，以及在所述分組是替代分組時呈現所述合成音頻內容。 A method for decoding a sequence of packets in an extended decoder, wherein the extended decoder has an extended function compared to the base decoder, comprising: receiving (21) a packet in the sequence, the packet syntax is for The sequence specifies that the sequence includes at least one data packet and at least one substitute packet, the substitute packet being a valid packet with respect to the packet syntax, the audio content of the substitute packet being a reservation that the base decoder can reproduce Audio content and having predetermined content characteristics, and the substitute packet includes an indication of a fact that the packet is a substitute packet, wherein the indication enables the substitute packet to be received by the base decoder and the substitute packet Capable of decoding as the valid packet according to the predetermined content characteristic, and enabling the extension decoder to interpret the indication to perform error concealment measures for generating content of the replacement packet, the content of the replacement packet having Content characteristics that are different in predetermined content characteristics; Detecting (22) whether the packet contains the indication and is therefore a substitute packet; for the substitute packet, synthesizing (23) audio content by the error concealment measure, the content characteristic of the audio content being different from the predetermined Content characteristics; and presenting (24) the audio content of the packet when the packet is not a substitute packet, and presenting the synthesized audio content when the packet is a substitute packet.

一種電腦可讀程式，包括程式碼，用於當依據申請專利範圍第35、36項中的任一項所述的方法在電腦上運行時執行所述方法。 A computer readable program comprising a code for performing the method when run on a computer in accordance with the method of any one of claims 35, 36.