Patent Description:
Coded audio usually comes in chunks of samples, often <NUM>, <NUM> or <NUM> samples in number per chunk. Such chunks are called frames in the following. In the context of MPEG audio codecs like AAC or MPEG-H 3D Audio, these chunks/frames are called granules, the encoded chunks/frames are called access units (AU) and the decoded chunks are called composition units (CU). In transport systems the audio signal is only accessible and addressable in granularity of these coded chunks (access units). It would be favorable, however, to be able to address the audio data at some final granularity, especially for purposes like stream splicing or changes of the configuration of the coded audio data, synchronous and aligned to another stream such as a video stream, for example.

What is known so far is the discarding of some samples of a coding unit. The MPEG-<NUM> file format, for example, has so-called edit lists that can be used for the purpose of discarding audio samples at the beginning and the end of a coded audio file/bitstream [<NUM>]. Disadvantageously, this edit list method works only with the MPEG-<NUM> file format, i.e. is file format specific and does not work with stream formats like MPEG-<NUM> transport streams. Beyond that, edit lists are deeply embedded in the MPEG-<NUM> file format and accordingly cannot be easily modified on the fly by stream splicing devices. In AAC [<NUM>], truncation information may be inserted into the data stream in the form of extension_payload. Such extension_payload in a coded AAC access unit is, however, disadvantageous in that the truncation information is deeply embedded in the AAC AU and cannot be easily modified on the fly by stream splicing devices. <CIT> discloses a device that decodes encoded data with information on the amount of invalid data at the beginning and end of a frame.

Accordingly, it is an object of the present invention to provide a concept for audio splicing which is more efficient in terms of, for example, procedural complexity of the splicing process at stream splicers, and/or audio decoders.

This object is achieved by the subject matter of the independent claims attached herewith.

Throughout this description, reference is made to an audio data stream, an encoder and encoding method, which are not part of the present invention but are useful for understanding the decoder and decoding method of the invention as claimed.

The invention of the present application is inspired by the idea that audio splicing may be rendered more effectively by the use of one or more truncation unit packets inserted into the audio data stream so as to indicate to an audio decoder, for a predetermined access unit, an end portion of an audio frame with which the predetermined access unit is associated, as to be discarded in playout.

In accordance with an aspect of the present application, an audio data stream is initially provided with such a truncation unit packet in order to render the thus provided audio data stream more easily spliceable at the predetermined access unit at a temporal granularity finer than the audio frame length. The one or more truncation unit packets are, thus, addressed to audio decoder and stream splicer, respectively. In accordance with embodiments, a stream splicer simply searches for such a truncation unit packet in order to locate a possible splice point. The stream splicer sets the truncation unit packet accordingly so as to indicate an end portion of the audio frame with which the predetermined access unit is associated, to be discarded in playout, cuts the first audio data stream at the predetermined access unit and splices the audio data stream with another audio data stream so as to abut each other at the predetermined access unit. As the truncation unit packet is already provided within the spliceable audio data stream, no additional data is to be inserted by the splicing process and accordingly, bitrate consumption remains unchanged insofar.

Alternatively, a truncation unit packet may be inserted at the time of splicing. Irrespective of initially providing an audio data stream with a truncation unit packet or providing the same with a truncation unit packet at the time of splicing, a spliced audio data stream has such truncation unit packet inserted thereinto with the end portion being a trailing end portion in case of the predetermined access unit being part of the audio data stream leading the splice point and a leading end portion in case of the predetermined access unit being part of the audio data stream succeeding the splice point.

In particular, preferred embodiments of the present application are described below with respect to the figures, among which:.

<FIG> shows an exemplary portion out of an audio data stream in order to illustrate the problems occurring when trying to splice the respective audio data stream with another audio data stream. Insofar, the audio data stream of <FIG> forms a kind of basis of the audio data streams shown in the subsequent figures. Accordingly, the description brought forward with the audio data stream of <FIG> is also valid for the audio data streams described further below.

The audio data stream of <FIG> is generally indicated using reference sign <NUM>. The audio data stream has encoded there into an audio signal <NUM>. In particular, the audio signal <NUM> is encoded into audio data stream in units of audio frames <NUM>, i.e. temporal portions of the audio signal <NUM> which may, as illustrated in <FIG>, be non-overlapping and abut each other temporally, or alternatively overlap each other. The way the audio signal <NUM> is, in units of the audio frames <NUM>, encoded audio data stream <NUM> may be chosen differently: transform coding may be used in order to encode the audio signal in the units of the audio frames <NUM> into data stream <NUM>. In that case, one or several spectral decomposition transformations may be applied onto the audio signal of audio frame <NUM>, with one or more spectral decomposition transforms temporally covering the audio frame <NUM> and extending beyond its leading and trailing end. The spectral decomposition transform coefficients are contained within the data stream so that the decoder is able to reconstruct the respective frame by way of inverse transformation. The mutually and even beyond audio frame boundaries overlapping transform portions in units of which the audio signal is spectrally decomposed are windowed with so called window functions at encoder and/or decoder side so that a so-called overlap-add process at the decoder side according to which the inversely transformed signaled spectral composition transforms are overlapped with each other and added, reveals the reconstruction of the audio signal <NUM>.

Alternatively, for example, the audio data stream <NUM> has audio signal <NUM> encoded thereinto in units of the audio frames <NUM> using linear prediction, according to which the audio frames are coded using linear prediction coefficients and the coded representation of the prediction residual using, in turn, long term prediction (LTP) coefficients like LTP gain and LTP lag, codebook indices and/or a transform coding of the excitation (residual signal). Even here, the reconstruction of an audio frame <NUM> at the decoding side may depend on a coding of a preceding frame or into, for example, temporal predictions from one audio frame to another or the overlap of transform windows for transform coding the excitation signal or the like. The circumstance is mentioned here, because it plays a role in the following description.

For transmission and network handling purposes, the audio data stream <NUM> is composed of a sequence of payload packets <NUM>. Each of the payload packets <NUM> belongs to a respective one of the sequence of access units <NUM> into which the audio data stream <NUM> is partitioned along stream order <NUM>. Each of the access units <NUM> is associated with a respective one of the audio frames <NUM> as indicated by double-headed arrows <NUM> in <FIG>. As illustrated in <FIG>, the temporal order of the audio frames <NUM> may coincide with the order of the associated audio frames <NUM> in data stream <NUM>: an audio frame <NUM> immediately succeeding another frame may be associated with an access unit in data stream <NUM> immediately succeeding the access unit of the other audio frame in data stream <NUM>.

That is, as depicted in <FIG>, each access unit <NUM> may have one or more payload packets <NUM>. The one or more payload packets <NUM> of a certain access unit <NUM> has/have encoded thereinto the aforementioned coding parameters describing the associated frame <NUM> such as spectral decomposition transform coefficients, LPCs, and/or a coding of the excitation signal.

The audio data stream <NUM> may also comprise timestamp information <NUM> which indicates for each access unit <NUM> of the data stream <NUM> this timestamp ti at which the audio frame i with which the respective access unit <NUM> AUi is associated, is to be played out. The timestamp information <NUM> may, as illustrated in <FIG>, be inserted into one of the one or more packets <NUM> of each access unit <NUM> so as to indicate the timestamp of the associated audio frame, but different solutions are feasible as well, such as the insertion of the timestamp information ti of an audio frame i into each of the one or more packets of the associated access unit AUi.

Owing to the packetization, the access unit partitioning and the timestamp information <NUM>, the audio data stream <NUM> is especially suitable for being streamed between encoder and decoder. That is, the audio data stream <NUM> of <FIG> is an audio data stream of the stream format. The audio data stream of <FIG> may, for instance, be an audio data stream according to MPEG-H 3D Audio or MHAS [<NUM>].

In order to ease the transport/network handling, packets <NUM> may have byte-aligned sizes and packets <NUM> of different types may be distinguished. For example, some packets <NUM> may relate to a first audio channel or a first set of audio channels and have a first packet type associated therewith, while packets having another packet type associated therewith have encoded thereinto another audio channel or another set of audio channels of audio signal <NUM> encoded thereinto. Even further packets may be of a packet type carrying seldom changing data such as configuration data, coding parameters being valid, or being used by, sequence of access units. Even other packets <NUM> may be of a packet type carrying coding parameters valid for the access unit to which they belong, while other payload packets carry codings of samples values, transform coefficients, LPC coefficients, or the like. Accordingly, each packet <NUM> may have a packet type indicator therein which is easily accessible by intermediate network entities and the decoder, respectively. The TU packets described hereinafter may be distinguishable from the payload packets by packet type.

As long as the audio data stream <NUM> is transmitted as it is, no problem occurs. However, imagine that the audio signal <NUM> is to be played out at decoding side until some point in time exemplarily indicated by τ in <FIG>, only. <FIG> illustrates, for example, that this point in time τ may be determined by some external clock such as a video frame clock. <FIG>, for instance, illustrates at <NUM> a video composed of a sequence of frames <NUM> in a time-aligned manner with respect to the audio signal <NUM>, one above the other. For instance, the timestamp Tframe could be the timestamp of the first picture of a new scene, new program or the like, and accordingly it could be desired that the audio signal <NUM> is cut at that time τ = Tframe and replaced by another audio signal <NUM> from that time onwards, representing, for instance, the tone signal of the new scene or program. <FIG>, for instance, illustrates an already existing audio data stream <NUM> constructed in the same manner as audio data stream <NUM>, i.e. using access units <NUM> composed of one or more payload packets <NUM> into which the audio signal <NUM> accompanying or describing the sequence of pictures of frames <NUM> starting at timestamp Tframe in audio frames <NUM> in such a manner that the first audio frame <NUM> has its leading end coinciding with time timestamp Tframe, i.e. the audio signal <NUM> is to be played out with the leading end of frame <NUM> registered to the playout of timestamp Tframe.

Disadvantageously, however, the frame rate of frames <NUM> of audio data stream <NUM> is completely independent from the frame rate of video <NUM>. It is accordingly completely random where within a certain frame <NUM> of the audio signal <NUM>τ = Tframe falls into. That is, without any additional measure, it would merely be possible to completely leave off access unit AUj associated with the audio frame <NUM>, j, within which τ lies, and appending at the predecessor access unit AUj-<NUM> of audio data stream <NUM> the sequence of access units <NUM> of audio data stream <NUM>, thereby however causing a mute in the leading end portion <NUM> of audio frame j of audio signal <NUM>.

The various embodiments described hereinafter overcome the deficiency outlined above and enable a handling of such splicing problems.

<FIG> shows an audio data stream in accordance with an embodiment of the present application. The audio data stream of <FIG> is generally indicated using reference sign <NUM>.

Primarily, the construction of the audio signal <NUM> coincides with the one explained above with respect to the audio data stream <NUM>, i.e. the audio data stream <NUM> comprises a sequence of payload packets, namely one or more for each access unit <NUM> into which the data stream <NUM> is partitioned. Each access unit <NUM> is associated with a certain one of the audio frames of the audio signal which is encoded into data stream <NUM> in the units of the audio frames <NUM>. Beyond this, however, the audio data stream <NUM> has been "prepared" for being spliced within an audio frame with which any predetermined access unit is associated. Here, this is exemplarily access unit AUi and access unit AUj. Let us refer to access unit AUi first. In particular, the audio data stream <NUM> is rendered "spliceable" by having a truncation unit packet <NUM> inserted thereinto, the truncation unit packet <NUM> being settable so as to indicate, for access unit AUi, an end portion of the associated audio frame i as to be discarded out in playout. The advantages and effects of the truncation unit packet <NUM> will be discussed hereinafter. Some preliminary notes, however, shall be made with respect to the positioning of the truncation unit packet <NUM> and the content thereof. For example, although <FIG> shows truncation unit packet <NUM> as being positioned within the access unit AUi, i.e. the one the end portion of which truncation unit packet <NUM> indicates, truncation unit packet <NUM> may alternatively be positioned in any access unit preceding access unit AUi. Likewise, even if the truncation unit packet <NUM> is within access unit AUi, access unit <NUM> is not required to be the first packet in the respective access unit AUi as exemplarily illustrated <FIG>.

In accordance with an embodiment which is illustrated in <FIG>, the end portion indicated by truncation unit packet <NUM> is a trailing end portion <NUM>, i.e. a portion of frame <NUM> extending from some time instant tinner within the audio frame <NUM> to the trailing end of frame <NUM>. In other words, in accordance with the embodiment of <FIG>, there is no syntax element signaling whether the end portion indicated by truncation unit packet <NUM> shall be a leading end portion or a trailing end portion. However, the truncation unit packet <NUM> of <FIG> comprises a packet type index <NUM> indicating that the packet <NUM> is a truncation unit packet, and a truncation length element <NUM> indicating a truncation length, i.e. the temporal length Δt of trailing end portion <NUM>. The truncation length <NUM> may measure the length of portion <NUM> in units of individual audio samples, or in n-tuples of consecutive audio samples with n being greater than one and being, for example, smaller than N samples with N being the number of samples in frame <NUM>.

It will be described later that the truncation unit packet <NUM> may optionally comprise one or more flags <NUM> and <NUM>. For example, flag <NUM> could be a splice-out flag indicating that the access unit AUi for which the truncation unit packet <NUM> indicates the end portion <NUM>, is prepared to be used as a splice-out point. Flag <NUM> could be a flag dedicated to the decoder for indicating whether the current access unit AUi has actually been used as a splice-out point or not. However, flags <NUM> and <NUM> are, as just outlined, merely optional. For example, the presence of TU packet <NUM> itself could be a signal to stream splicers and decoders that the access unit to which the truncation unit <NUM> belongs is such a access unit suitable for splice-out, and a setting of truncation length <NUM> to zero could be an indication to the decoder that no truncation is to be performed and no splice-out, accordingly.

The notes above with respect to TU packet <NUM> are valid for any TU packet such as TU packet <NUM>.

As will be described further below, the indication of a leading end portion of an access unit may be needed as well. In that case, a truncation unit packet such as TU packet <NUM>, may be settable so as to indicate a trailing end portion as the one depicted in <FIG>. Such a TU packet <NUM> could be distinguished from leading end portion truncation unit packets such as <NUM> by means of the truncation unit packet's type index <NUM>. In other words, different packet types could be associated with TU packets <NUM> indicating trailing end portions and TU packets being for indicating leading end portions, respectively.

For the sake of completeness, <FIG> illustrates a possibility according to which truncation unit packet <NUM> comprises, in addition to the syntax elements shown in <FIG>, a leading/trailing indicator <NUM> indicating whether the truncation length <NUM> is measured from the leading end or the trailing end of audio frame i towards the inner of audio frame i, i.e. whether the end portion, the length of which is indicated by truncation length <NUM> is a trailing end portion <NUM> or a leading end portion <NUM>. The TU packets' packet type would be the same then.

As will be outlined in more detail below, the truncation unit packet <NUM> renders access unit AUi suitable for a splice-out since it is feasible for stream splicers described further below to set the trailing end portion <NUM> such that from the externally defined splice-out time τ (compare <FIG>) on, the playout of the audio frame i is stopped. From that time on, the audio frames of the spliced-in audio data stream may be played out.

However, <FIG> also illustrates a further truncation unit packet <NUM> as being inserted into the audio data stream <NUM>, this further truncation unit packet <NUM> being settable so as to indicate for access unit AUj, with j > i, that an end portion thereof is to be discarded in playout. This time, however, the access unit AUj, i.e. access unit AUj+<NUM>, has encoded thereinto its associated audio frame j in a manner independent from the immediate predecessor access unit AUj-<NUM>, namely in that no prediction references or internal decoder registers are to be set dependent on the predecessor access unit AUj-<NUM>, or in that no overlap-add process renders a reconstruction of the access unit AUj-<NUM> a requirement for correctly reconstructing and playing-out access unit AUj. In order to distinguish access unit AUj, which is an immediate playout access unit, from the other access units which suffer from the above-outlined access unit interdependencies such as, inter alias, AUi, access unit AUj is highlighted using hatching.

<FIG> illustrates the fact that the other access units shown in <FIG> have their associated audio frame encoded thereinto in a manner so that their reconstruction is dependent on the immediate predecessor access unit in the sense that correct reconstruction and playout of the respective audio frame on the basis of the associated access unit is merely feasible in the case of having access to the immediate predecessor access unit, as illustrated by small arrows <NUM> pointing from predecessor access unit to the respective access unit. In the case of access unit AUj, the arrow pointing from the immediate predecessor access unit, namely AUj-<NUM>, to access unit AUj is crossed-out in order to indicate the immediate-playout capability of access unit AUj. For example, in order to provide for this immediate playout capability, access unit AUj has additional data encoded therein, such as initialization information for initializing internal registers of the decoder, data allowing for an estimation of aliasing cancelation information usually provided by the temporally overlapping portion of the inverse transforms of the immediate predecessor access unit or the like.

The capabilities of access units AUi and AUj are different from each other: access unit AUi is, as outlined below, suitable as a splice-out point owing to the presence of the truncation unit packet <NUM>. In other words, a stream splicer is able to cut the audio data stream <NUM> at access unit AUi so as to append access units from another audio data stream, i.e. a spliced-in audio data stream.

This is feasible at access unit AUj as well, provided that TU packet <NUM> is capable of indicating a trailing end portion <NUM>. Additionally or alternatively, truncation unit packet <NUM> is settable to indicate a leading end portion, and in that case access unit AUj is suitable to serve as a splice-(back-)in occasion. That is, truncation unit packet <NUM> may indicate a leading end portion of audio frame j not to be played out and until that point in time, i.e. until the trailing end of this trailing end portion, the audio signal of the (preliminarily) spliced-in audio data stream may be played-out.

For example, the truncation unit packet <NUM> may have set splice-out flag <NUM> to zero, while the splice-out flag <NUM> of truncation unit packet <NUM> may be set to zero or may be set to <NUM>. Some explicit examples will be described further below such as with respect to <FIG>.

It should be noted that there is no need for the existence of a splice-in capable access unit AUj. For example, the audio data stream to be spliced-in could be intended to replace the play-out of audio data stream <NUM> completely from time instant τ onwards, i.e. with no splice-(back-)in taking place to audio data stream <NUM>. However, if the audio data stream to be spliced-in is to replace the audio data stream's <NUM> audio signal merely preliminarily, then a splice-in back to the audio data stream <NUM> is necessary, and in that case, for any splice-out TU packet <NUM> there should be a splice-in TU packet <NUM> which follows in data stream order <NUM>.

<FIG> shows an audio encoder <NUM> for generating the audio data stream <NUM> of <FIG>. The audio encoder <NUM> comprises an audio encoding core <NUM> and a truncation packet inserter <NUM>. The audio encoding core <NUM> is configured to encode the audio signal <NUM> which enters the audio encoding core <NUM> in units of the audio frames of the audio signal, into the payload packets of the audio data stream <NUM> in a manner having been described above with respect to <FIG>, for example. That is, the audio encoding core <NUM> may be a transform coder encoding the audio signal <NUM> using a lapped transform, for example, such as an MDCT, and then coding the transform coefficients, wherein the windows of the lapped transform may, as described above, cross frame boundaries between consecutive audio frames, thereby leading to an interdependency of immediately consecutive audio frames and their associated access units. Alternatively, the audio encoder core <NUM> may use linear prediction based coding so as to encode the audio signal <NUM> into data stream <NUM>. For example, the audio encoding core <NUM> encodes linear prediction coefficients describing the spectral envelope of the audio signal <NUM> or some pre-filtered version thereof on an at least frame-by-frame basis, with additionally coding the excitation signal. Continuous updates of predictive coding or lapped transform issues concerning the excitation signal coding may lead to the interdependencies between immediately consecutive audio frames and their associated access units. Other coding principles are, however, imaginable as well.

The truncation unit packet inserter <NUM> inserts into the audio data stream <NUM> the truncation unit packets such as <NUM> and <NUM> in <FIG>. As shown in <FIG>, TU packet inserter <NUM> may, to this end, be responsive to a splice position trigger <NUM>. For example, the splice position trigger <NUM> may be informed of scene or program changes or other changes in a video, i.e. within the sequence of frames, and may accordingly signal to the truncation unit packet inserter <NUM> any first frame of such new scene or program. The audio signal <NUM>, for example, continuously represents the audio accompaniment of the video for the case that, for example, none of the individual scenes or programs in the video are replaced by other frame sequences or the like. For example, imagine that a video represents a live soccer game and that the audio signal <NUM> is the tone signal related thereto. Then, splice position trigger <NUM> may be operated manually or automatically so as to identify temporal portions of the soccer game video which are subject to potential replacement by ads, i.e. ad videos, and accordingly, trigger <NUM> would signal beginnings of such portions to TU packet inserter <NUM> so that the latter may, responsive thereto, insert a TU packet <NUM> at such a position, namely relating to the access unit associated with the audio frame within which the first video frame of the potentially to be replaced portion of the video starts, lies. Further, trigger <NUM> informs the TU packet inserter <NUM> on the trailing end of such potentially to be replaced portions, so as to insert a TU packet <NUM> at a respective access unit associated with an audio frame into which the end of such a portion falls. As far as such TU packets <NUM> are concerned, the audio encoding core <NUM> is also responsive to trigger <NUM> so as to differently or exceptionally encode the respective audio frame into such an access unit AUj (compare <FIG>) in a manner allowing immediately playout as described above. In between, i.e. within such potentially to be replaced portions of the video, trigger <NUM> may intermittently insert TU packets <NUM> in order to serve as a splice-in point or splice-out point. In accordance with a concrete example, trigger <NUM> informs, for example, the audio encoder <NUM> of the timestamps of the first or starting frame of such a portion to be potentially replaced, and the timestamp of the last or end frame of such a portion, wherein the encoder <NUM> identifies the audio frames and associated access units with respect to which TU packet insertion and, potentially, immediate playout encoding shall take place by identifying those audio frames into which the timestamps received from trigger <NUM> fall.

In order to illustrate this, reference is made to <FIG> which shows the fixed frame raster at which audio encoding core <NUM> works, namely at <NUM>, along with the fixed frame raster <NUM> of a video to which the audio signal <NUM> belongs. A portion <NUM> out of video <NUM> is indicated using a curly bracket. This portion <NUM> is for example manually determined by an operator or fully or partially automatically by means of scene detection. The first and the last frames <NUM> and <NUM> have associated therewith timestamps Tb and Te, which lie within audio frames i and j of the frame raster <NUM>. Accordingly, these audio frames <NUM>, i.e. i and j, are provided with TU packets by TU packet inserter <NUM>, wherein audio encoding core <NUM> uses immediate playout mode in order to generate the access unit corresponding to audio frame j.

It should be noted that the TU packet inserter <NUM> may be configured to insert the TU packets <NUM> and <NUM> with default values. For example, the truncation length syntax element <NUM> may be set to zero. As far as the splice-in flag <NUM> is concerned, which is optional, same is set by TU packet inserter <NUM> in the manner outlined above with respect to <FIG>, namely indicating splice-out possibility for TU packets <NUM> and for all TU packets <NUM> besides those registered with the final frame or image of video <NUM>. The splice-active flag <NUM> would be set to zero since no splice has been applied so far.

It is noted with respect to the audio encoder of <FIG>, that the way of controlling the insertion of TU packets, i.e. the way of selecting the access units for which insertion is performed, as explained with respect to <FIG> is illustrative only and other ways of determining those access units for which insertion is performed is feasible as well. For example, each access unit, every N-th (N><NUM>) access unit or each IPF access unit could alternatively be provided with a corresponding TU packet.

It has not been explicitly mentioned above, but preferably the TU packets are coded in uncompressed form so that a bit consumption (coding bitrate) of a respective TU packet is independent from the TU packet's actual setting. Having said this, it is further worthwhile to note that the encoder may, optionally, comprise a rate control (not shown in <FIG>), configured to log a fill level of a coded audio buffer so as to get sure that a coded audio buffer at the decoder's side at which the data stream <NUM> is received neither underflows, thereby resulting in stalls, nor overflows thereby resulting in loss of packets <NUM>. The encoder may, for example, control/vary a quantization step size in order to obey the fill level constraint with optimizing some rate/distortion measure. In particular, the rate control may estimate the decoder's coded audio buffer's fill level assuming a predetermined transmission capacity/bitrate which may be constant or quasi constant and, for example, be preset by an external entity such as a transmission network. The coding rate of the TU packets of data stream <NUM> are taken into account by the rate control. Thus, in the form shown in <FIG>, i.e. in the version generated by encoder <NUM>, the data stream <NUM> keeps the preset bitrate with varying, however, therearound in order to compensate for the varying coding complexity if the audio signal <NUM> in terms of its rate/distortion ratio with neither overloading the decoder's coded audio fill level (leading to overflow) nor derating the same (leading to underflow). However, as has already been briefly outlined above, and will be described in more detail below, every splice-out access unit AUi is, accordance to preferred embodiments, supposed to contribute to the playout at decoder side merely for a temporal duration smaller than the temporal length of its audio frame i. As will get clear from the description brought forward below, the (leading) access unit of a spliced-in audio data stream spliced with data stream <NUM> at the respective splice-out AU such as AUi as a splice interface, will displace the respective splice-out AU's successor AUs. Thus, from that time onwards, the bitrate control performed within encoder <NUM> is obsolete. Beyond that, said leading AU is preferably coded in a self-contained manner so as to allow immediate playout, thereby consuming more coded bitrate compared to non-IPF AUs. Thus, in accordance with an embodiment, the encoder <NUM> plans or schedules the rate control such that the logged fill level at the respective splice-out AU's end, i.e. at its border to the immediate successor AU, assumes, for example, a predetermined value such as for example, ¼ or a value between ¾ and <NUM>/<NUM> of the maximum fill level. By this measure, other encoders preparing the audio data streams supposed to be spliced in into data stream <NUM> at the splice-out AUs of data stream <NUM> may rely on the fact that the decoder's coded audio buffer fill level at the time of starting to receive their own AUs (in the following sometimes distinguished from the original ones by an apostrophe) is at the predetermined value so that these other encoders may further develop the rate control accordingly. The description brought forward so far concentrated on splice-out AUs of data stream <NUM>, but the adherence to predetermined estimated/logged fill level is may also be achieved by the rate control for splice-(back)-in AUs such as AUj even if not playing a double role as splice-in and splice-out point. Thus, said other encoders may, likewise, control their rate control in such a manner that the estimated or logged fill level assumes a predetermined fill level at a trailing AU of their data stream's AU sequence. Same may be the same as the one mentioned for encoder <NUM> with respect to splice-out AUs. Such trailing AUs may be supposed to from splice-back AUs supposed to from a splice point with the splice-in AUs of data stream <NUM> such as AUj. Thus, if the encoder's <NUM> rate control has planned/scheduled the coded bit rate such that the estimated/logged fill level assumes the predetermined fill level at (or better after) AUj, then this bit rate control remains even valid in case of splicing having been performed after encoding and outputting data stream <NUM>. The predetermined fill level just-mentioned could be known to encoders by default, i.e. agreed therebetween. Alternatively, the respective AU could by provided with an explicit signaling of that estimated/logged fill level as assumed right after the respective splice-in or splice-out AU. For example, the value could be transmitted in the TU packet of the respective splice-in or splice-out AU. This costs additional side information overhead, but the encoder's rate control could be provided with more freedom in developing the estimated/logged fill level at the splice-in or splice-out AU: for example, it may suffice then that the estimated/logged fill level after the respective splice-in or splice-out AU is below some threshold such as ¾ the maximum fill level, i.e. the maximally guaranteed capacity of the decoder's coded audio buffer.

With respect to data stream <NUM>, this means that same is rate controlled to vary around a predetermined mean bitrate, i.e. it has a mean bitrate. The actual bitrate of the splicable audio data stream varies across the sequence of packets, i.e. temporally. The (current) deviation from the predetermined mean bitrate may be integrated temporally. This integrated deviation assumes, at the splice-in and splice-out access units, a value within a predetermined interval which may be less than ½ wide than a range (max-min) of the integrated bitrate deviation, or may assume a fixed value, e.g. a value equal for all splice-in and splice-out AUs, which may be smaller than ¾ of a maximum of the integrated bitrate deviation. As described above, this value may be pre-set by default. Alternatively, the value is not fixed and not equal for all splice-in and splice-out AUs, but may by signaled in the data stream.

<FIG> shows a stream splicer for splicing audio data streams in accordance with an embodiment. The stream splicer is indicated using reference <NUM> and comprises a first audio input interface <NUM>, a second audio input interface <NUM>, a splice point setter <NUM> and a splice multiplexer <NUM>.

At interface <NUM>, the stream splicer expects to receive a "spliceable" audio data stream, i.e. an audio data stream provided with one or more TU packets. In <FIG> it has been exemplarily illustrated that audio data stream <NUM> of <FIG> enters stream splicer <NUM> at interface <NUM>.

Another audio data stream <NUM> is expected to be received at interface <NUM>. Depending on the implementation of the stream splicer <NUM>, the audio data stream <NUM> entering at interface <NUM> may be a "non-prepared" audio data stream such as the one explained and described with respect to <FIG>, or a prepared one as it will be illustratively set out below.

The splice point setter <NUM> is configured to set the truncation unit packet included in the data stream entering at interface <NUM>, i.e. TU packets <NUM> and <NUM> of data stream <NUM> in the case of <FIG>, and if present the truncation unit packets of the other data stream <NUM> entering at interface <NUM>, wherein two such TU packets are exemplarily shown in <FIG>, namely a TU packet <NUM> in a leading or first access unit AU'<NUM> of audio data stream <NUM>, and a TU packet <NUM> in a last or trailing access unit AU'K of audio data stream <NUM>. In particular, the apostrophe is used in <FIG> in order to distinguish between access units of audio data stream <NUM> from access units of audio data stream <NUM>. Further, in the example outlined with respect to <FIG>, the audio data stream <NUM> is assumed to be pre-encoded and of fixed-length, namely here of K access units, corresponding to K audio frames which together temporally cover a time interval within which the audio signal having been encoded into data stream <NUM> is to be replaced. In <FIG>, it is exemplarily assumed that this time interval to be replaced extends from the audio frame corresponding to access unit AUi to the audio frame corresponding to access unit AUj.

In particular, the splice point setter <NUM> is to, in a manner outlined in more detail below, configured to set the truncation unit packets so that it becomes clear that a truncation actually takes place. For example, while the truncation length <NUM> within the truncation units of the data streams entering interfaces <NUM> and <NUM> may be set to zero, splice point setter <NUM> may change the setting of the transform length <NUM> of the TU packets to a non-zero value. How the value is determined is the subject of the explanation brought forward below.

The splice multiplexer <NUM> is configured to cut the audio data stream <NUM> entering at interface <NUM> at an access unit with a TU packet such as access unit AUi with TU packet <NUM>, so as to obtain a subsequence of payload packets of this audio data stream <NUM>, namely here in <FIG> exemplarily the subsequence of payload packets corresponding to access units preceding and including access unit AUi, and then splicing this subsequence with a sequence of payload packets of the other audio data stream <NUM> entering at interface <NUM> so that same are immediately consecutive with respect to each other and abut each other at the predetermined access unit. For example, splice multiplexer <NUM> cuts audio data stream <NUM> at access unit AUi so as to just include the payload packet belonging to that access unit AUi with then appending the access units AU' of audio data stream <NUM> starting with access unit AU'<NUM> so that access units AUi and AU'<NUM> abut each other. As shown in <FIG>, splice multiplexer <NUM> acts similarly in the case of access unit AUj comprising TU packet <NUM>: this time, splice multiplexer <NUM> appends data stream <NUM>, starting with payload packets belonging to access unit AUj, to the end of audio data stream <NUM> so that access unit AU'K abuts access unit AUj.

Accordingly, the splice point setter <NUM> sets the TU packet <NUM> of access unit AUi so as to indicate that the end portion to be discarded in playout is a trailing end portion since the audio data stream's <NUM> audio signal is to be replaced, preliminarily, by the audio signal encoded into the audio data stream <NUM> from that time onwards. In case of truncation unit <NUM>, the situation is different: here, splice point setter <NUM> sets the TU packet <NUM> so as to indicate that the end portion to be discarded in playout is a leading end portion of the audio frame with which access unit AUj is associated. It should be recalled, however, that the fact that TU packet <NUM> pertains to a trailing end portion while TU packet <NUM> relates to a leading end portion is already derivable from the inbound audio data stream <NUM> by way of using, for example, different TU packet identifiers <NUM> for TU packet <NUM> on the one hand and TU packet <NUM> on the other hand.

The stream splicer <NUM> outputs the spliced audio data stream thus obtained an output interface <NUM>, wherein the spliced audio data stream is indicated using reference sign <NUM>.

It should be noted that the order in which splice multiplexer <NUM> and splice point setter <NUM> operate on the access units does not need to be as depicted in <FIG>. That is, although <FIG> suggests that splice multiplexer <NUM> has its input connected to interfaces <NUM> and <NUM>, respectively, with the output thereof being connected to output interface <NUM> via splice point setter <NUM>, the order among splice multiplexer <NUM> and splice point setter <NUM> may be switched.

In operation, the stream splicer <NUM> may be configured to inspect the splice-in syntax element <NUM> comprised by truncation unit packets <NUM> and <NUM> within audio data stream <NUM> so as to perform the cutting and splicing operation on the condition of whether or not the splice-in syntax element indicates the respective truncation unit packet as relating to a splice-in access unit. This means the following: the splice process illustrated so far and outlined in more detail below may have been triggered by TU packet <NUM>, the splice-in flag <NUM> is set to one, as described with respect to <FIG>. Accordingly, the setting of this flag to one is detected by stream splicer <NUM>, whereupon the splice-in operation described in more detail below, but already outlined above, is performed.

As outlined above, splice point setter <NUM> may not need to change any settings within the truncation unit packets as far as the discrimination between splice-in TU packets such as TU packet <NUM> and the splice-out TU packets such as TU packets <NUM> is concerned. However, the splice point setter <NUM> sets the temporal length of the respective end portion to be discarded in playout. To this end, the splice point setter <NUM> may be configured to set a temporal length of the end portion to which the TU packets <NUM>, <NUM>, <NUM> and <NUM> refer, in accordance with an external clock. This external clock <NUM> stems, for example, from a video frame clock. For example, imagine the audio signal encoded into audio data stream <NUM> represents a tone signal accompanying a video and that this video is video <NUM> of <FIG>. Imagine further that frame <NUM> is encountered, i.e. the frame starting a temporal portion <NUM> into which an ad is to be inserted. Splice point setter <NUM> may have already detected that the corresponding access unit AUi comprises the TU packet <NUM>, but the external clock <NUM> informs splice point setter <NUM> on the exact time Tb at which the original tone signal of this video shall end and be replaced by the audio signal encoded into data stream <NUM>. For example, this splice-point time instant may be the time instant corresponding to the first picture or frame to be replaced by the ad video which in turn is accompanied by a tone signal encoded into data stream <NUM>.

In order to illustrate the mode of operation of the stream splicer <NUM> of <FIG> in more detail, reference is made to <FIG>, which shows the sequence of steps performed by stream splicer <NUM>. The process starts with a weighting loop <NUM>. That is, stream splicer <NUM>, such as splice multiplexer <NUM> and/or splice point setter <NUM>, checks audio data stream <NUM> for a splice-in point, i.e. for an access unit which a truncation unit packet <NUM> belongs to. In the case of <FIG>, access unit i is the first access unit passing check <NUM> with yes, until then check <NUM> loops back to itself. As soon as the splice-in point access unit AUi has been detected, the TU packet thereof, i.e. <NUM>, is set so as to register the splice-in point access unit's trailing end portion (its leading end thereof) with the time instant derived from the external clock <NUM>. After this setting <NUM> by splice point setter <NUM>, the splice multiplexer <NUM> switches to the other data stream, i.e. audio data stream <NUM>, so that after the current splice-in access unit AUi, the access units of data stream <NUM> are put to output interface <NUM>, rather than the subsequent access units of audio data stream <NUM>. Assuming that the audio signal which is to replace the audio signal of audio data stream <NUM> from the splice-in time instant onward, is coded into audio data stream <NUM> in a manner so that this audio signal is registered with, i.e. starts right away, with the beginning of the first audio frame which is associated with a first access unit AU'<NUM>, the stream splicer <NUM> merely adapts the timestamp information comprised by audio data stream <NUM> so that a timestamp of the leading frame associated with a first access unit AU'<NUM>, for example, coincides with the splice-in time instant, i.e. the time instant of AUi plus the temporal length of the audio frame associated with AUi minus the temporal length of the trailing end portion as set in step <NUM>. That is, after multiplexer switching <NUM>, the adaptation <NUM> is a task continuously performed for the access unit AU' of data stream <NUM>. However, during this time the splice-out routine described next is performed as well.

In particular, the splice-out routine performed by stream splicer <NUM> starts with a waiting loops according to which the access units of the audio data stream <NUM> are continuously checked for same being provided with a TU packet <NUM> or for being the last access unit of audio data stream <NUM>. This check <NUM> is continuously performed for the sequence of access units AU'. As soon as the splice-out access unit has been encountered, namely AU'K in the case of <FIG>, then splice point setter <NUM> sets the TU packet <NUM> of this splice-out access unit so as to register the trailing end portion to be discarded in playout, the audio frame corresponding to this access unit AUK with a time instant obtained from the external clock such as a timestamp of a video frame, namely the first after the ad which the tone signal coded into audio data stream <NUM> belongs to. After this setting <NUM>, the splice multiplexer <NUM> switches from its input at which data stream <NUM> is inbound, to its other input. In particular, the switching <NUM> is performed in a manner so that in the spliced audio data stream <NUM>, access unit AUj immediately follows access unit AU'K. In particular, the access unit AUj is the access unit of data stream <NUM>, the audio frame of which is temporally distanced from the audio frame associated with the splice-in access unit AUi by a temporal amount which corresponds to the temporal length of the audio signal encoded into data stream <NUM> or deviates therefrom by less than a predetermined amount such as a length or half a length of the audio frames of the access units of audio data stream <NUM>.

Thereinafter, splice point setter <NUM> sets in step <NUM> the TU packet <NUM> of access unit AUj to register the leading end portion thereof to be discarded in playout, with the time instant with which the trailing end portion of the audio frame of access unit AU'K had been registered in step <NUM>. By this measure, the timestamp of the audio frame of access unit AUj equals the timestamp of the audio frame of access unit AU'K plus a temporal length of the audio frame of access unit AU'K minus the sum of the trailing end portion of audio frame of access unit AU'K and the leading end portion of the audio frame of access unit AUj. This fact will become clearer looking at the examples provided further below.

This splice-in routine is also started after the switching <NUM>. Similar to ping-pong, the stream splicer <NUM> switches between the continuous audio data stream <NUM> on the one hand and audio data streams of predetermined length so as to replace predetermined portions, namely those between access units with TU packets on the one hand and TU packets <NUM> on the other hand, and back again to audio stream <NUM>.

Switching from interface <NUM> to <NUM> is performed by the splice-in routine, while the splice-out routine leads from interface <NUM> to <NUM>.

It is emphasized, however, again that the example provided with respect to <FIG> has merely been chosen for illustration purposes. That is, the stream splicer <NUM> of <FIG> is not restricted to "bridge" portions to be replaced from one audio data stream <NUM> by audio data streams <NUM> having encoded thereinto audio signals of appropriate length with the first access unit having the first audio frame encoded thereinto registered to the beginning of the audio signal to be inserted into the temporal portion to be replaced. Rather, the stream splicer may be, for instance, for performing a one-time splice process only. Moreover, audio data stream <NUM> is not restricted to have its first audio frame registered with the beginning of the audio signal to be spliced-in. Rather, the audio data stream <NUM> itself may stem from some source having its own audio frame clock which runs independently from the audio frame clock underlying audio data stream <NUM>. In that case, switching from audio data stream <NUM> to audio data stream <NUM> would, in addition to the steps shown in <FIG>, also comprise the setting step corresponding to step <NUM>: the setting of the TU packet of the audio data stream <NUM>.

It should be noted that the above description of the stream splicer's operation may be varied with respect to the timestamp of AUs of the spliced audio data stream <NUM> for which a TU packet indicates a leading end portion to be discarded in playout. Instead of leaving the AU's original timestamp, the stream multiplexer <NUM> could be configured to modify the original timestamp thereof by adding the leading end portion's temporal length to the original timestamp thereby pointing to the trailing end of the leading end portion and thus, to the time from which on the AU's audio frame fragment is be actually played out. This alternative is illustrated by the timestamp examples in <FIG> discussed later.

<FIG> shows an audio decoder <NUM> in accordance with an embodiment of the present application. Exemplarily, the audio decoder <NUM> is shown as receiving the spliced audio data stream <NUM> generated by stream splicer <NUM>. However, similar to the statement made with respect to the stream splicer, the audio decoder <NUM> of <FIG> is not restricted to receive spliced audio data streams <NUM> of the sort explained with respect to <FIG>, where one base audio data stream is preliminarily replaced by other audio data streams having the corresponding audio signal length encoded thereinto.

The audio decoder <NUM> comprises an audio decoder core <NUM> which receives the spliced audio data stream and an audio truncator <NUM>. The audio decoding core <NUM> performs the reconstruction of the audio signal in units of audio frames of the audio signal from the sequence of payload packets of the inbound audio data stream <NUM>, wherein, as explained above, the payload packets are individually associated with a respective one of the sequence of access units into which the spliced audio data stream <NUM> is partitioned. As each access unit <NUM> is associated with a respective one of the audio frames, the audio decoding core <NUM> outputs the reconstructed audio samples per audio frame and associated access unit, respectively. As described above, the decoding may involve an inverse spectral transformation and owing to an overlap/add process or, optionally, predictive coding concepts, the audio decoding core <NUM> may reconstruct the audio frame from a respective access unit while additionally using, i.e. depending on, a predecessor access unit. However, whenever an immediate playout access unit arrives, such as access unit AUj, the audio decoding core <NUM> is able to use additional data in order to allow for an immediate playout without needing or expecting any data from a previous access unit. Further, as explained above, the audio decoding core <NUM> may operate using linear predictive decoding. That is, the audio decoding core <NUM> may use linear prediction coefficients contained in the respective access unit in order to form a synthesis filter and may decode an excitation signal from the access unit involving, for instance, transform decoding, i.e. inverse transforming, table lookups using indices contained in the respective access unit and/or predictive coding or internal state updates with then subjecting the excitation signal thus obtained to the synthesis filter or, alternatively, shaping the excitation signal in the spectral domain using a transfer function formed so as to correspond to the transfer function of the synthesis filter. The audio truncator <NUM> is responsive to the truncation unit packets inserted into the audio data stream <NUM> and truncates an audio frame associated with a certain access unit having such TU packets so as to discard the end portion thereof, which is indicated to be discarded in playout of the TU packet.

<FIG> shows a mode of operation of the audio decoder <NUM> of <FIG>. Upon detecting <NUM> a new access unit, the audio decoder checks whether or not this access unit is one coded using immediate playout mode. If the current access unit is an immediate playout frame access unit, the audio decoding core <NUM> treats this access unit as a self-contained source of information for reconstructing the audio frame associated with this current access unit. That is, as explained above the audio decoding core <NUM> may pre-fill internal registers for reconstructing the audio frame associated with a current access unit on the basis of the data coded into this access unit. Additionally or alternatively, the audio decoding core <NUM> refrains from using prediction from any predecessor access unit as in the non-IPF mode. Additionally or alternatively, the audio decoding core <NUM> does not perform any overlap-add process with any predecessor access unit or its associated predecessor audio frame for the sake of aliasing cancelation at the temporally leading end of the audio frame of the current access unit. Rather, for example, the audio decoding core <NUM> derives temporal aliasing cancelation information from the current access unit itself. Thus, if the check <NUM> reveals that the current access unit is an IPF access unit, then the IPF decoding mode <NUM> is performed by the audio decoding core <NUM>, thereby obtaining the reconstruction of the current audio frame. Alternatively, if check <NUM> reveals that the current access unit is not an IPF one, then the audio decoding core <NUM> applies as usual non-IPF decoding mode onto the current access unit. That is, internal registers of the audio decoding core <NUM> may be adopted as they are after processing the previous access unit. Alternatively or additionally, an overlap-add process may be used so as to assist in reconstructing the temporally trailing end of the audio frame of the current access unit. Alternatively or additionally, prediction from the predecessor access unit may be used. The non-IPF decoding <NUM> also ends-up in a reconstruction of the audio frame of the current access unit. A next check <NUM> checks whether any truncation is to be performed. Check <NUM> is performed by audio truncator <NUM>. In particular, audio truncator <NUM> checks whether the current access unit has a TU packet and whether the TU packet indicates an end portion to be discarded in playout. For example, the audio truncator <NUM> checks whether a TU packet is contained in the data stream for the current access unit and whether the splice active flag <NUM> is set and/or whether truncation length <NUM> is unequal to zero. If no truncation takes place, the reconstructed audio frame as reconstructed from any of steps <NUM> or <NUM> is played out completely in step <NUM>. However, if truncation is to be performed, audio truncator <NUM> performs the truncation and merely the remaining part is played out in step <NUM>. In the case of the end portion indicated by the TU packet being a trailing end portion, the remainder of the reconstructed audio frame is played out starting with the timestamp associated with that audio frame. In case of the end portion indicated to be discarded in playout by the TU packet being a leading end portion, the remainder of the audio frame is played-out at the timestamp of this audio frame plus the temporal length of the leading end portion. That is, the playout of the remainder of the current audio frame is deferred by the temporal length of the leading end portion. The process is then further prosecuted with the next access unit.

See the example in <FIG>: the audio decoding core <NUM> performs normal non-IPF decoding <NUM> onto access units AUi-<NUM> and AUi. However, the latter has TU packet <NUM>. This TU packet <NUM> indicates a trailing end portion to be discarded in playout, and accordingly the audio truncator <NUM> prevents a trailing end <NUM> of the audio frame <NUM> associated with access unit AUi from being played out, i.e. from participating in forming the output audio signal <NUM>. Thereinafter, access unit AU'<NUM> arrives. Same is an immediate playout frame access unit and is treated by audio decoding core <NUM> in step <NUM> accordingly. It should be noted that audio decoding core <NUM> may, for instance, comprise the ability to open more than one instantiation of itself. That is, whenever an IPF decoding is performed, this involves the opening of a further instantiation of the audio decoding core <NUM>. In any case, as access unit AU'<NUM> is an IPF access unit, it does not matter that its audio signal is actually related to a completely new audio scene compared to its predecessors AUi-<NUM> and AUi. The audio decoding core <NUM> does not care about that. Rather, it takes access unit AU'<NUM> as a self-contained access unit and reconstructs the audio frame therefrom. As the length of the trailing end portion of the audio frame of the predecessor access unit AUi has probably been set by the stream splicer <NUM>, the beginning of the audio frame of access unit AU'<NUM> immediately abuts the trailing end of the remainder of the audio frame of access unit AUi. That is, they abut at the transition time T<NUM> somewhere in the middle of the audio frame of access unit AUi. Upon encountering access unit AU'K, the audio decoding core <NUM> decodes this access unit in step <NUM> in order to reveal or reconstruct this audio frame, whereupon this audio frame is truncated at its trailing end owing to the indication of the trailing end portion by its TU packet <NUM>. Thus, merely the remainder of the audio frame of access unit AU'K up to the trailing end portion is played-out. Then, access unit AUj is decoded by audio decoding core <NUM> in the IPF decoding <NUM>, i.e. independently from access unit AU'K in a self-contained manner and the audio frame obtained therefrom is truncated at its leading end as its truncation unit packet <NUM> indicates a leading end portion. The remainders of the audio frames of access units AU'K and AUj abut each other at a transition time instant T<NUM>.

The embodiments described above basically use a signaling that describes if and how many audio samples of a certain audio frame should be discarded after decoding the associated access unit. The embodiments described above may for instance be applied to extend an audio codec such as MPEG-H 3D Audio. The MEPG-H 3D Audio standard defines a self-contained stream format to transform MPEG-H 3D audio data called MHAS [<NUM>]. In line with the embodiments described above, the truncation data of the truncation unit packets described above could be signaled at the MHAS level. There, it can be easily detected and can be easily modified on the fly by stream splicing devices such as the stream splicer <NUM> of <FIG>. Such a new MHAS packet type could be tagged with PACTYP_CUTRUNCATION, for example. The payload of this packet type could have the syntax shown in <FIG>. In order to ease the concordance between the specific syntax example of <FIG> and the description brought forward above with respect to <FIG>, for example, the reference signs of <FIG> have been reused in order to identify corresponding syntax elements in <FIG>. The semantics could be as follows:.

Note that the MHAS stream guarantees that a MHAS packet payload is always byte-aligned so the truncation information is easily accessible on the fly and can be easily inserted, removed or modified by e.g. a stream splicing device. A MPEG-H 3D Audio stream could contain a MHAS packet type with pactype PACTYP_CUTRUNCATION for every AU or for a suitable subset of AUs with isActive set to <NUM>. Then a stream splicing device can modify this MHAS packet according to its need. Otherwise a stream splicing device can easily insert such a MHAS packet without adding significant bitrate overhead as it is described hereinafter. The largest granule size of MPEG-H 3D Audio is <NUM> samples, so <NUM> bits for nTruncSamples are sufficient to signal all meaningful truncation values. nTruncSamples and the <NUM> one bit flags together occupy <NUM> bits or <NUM> bytes so that no further byte alignment is needed.

<FIG> illustrate how the method of CU truncation can be used to implement sample accurate stream splicing.

<FIG> shows a video stream and an audio stream. At video frame number <NUM> the program is switched to a different source. The alignment of video and audio in the new source is different than in the old source. To enable sample accurate switching of the decoded audio PCM samples at the end of the last CU of the old stream and at the beginning of the new stream have to be removed. A short period of cross-fading in the decoded PCM domain may be required to avoid glitches in the output PCM signal. <FIG> shows an example with concrete values. If for some reason the overlap of AUs/CUs is not desired, the two possible solutions depicted in <FIG>) exist. The first AU of the new stream has to carry the configuration data for the new stream and all pre-roll that is needed to initialize the decoder with the new configuration. This can be done by means of an Immediate Playout Frame (IPF) that is defined in the MPEG-H 3D Audio standard.

Another application of the CU truncation method is changing the configuration of a MPEG-H 3D Audio stream. Different MPEG-H 3D Audio streams may have very different configurations. a stereo program may be followed by a program with <NUM> channels and additional audio objects. The configuration will usually change at a video frame boundary that is not aligned with the granules of the audio stream. The method of CU truncation can be used to implement sample accurate audio configuration change as illustrated in <FIG>.

<FIG> shows a video stream and an audio stream. At video frame number <NUM> the program is switched to a different configuration. The first CU with the new audio configuration is aligned with the video frame at which the configuration change occurred. To enable sample accurate configuration change audio PCM samples at the end of the last CU with the old configuration have to be removed. The first AU with the new configuration has to carry the new configuration data and all pre-roll that is needed to initialize the decoder with the new configuration. This can be done by means of an Immediate Playout Frame (IPF) that is defined in the MPEG-H 3D Audio standard. An encoder may use PCM audio samples from the old configuration to encode pre-roll for the new configuration for channels that are present in both configurations. Example: If the configuration change is from stereo to <NUM>, then the left and right channels of the new <NUM> configuration can use pre-roll data form left and right from the old stereo configuration. The other channels of the new <NUM> configuration use zeros for pre-roll. <FIG> illustrates encoder operation and bitstream generation for this example.

<FIG> shows further examples for spliceable or spliced audio data streams. See <FIG>, for example. <FIG> shows a portion out of a spliceable audio data stream exemplarily comprising seven consecutive access units AU<NUM> to AU<NUM>. The second and sixth access units are provided with a TU packet, respectively. Both are not used, i.e. non-active, by setting flag <NUM> to zero. The TU packet of access unit AU<NUM> is comprised by an access unit of the IPF type, i.e. it enables a splice back into the data stream. At B, <FIG> shows the audio data stream of A after insertion of an ad. The ad is coded into a data stream of access units AU'<NUM> to AU'<NUM>. At C and D, <FIG> shows a modified case compared to A and B. In particular, here the audio encoder of the audio data stream of access units AU<NUM>. , has decided to change the coding settings somewhere within the audio frame of access unit AU<NUM>. Accordingly, the original audio data stream of C already comprises two access units of timestamp <NUM>, namely AU<NUM> and AU'<NUM> with respective trailing end portion and leading end portion indicated as to be discarded in playout, respectively. Here, the truncation activation is already preset by the audio decoder. Nevertheless, the AU'<NUM> access unit is still usable as a splice-back-in access unit, and this possibility is illustrated in D.

An example of changing the coding settings at the splice-out point is illustrated in E and F. Finally, at G and H the example of A and B in <FIG> is extended by way of another TU packet provided access unit AU<NUM>, which may serve as a splice-in or continue point.

As has been mentioned above, although the pre-provision of the access units of an audio data stream with TU packets may be favorable in terms of the ability to take the bitrate consumption of these TU packets into account at a very early stage in access unit generation, this is not mandatory. For example, the stream splicer explained above with respect to <FIG> may be modified in that the stream splicer identifies splice-in or splice-out points by other means than the occurrence of a TU packet in the inbound audio data stream at the first interface <NUM>. For example, the stream splicer could react to the external clock <NUM> also with respect to the detection of splice-in and splice-out points. According to this alternative, the splice point setter <NUM> would not only set the TU packet but also insert them into the data stream. However, please note that the audio encoder is not freed from any preparation task: the audio encoder would still have to choose the IPF coding mode for access units which shall serve as splice-back-in points.

Finally, <FIG> shows that the favorable splice technique may also be used within an audio encoder which is able to change between different coding configurations. The audio encoder <NUM> in <FIG> is constructed in the same manner as the one of <FIG>, but this time the audio encoder <NUM> is responsive to a configuration change trigger <NUM>. That is, see for example case C in <FIG>: the audio encoding core <NUM> continuously encodes the audio signal <NUM> into access units AU<NUM> to AU<NUM>. Somewhere within the audio frame of access unit AU<NUM>, the configuration change time instant is indicated by trigger <NUM>. Accordingly, audio encoding core <NUM>, using the same audio frame raster, also encodes the current audio frame of access unit AU<NUM> using a new configuration such as an audio coding mode involving more coded audio channels or the like. The audio encoding core <NUM> encodes the audio frame the other time using the new configuration with additionally using the IPF coding mode. This ends up into access unit AU'<NUM>, which immediately follows an access unit order. Both access units, i.e. access unit AU<NUM> and access unit AU'<NUM> are provided with TU packets by TU packet inserter <NUM>, the former one having a trailing end portion indicated so as to be discarded in playout and the latter one having a leading end portion indicated as to be discarded in playout. The latter one may, as it is an IPF access unit, also serve as a splice-back-in point. For all of the above-described embodiments it should be noted that, possibly, cross-fading is performed at the decoder between the audio signal reconstructed from the subsequence of AUs of the spliced audio data stream up to a splice-out AU (such as AUi), which is actually supposed to terminate at the leading end of the trailing end portion of the audio frame of this splice-out AU on the one hand and the audio signal reconstructed from the subsequence of AUs of the spliced audio data stream from the AU immediately succedding the splice-out AU (such as AU'<NUM>) which may be supposed to start rightaway from the leading end of audio frame of the successor AU, or at the trailing end of the leading end portion of the audio frame of this successor AU: That is, within a temporal interval surrounding and crossing the timestant where the portions of the immediately consecutive AUs, to be played-out abut each other, the actually played-out audio signal as played out from the spliced audio data stream by the decoder could be formed by a combination of the audio frames of both immediately abutting AUs with a combinational contribution of the audio frame of the successor AU temporally increasing within this temporal interval and the combinational contribution of the audio frame of the splice-out AU temporally decreasing in the temporal interval. Similarly, cross fading could be performed between splice-in AUs such as AUj and their immediate predecessor AUs (such as AU'K), namely by forming the acutally played out audio signal by a combination of the audio frame of the splice-in AU and the audio frame of the predecessor AU within a time interval surrounding and crossing the time instant at which the leading end portion of the splice-in AU's audio frame and the trailnng end portion of the predecessor AU's audio frame abut each other.

Using another wording, above embodiments, inter alias revealed, a possibility to exploit bandwidth available by the transport stream, and available decoder MHz: a kind of Audio Splice Point Message is sent along with the audio frame it would replace. Both the outgoing audio and the incoming audio around the splice point are decoded and a crossfade between them may be performed. The Audio Splice Point Message merely tells the decoders where to do the crossfade. This is in essence a "perfect" splice because the splice occurs correctly registered in the PCM domain.

Claim 1:
Audio decoder comprising:
an audio decoding core (<NUM>) configured to reconstruct an audio signal (<NUM>), in units of audio frames (<NUM>) of the audio signal, from a sequence of payload packets (<NUM>) of an audio data stream (<NUM>), wherein each of the payload packets belongs to a respective one of a sequence of access units (<NUM>) into which the audio data stream is partitioned, wherein each access unit is associated with a respective one of the audio frames; and
an audio truncator (<NUM>) configured to be responsive to a truncation unit packet (<NUM>; <NUM>; <NUM>) which is inserted, within a predetermined access unit, into the audio data stream, and comprises
a packet type index (<NUM>) indicating that the truncation unit packet is a truncation unit packet,
a truncation length element (<NUM>) indicating a temporal length of an end portion of an audio frame associated with the predetermined access unit in units of individual audio samples, or in n-tuples of consecutive audio samples, and
a flag (<NUM>) indicating whether the predetermined access unit has actually been used as a splice-out point or not; and
a leading/trailing indicator (<NUM>) indicating whether the temporal length is measured from a leading end or a trailing end of the audio frame towards an inner of the audio frame
so as to check whether the flag is set and if so, truncate the audio frame so as to discard, in playing out the audio signal, the end portion indicated to be discarded in playout by the truncation unit packet, and if not, play out the audio frame completely.