The present invention relates to a method and apparatus for conversion of audio
file format and, more specifically to a better manipulation of audio data streams
in a file format where the audio data associated to a time mark can be
distributed among different data blocks, such as is the case in MP3 format.
MPEG audio compression is a particularly effective way to store audio signals,
such as music or the sound for a film, in digital form while requiring, on the one
hand, as little memory space as possible and, on the other hand, maintaining the
audio quality as good as possible. Over the last years, MPEG audio compression
has proved to be one of the most successful solutions in this filed.
Meanwhile, different versions of MPEG audio compression methods exist.
Generally, the audio signal is sampled with a certain sample rate, the resulting
sequence of audio samples being associated to overlapping time periods or time
marks, respectively. These time marks are then individually supplied to, for
example, a hybrid filter bank consisting of polyphase and a modified discrete
cosine transform (MDCT)/ suppressing aliasing effects. The actual data
compression takes place during quantization of the MDCT coefficients. The MDCT
coefficients quantized in that way are then converted into a Huffman code of
Huffman code words generating a further compression of associating shorter
code words to more frequently occurring coefficients. Thus, overall, the MPEG
compressions are lossy. The "audible" losses, however, being limited, since
psychoacoustic knowledge has been incorporated in the way of quantizing the
DCT coefficients.
A widely used MPEG standards is the so-called MP3 standard, as described in
IOS/IEC 11172-3 and 13818-3. This standard
allows an adaptation of the information loss generated by
compression to the bit rate by which the audio information
is to be transmitted in real time. The transmission of the
compressed data signal in a channel with constant bit rate
should also be performed in other MPEG standards. In order
to ensure that the listening quality at the receiving
decoder remains sufficient, even at low bit rates, the MP3
standard provides for an MP3 coder having a so-called bit
reservoir. This means the following. Normally, due to the
fixed bit rate, the MP3 coder should code every time mark
into a block of code words having the same size, this block
could then be transmitted with given bit rate in the time
period of the time period repetition rate. However, this
would not accommodate the case that some parts of an audio
signal, such as the sounds following a very loud sound in a
piece of music, require less exact quantization with
constant quality compared to other parts of the audio
signal, such as parts with a plurality of different
instruments. Thus, an MP3 coder does not generate a simple
bit stream format where every time mark is coded in one
frame with the same frame length for all frames. Such a
self-contained frame would consist of a frame header, side
information and main data associated to the time mark
associated to the frame, namely the coded MDCT
coefficients, wherein the side information is information
for the decoder how the DCT coefficients are to be decoded,
such as how many subsequent DCT coefficients are 0, for
indicating which DCT coefficients are successively included
in the main data. Rather, a backpointer is included in the
side information or in the header, pointing to a position
within the main data in one of the previous frames. This
position is the beginning of the main data pertaining to
the time mark to which the frame is associated wherein the
corresponding backpointer is included. The backpointer
indicates, for example, the number of bites by which the
beginning of the main data is offset in the bit stream. The
end of these main data can be in any frame, depending on
how high the compression rate for this time mark is. The
length of the main data of the individual time marks is
thus no longer constant. Thus, the number of bits by which
a_block is coded can be adapted^ to the properties_~og^the
signal. At the same time, a constant bit rate can be
achieved. This technique is called "bit reservoir".
Generally, the bit reservoir is a buffer of bits, which can
be used to provide more bits for coding a block of time
samples than would generally be allowed by the constant
output data rate. The technique of bit reservoir
accommodates the fact that some blocks of audio samples can
be coded with less bits than specified by the constant
transmission rate, so that these blocks fill the bit
reservoir, while other blocks of audio samples have
psychoacoustic properties that do not allow such a high
compression, so that the available bits would actually not
be sufficient for low-interference or interference-free
decoding, respectively, of these blocks. The required
excessive bits are taken from the bit reservoir, so that
the bit reservoir empties during such blocks. The technique
of the bit reservoir is also described in the above-
indicated standard MPEG layer 3.
Although the MP3 format does have advantages on the coder
side by providing the backpointers, there are undeniable
disadvantages on the decoder side. If, for example, a
decoder receives an MP3 bit stream not from the beginning
but starting from a certain frame in the middle, the coded
audio signal at the time mark associated to this frame can
only be played instantly when the backpointer is
incidentally 0, which would indicate that the beginning of
the main data to this frame is incidentally immediately
after the header or side information, respectively.
However, this is normally not the case. Thus, playing the
audio signal at this time mark is not possible when the
backpointer of the frame that was received first points to
a previous frame, which, however, has not (yet) been
received. In that case, (at first) only the next frame can
be played.
Further problems occur on the receiver side when dealing
with the frames in general, which are interconnected by the
backpointers and are thus not self-contained. A further
problem of bit streams with return addresses for a bit
reservoir is that, when different channels of an audio
"signal are individually MP3 coded, main data pertaining to
each other in the two bit streams since they are associated
to the same time mark, might be offset to each other, and
with variable offset across the sequence of frames, so that
here again combining these individual MP3 streams into a
multi-channel audio data stream is impeded.
Additionally, there is a need for a simple possibility for
generating easily manageable MP3-compliant multi-channel
audio data streams. Multi-channel MP3 audio data streams
according to ISO/IEC standard 13818-3 require matrix
operations for retrieving the input channels from the
transmitted channels on the decoder side and the usage of
several backpointers and are thus complicated to
manipulate.
MPEG 1/2 layer 2 audio data streams correspond to the MP3
audio data streams in their composition of subsequent
frames and in the structure and arrangement of the frames,
namely the structure of header, side information and main
data part, and the arrangement with a quasi statical frame
distance depending on the sample rate and the bit rate
variable from frame to frame, however, they differ from the
same by the lack of backpointers or bit reservoir,
respectively, during coding. Coding-expensive and
inexpensive time periods of the audio signal are coded with
the same frame length. The main data pertaining to a time
mark are in the respective frame together with the
respective header.
It is the object of the present invention to provide a
scheme for converting an audio data stream into a further
audio data stream or vice versa, so that the manipulation
with the audio data is made easier, such as with regard to
combining individual audio data streams into multi-channel
audio data streams or the manipulation of an audio data
stream in general.
The manipulation of audio data can be simplified, such as,
for example, with regard to the combination of individual
audio data streams into multi-channel audio data streams or
the general manipulation of an audio data stream, by
modifying a data block in an audio data stream divided into
data blocks with determination block and data block data,
such as by completing or adding or replacing part of the
same, so that the same includes a length indicator
indicating an amount or length of data, respectively, of
the data block audio data or an amount or length of data,
respectively, of the data block, to obtain a second audio
data stream with modified data blocks. Alternatively, an
audio data stream with pointers in determination blocks,
which point to determination block audio data associated to
those determination blocks, but distributed among different
data blocks, is converted into an audio data stream,
wherein the determination block audio data are combined to
contiguous determination block audio data. The contiguous
determination block audio data can then be included in a
self-contained channel element together with their
determination block.
It is a finding of the present invention that a pointer-
based audio data stream where a pointer points to the
beginning of the determination block audio data of the
respective data block is easier to handle when this audio
data stream is manipulated so that all determination block
audio data, i.e. audio data concerning the same time mark
or coding the audio values for the same audio mark, are
combined into a contiguous block of contiguous
determination block audio data, and that the respective
determination block, to which the contiguous determination
block audio data are associated, is added to the same.
After arranging or lining-up the same, respectively, the
channel elements obtained that way result in the new audio
data stream wherein all audio data pertaining to one time
mark or coding the audio values or samples, respectively,
for this time mark, are also combined in one channel
element, so that the new audio data stream is easier to
handle.
According to an embodiment of the present invention, every
determination block or every channel element is modified in
the new audio data stream, such as by adding or replacing a
part to obtain a length indication indicating the length or
amount of data, respectively, of the channel element of the
contiguous audio data included therein, to ease decoding
the new audio data stream with channel elements of variable
length. Advantageously, modification is performed by
replacing a redundant part of these determination blocks
identical for all determination blocks of the input audio
data stream by the respective length indication. This
measure can achieve that the data bit rate of the resulting
audio data stream is equal to the one of the original audio
data stream despite the additional length indication
compared to the original pointer-based audio data stream,
and that thereby further the actually unnecessary
backpointer in the new audio data stream can be obtained in
order to be able to reconstruct the original audio data
stream from the new audio data stream.
The identical redundant part of these determination blocks
can be placed before the new resulting audio data stream in
an overall determination block. On the receiver side, the
resulting second audio data stream can thus be reconverted
into the original audio data stream in order to use
existing decoders that can only decode audio data streams
of the original file format for decoding the resulting
audio data stream in the pointer-less format.
According to a further embodiment of the present invention,
a conversion of a first audio data stream into a second
"aTrdxu-data- stream of another file format is used' to" form a
rriuTti-channel audio" data stream of several audio data
Streams of the first file format. A receiver-side
manageability is improved compared to the mere combination
of the original audio data streams with pointer, since in
the multi-channel audio data stream all channel elements
pertaining to a time mark or containing the contiguous
determination block audio data, respectively, were obtained
by coding a simultaneous time period of a channel of a
multi-channel audio signal, i.e. by coding time periods of
different channels pertaining to the time mark, can be
combined to access units. This is not possible with
pointer-based audio data formats, since there the audio
data for one time mark can be distributed among different
data blocks. Providing data blocks in several audio data
streams to different channels with a length indication
allows better parsing by the access units during
combination of the audio data streams to a multi-channel
data stream with access units.
Further, the present invention resulted from the finding
that it is very easy to reconvert the above-described
resulting audio data streams into an original file format,
which can then be decoded into the audio signal by existing
decoders. While the resulting channel elements have a
different length and are thus sometimes longer and
sometimes shorter than the length available in the data
block of the original audio data stream, it is not required
to offset or combine the main data according to the
eventually unnecessarily obtained backpointers for playing
the audio data stream in a new file format, but it is
sufficient to increase a bit rate indication in the
determination blocks of the audio data stream of the
original file format to be generated. The effect of this is
that according to this bit rate indication, even the
longest of the channel elements in the audio data stream to
be" decoded is smaller or the same as the data block length
which the data blocks have in an audio data stream of the
first file format. The backpointers are set to zero and the
channel elements are increased to the length corresponding
to the increased bit rate indication by adding bits of
don't care values. Thus, data blocks of an audio data
stream in original file format are generated, wherein the
pertaining main data are merely included in the data block
itself and not in any other one. An audio data stream of
the first file format reconverted in that way can then be
supplied to an existing decoder for audio data streams of
the first file format by using the bit rate increased
according to the increased bit indication. Thus, expensive
shift operations for reconverting are omitted, as well as
the requirement to replace existing decoders by new ones.
On the other hand, according to a further embodiment, it is
possible to retrieve the original audio data stream from
the resulting audio data stream by using the information
"included in the overall determination block of the
resulting audio data stream across the identical redundant
part of the determination blocks to retrieve the part
overwritten by the length indication.
Preferred embodiments of the present invention will be
discussed below with reference to the A accompanying
drawings. They show:
Fig. 1 a schematical drawing for illustrating the MP3
file format with backpointer;
Fig. 2 a block diagram for illustrating a structure for
converting an MP3 audio data stream into an MPEG-
4 audio data stream;
Fig. 3 a flow diagram of a method for converting an MP3
audio data stream into an MPEG-4 audio data
stream according to an embodiment of the present
invention;
Fig. 4 a schematical drawing for illustrating the step
of combining associated audio data by adding the
determination blocks and the step of modifying
the determination blocks in the method of Fig. 3;
Fig. 5 a schematical drawing for illustrating a method
for converting several MP3 audio data streams
into a multi-channel MPEG-4 audio data stream
according to a further embodiment of the present
invention;
Fig. 6 a block diagram of an arrangement for converting
an MPEG-4 audio data stream obtained according to
Fig. 3 back to an MP3 audio data stream for being
able to decode the same by existing MP3 decoders;
Fig. 7 a flow diagram of a method for reconverting the
MPEG-4 audio data stream obtained according to
Fig. 3 into one or several audio data streams in
MP3 format;
Fig. 8 a flow diagram of a method for reconverting the
MPEG-4 audio data stream obtained according to
Fig. 3 into one or several audio data streams in
MP3 format according to a further embodiment of
the present invention; and
Fig. 9 a flow diagram of a method for converting an MP3
audio data stream into an MPEG-4 audio data
stream according to a further embodiment of the
present invention.
The present invention will be discussed below with
reference to the drawings based on embodiments where the
original audio data stream in a file format where
backpointers are used in the determination blocks of the
data blocks for pointing to the beginning of main data
pertaining to the determination block is merely exemplarily
an MP3 audio data stream, while the resulting audio data
stream consisting of self-contained channel elements where
the audio data pertaining to the respective time mark are
each combined, is also merely exemplarily an MPEG-4 audio
data stream. The MP3 format is described in the standard
ISO/IEC 11172-3 and 13818-3 cited in the background period,
while the MPEG-4 file format is described in standard
ISO/IEC 14496-3.
First, the MP3 format will be briefly discussed with
reference to Fig. 1. Fig. 1 shows a portion of an MP3 audio
data stream 10. The audio data stream 10 consists of a
sequence of frames or data blocks, respectively, of which
only three can be fully seen in Fig. 1, namely 10a, 10b and
10c. The MP3 audio data stream 10 has been generated by an
MP3 coder from an audio or sound signal, respectively. The
audio signal coded by the data stream 10 is, for example,
music, noise, a mixture of the same and the like. The data
blocks 10a, 10b and 10c are each associated to one of
successive, possibly overlapping time periods into which
the audio signal has been divided by the MP3 coder. Every
time period corresponds to a time mark of the audio signal,
and thus, in the description, the term time mark is often
used for the time period. Every time period has been
encoded into main data (main_data) by the MP3 coder
individually by, for example, a hybrid filter bank
consisting of a polyphase filter bank and a modified
discrete cosine transform with subsequent entropy, such as
Huffman, coding. The main data pertaining to the successive
three time marks, to which the data blocks lOa-lOc are
associated, are illustrated in Fig. 1 by 12a, 12b and 12c
as contiguous blocks aside from the actual audio data
stream 10.
The data blocks lOa-lOc of the audio data stream 10 are
equidistantly arranged in the audio data stream 10. This
means that every data block lOa-lOc has the same data block
length or frame length, respectively. The frame length,
again, depends on the bit rate at which the audio data
stream 10 is to be at least played in real time, and on the
sample rate which the MP3 coder has used for sampling the
audio signal prior to the actual coding. The connection is
that the sample rate indicates in connection with the fixed
number of samples per time mark how long a time mark is,
and that it can be calculated from the bit rate and the
time mark period how many bits can be transmitted in this
time period.
Both parameters, i.e. bit rate and sample rate, are
indicated in frame headers 14 in the data blocks lOa-lOc.
Thus, every data block lOa-lOc has its own frame header 14.
Generally, all information important for decoding the audio
data stream are stored in every frame lOa-lOc itself, so
that a decoder can begin decoding in the middle of an MP3
audio data stream 10.
Apart from the frame header 14, which is at the beginning,
every data block lOa-lOc has a side information part 16 and
a main data part 18 containing data block audio data. The
side information part 16 immediately follows the header 14.
The same includes information essential for the decoder of
the audio data stream 10 for finding the main data or
determination block audio data, respectively, associated to
the respective data block, which are merely Huffman code
words disposed linearly in series and to decode the same in
a correct way to the DCT or MDCT coefficients,
respectively. The main data part 18 forms the end of every
data block.
As mentioned in the background section of the description,
the MP3 standard supports a reservoir function. This is
enabled by backpointers included in the side information
within the side information part 16 indicated in Fig. 1 by
20. If a backpointer is set to 0, the main data for these
side information begin immediately after the side
information part 16. Otherwise, the pointer 20
(main_data_begin) indicates the beginning of the main data
coding the time mark to which the data block is associated,
wherein the side information 16 containing the backpointer
20 is included in a previous data block. In Fig. 1, for
example, the data block 10a is associated to a time mark
coded by the main data 12a. The backpointer 20 in the side
information 16 of this data block 10a points, for example,
to the beginning of the main data 12a, which is in a data
block prior to the data block 10a in stream direction 22 by
indicating a bit or byte offset measured from the beginning
of the header 14 of the data block 16a. This means that at
this time during coding of the audio signal, the bit
reservoir of the MP3 coder generating the MP3 audio data
stream 10 has not been full but could be loaded up to the
height of the backpointer. From the position, to which the
backpointer 20 of the data block 10a points, onwards, the
main data 12a are inserted in the audio data stream 10 with
equidistantly disposed pairs of headers and side
information 14, 16. In the present example, the main data
12a extend up to slightly over half of the main data part
18 of the data block 10a. The backpointer 20 in the side
information part 16 of the subsequent 10b points to a
position immediately after the main data 12a in the data
block 10a. The same applies to the backpointer 20 in the
side information part 16 of the data block 10c.
As can be seen, it is rather an exception in the MP3 audio
data stream 10 when the main data pertaining to a time mark
are actually exclusively in a data block associated to this
time mark. Rather, the data blocks are mostly distributed
among one or several data blocks, which might not even
include the corresponding data block itself, depending on
the size of the bit reservoir. The height of the
backpointer value is limited by the size of the bit
reservoir.
After the structure of an MP3 audio data stream has been
described with regard to Fig. 1, an arrangement will be
described with reference to Fig. 2, which is suitable to
convert an MP3 audio data stream into an MPEG-4 audio data
stream, or to obtain an MPEG-4 audio data stream from an
audio signal, which can easily be converted into an MP3
format.
Fig. 2 shows an MP3 coder 30 and an MP3-MPEG-4 converter
32. The MP3 coder 30 comprises an input where the same
receives an audio signal to be coded, and an output where
the same outputs an MP3 audio data stream coding the audio
signal at the input. The MP3 coder 30 operates according to
the above-mentioned MP3 standard.
The MP3 audio data stream whose structure has been
discussed with reference to Fig. 1 consists, as mentioned,
of frames with a fixed frame length, which depends on a set
bit rate and the underlying sample rate as well as a
padding byte, which is set or not set. The MP3-MPEG-4
converter 32 receives the MP3 audio data stream at an input
an outputs an MPEG-4 audio data stream at an output, the
structure of which results from the subsequent description
of the mode of operation* of the MP3-MPEG-4 converter 32.
The purpose of the converter 32 is to convert the MP3 audio
data stream from the MP3 format into the MPEG-4 format. The
MPEG-4 data format has the advantage that all main data
pertaining to a certain time mark are included in a
contiguous access unit or channel element, so that
manipulating the latter is eased significantly.
Fig. 3 shows the individual method steps during conversion
of the MP3 audio data stream into the MPEG-4 audio data
stream performed by the converter 32. First, the MP 3 audio
data stream is received in a step 40. Receiving can
comprise storing the full audio data stream or merely a
current part of the same in a latch. Correspondingly, the
subsequent steps during conversion can either be performed
during receiving 40 in real time or only following that.
Then, in a step 42, all audio data or main data,
respectively, pertaining to a time mark are combined in a
contiguous block, and this is performed for all time marks.
Step 42 is illustrated in more detail schematically in Fig.
4, wherein in this figure the elements of an MP3 audio data
stream similar to the elements illustrated in Fig. 1, are
provided with the same or similar reference numbers and a
repeated description of these elements is omitted.
As can be seen from the data stream direction 22, these
parts of the MP3 audio data stream 10 illustrated farther
to the left in Fig. 4 reach the converter 32 earlier than
the right parts of the same. Two data blocks 10a and 10b
are illustrated fully in Fig. 4. The time mark pertaining
to the data block 10a is coded by the main data MDl
included in Fig. 4 exemplarily partly in a data block prior
to the data block 10 and partly in the data block 10a, and
here particularly in the main data part 18 of the same.
Those main data coding the time mark to which the
subsequent data block 10b is associated, are exclusively
included in the main data part 18 of the data block 10a and
indicated by MD2. The main data MD3 pertaining to the data
block following the data block 10b are distributed among
the main data parts 18 of the data blocks 10a and 10b.
In step 42, the converter 42 combines all pertaining main
data, i.e. all main data coding one and the same time mark,
into contiguous blocks. In that way, the portion 44 prior
to the data block 10a of the portion 46 in the data block
10a in the main data MDl result in the contiguous block 48
by combining after step 42. The same is performed for the
other main data MD2, MD3 ....
For performing step 42, the converter 32 reads the pointer
in the side information 16 of a data block 10a and then,
based on this pointer, the respective first part 44 of the
determination block, audio data 12a for this data block 10a
included in the field 18 of a previous data block,
beginning at the position determined by the pointer up to
the header of the current data block 10a. Then he reads the
second part 46 of the determination block audio data
included in part 18 of the current data block 10a and
comprising the end of the determination block audio data
for this data block 10a beginning from the end of the side
information 16 of the current audio data block 10a to the
beginning of the next audio data, here indicated by MD2, to
the next data block 10b, to which the pointer in the side
information 16 of the subsequent data block 10b points,
which the converter 32 reads as well. Combining the two
parts 44 and 46 results, as described, in block 48.
In a step 50, the converter 32 adds the associated header
14 including the associated side information 16 to the
contiguous blocks to finally form MP3 channel elements 52a,
52b and 52c. Thus, every MP3 channel element 52a-52c
consists of the header 14 of a corresponding MP3 data
block, a subsequent side information part 16 of the same
MP3 data block, and the contiguous block 48 of main data
coding the time mark to which the data block is associated
from which header and side information originate.
The MP3 channel elements resulting from steps 42 and 50
have different channel element lengths, as indicated by
double arrows 54a-54c. It should be noted that the data
blocks 10a, 10b in the MP3 audio data stream 10 had a fixed
frame length 56, but that the number of main data for the
individual time marks varies around an average value due to
the bit reservoir function.
For easing decoding and particularly parsing of the
individual MP3 channel elements 52a-52c on the decoder
side, the headers 14 H1-H3 are modified to obtain the
length of the respective channel element 52a-52c, i.e. 54a-
54c. This is performed in a step 56. The length input is
written into a part identical or redundant, respectively,
for all headers 14 of the audio data stream 10. In the MP3
format, every header 14 receives in the beginning a fixed
synchronizations word (syncword) consisting of 12 bits. In
step 56, this syncword is occupied by the length of the
respective channel element. The 12 bits of the syncword are
sufficient to represent the length of the respective
channel element in binary form, so that the length of the
resulting MP3 channel elements 58a-58c with modified header
hl-h3 remains the same despite step 56, i.e. equal to 54a-
54c. In that way, the audio information can also be
transmitted with the same bit rate in real time or be
played like the original MP3 audio data stream 10 after
combining the MP3 channel elements 58a-58c according to the
order of the time mark coded by the same despite adding the
length indication, as long as no further overhead is added
by additional headers.
In a step 58, a file header, or for the case that the data
stream to be generated is not a file but streaming, a data
stream header is generated for the desired MPEG-4 audio
data stream (step 60). Since, according to the present
embodiment, an MPEG-4-compliant audio data stream is to be
generated, a file header is generated according to MPEG-4
standard, wherein in that case the file header has a fixed
structure due to the function AudioSpecificConfig, which is
defined in the above-mentioned MPEG-4 standard. The
interface to the MPEG-4 system is provided by the element
ObjectTypelndication set with the value 0x40, as well as by
the indication of an audioObjectType with the number 29.
The MPEG-4-specific AudioSpecificConfig is extended as
follows corresponding to its original definition in ISO/IEC
14496-3, wherein in the following example only the contents
of the AudioSpecificConfig significant for the present
description and not all of them are considered:
1 AudioSpecificConfig() {
2 audioObjectType;
3 samplingFrequencylndex;
4 if(samplingFrequencyIndex==Oxf)
5 samplingFrequency;
6 channelConfiguration;
7 if(audioObjectType==29){
8 MPEG_l_2_SpecificConfig();
9 }
10 }
The above list of the AudioSpecif icConfig is a
representation in common notation for the function
AudioSpecificConfig, which serves for parsing or reading
the call parameters in the file header in the decoder,
namely the samplingFrequencylndex, the
channelConfiguration, and the audioObjectType, or indicates
the instructions how the file header is to be decoded or to
be parsed.
As can be seen, the file header generated in step 60 begins
with the indication of the audioObjectType, which is set to
29 (line 2) as mentioned above. The parameter
audioObjectType indicates to the decoder in what way the
data have been coded, and particularly in what way further
information for coding the file header can be extracted, as
will be described below.
Then, the call parameter samplingFrequencylndex follows,
which points to a certain position in a normed table for
sample frequencies (line 3) . If the index is 0 (line 4),
the indication of the sample frequency follows without
pointing to a normed table (line 5).
Then, the indication of a channel configuration follows
(line 6), which indicates in a way that will be discussed
below in more detail, how many channels are included in the
generated MPEG-4 audio data stream, where it is also
possible, in contrast to the present embodiment, to combine
more than one MP3 audio data stream to one MPEG-4 audio
data stream, as will be described below with reference to
Fig. 5.
Then, if the audioObjectType is 29, which is the case here,
a part in the file header AudioSpecificConfig, containing a
redundant part of the MP3 frame header in the audio data
stream 10 follows, i.e. that part remaining the same among
the frame headers 14 (line 8). This part is here indicated
by MPEG_l_2_SpecificConfig() , again a function defining the
structure of this part.
Although the structure of MPEG_l_2_SpecificConfig can also
be taken from the MP3 standard, since it corresponds to the
fixed part of an MP3 frame header that does not change from
frame to frame, the structure of the same is listed below
exemplarily:
1 MPEG_1 2_SpecificConfig(channelConfiguration){
2 syncword
3 ID
4 layer
5 reserved
6 sampling_frequency
7 reserved
8 reserved
9 reserved
10 if(channelConfiguration==0){
11 channel configuration description;
12 }
13 }
In the part MPEG_l_2_SpecificConfig all bits differing from
frame header to frame header 14 in the MN3 audio data
stream are set to 0. In any case, the first parameter
MPEG_l_2_SpecificConfig, namely the 12-bit-synchronization
word syncword serving for synchronization of an MP3 coder
when receiving an MP3 audio data stream (line 2), is the
same for every frame header. The subsequent parameter ID
(line 3) indicates the MPEG version, i.e. 1 or 2, by the
corresponding standard ISO/IEC 13818-3 for version 2 and
the standard ISO/IEC 11172-3 for version 1. The parameter
layer (line 4) gives an indication to layer 3, which
corresponds to the MP3 standard. The following bit is
reserved (line 5), since its value can change from frame to
frame and is transmitted by the MP3 channel elements. This
bit shows possibly that the header is followed by a CRC
variable. The next variable sampling_frequency (line 6)
points to a table with sample rates defined in MP3 standard
and thus indicates the sample rate underlying the MP3-DCT
coefficients. Then, in line 7, the indication of a bit for
specific applications (reserved) follows, as well as in
lines 8 and 9. Then, (in lines 11, 12) the exact definition
of the channel configuration follows when the parameter
indicated in line 6 of the AudioSpecif icConf ig does not
point to a predefined channel configuration but has the
value 0. Otherwise, the channel configuration of 14496-3
subpart 1 table 1.11 applies.
By step 60 and in particularly by providing the element
MPEG_l_2_SpecificConfig in the file header, which includes
all redundant information in the frame headers 14 of the
original MP3 audio data stream 10, it is ensured that this
redundant part in the frame headers does not lead to
irretrievable loss of this information in the MPEG-4 file
to be generated during the insertion of data easing
decoding, such as in step 56 by inserting the channel
element length, but that this modified part can be
reconstructed based on the MPEG-4 file header.
Then, in step 62, the MPEG-4 audio data stream is output in
the order of the MPEG-4 file header generated in step 60
and the channel elements in the order of their associated
time marks, wherein the full MPEG-4 audio data stream
results in an MPEG-4 file or is transmitted by MPEG-4
systems.
The above description related to the conversion of an MP3
audio data stream into an MPEG-4 audio data stream.
However, as can be seen with dotted lines in Fig. 2, it is
also possible to convert two or more MP3 audio data streams
from two MP3 coders, namely 30 and 30' into an MPEG-4
multi-channel audio data stream. In that case, the MP3-
MPEG-4 converter 32 receives the MP3 audio data stream of
all coders 30 and 30' and outputs the multi-channel audio
data stream in MPEG-4 format.
In the upper half. Fig. 5 illustrates in relation to the
representation of Fig. 4 in what way the multi-channel
audio data stream according to MPEG-4 can be obtained,
wherein the conversion is again performed by the converter
32. Three channel element sequences 70, 72 and 74 are
illustrated, which have been generated according to steps
40-56 from the one audio signal each by an MP3 coder 30 or
30' (Fig. 2) . From every sequence of channel elements 70,
72 and 74, two respective channel elements are shown,
namely 70a, 70b, 72a, 72b or 74a, 74b, respectively. In
Fig. 5, the channel elements disposed above one another,
here 70a-74a or 70b-74b, respectively, are each associated
to the same time mark. The channel elements of sequence 70,
for example, code the audio signal that has been recorded
according to a suitable normation on the front left, right
(front), while the sequences 72 and 82 code audio signals
representing a recording of the same audio source from
other directions or with another frequency spectrum, such
as the central front loudspeaker (center) and from the back
right and left (surround).
As indicated by arrows 7 6, these channel elements are now
combined to units during the output (cf. step 62 in Fig. 3)
in the MPEG-4 audio data stream, referred to below as
access units 78. Thus, in the MPEG-4 audio data stream, the
data within an access unit 78 always relate to a time mark.
The arrangement of MP3 channel elements 70a, 72a and 74a
within the access unit 78, here in the order front, center
and surround channel, is considered in the file header as
generated for the MPEG-4 audio data stream to be generated
(cf. step 60 in Fig. 3) by respectively setting the call
parameter channel configuration in the AudioSpecificConfig,
reference again being made to subpart 1 in ISO/IEC 14496-3.
The access units 78 are again successively arranged in the
MPEG-4 stream according to the order of their time marks,
and they are preceded by the MPEG-4 file header. The
parameter channelConfiguration is set appropriately in the
MPEG-4 file header to indicate the order of channel
elements in the access units or their significance on
decoder side, respectively.
As the above description of Fig. 5 has shown, it is very
easy to combine MP3 audio data streams into a multi-channel
audio data stream when, as proposed according to the
present invention, the MP3 audio data streams are
manipulated to obtain self-contained channel elements from
the data blocks, wherein all data for one time mark are
included in one channel element, wherein these channel
elements of the individual channels can then easily be
combined into access units.
The present description related to the conversion of one or
several MP3 audio data streams into an MPEG-4 audio data
stream. However, it is a significant finding of the present
invention that all the advantages of the resulting MPEG-4
audio data stream, such as improved manageability of the
individual self-contained MP3 channel elements with equal
transmission rate and the possibility of multi-channel
transmission can be utilized without having to replace
existing MP3 coders fully by new decoders, but that the
reconversion can also be performed unproblematically, so
that the same can be used during decoding the above-
described MPEG-4 audio data stream.
In Fig. 6, this is illustrated in an arrangement of an MP3
reconstructor 100 whose mode of operation will be discussed
in more detail below, and of MP3 decoders 102, 102' .... An
MP3 reconstrutor receives at its input an MPEG-4 audio data
stream as generated according to one of the previous
embodiments, and outputs one or, in the case of a multi-
channel audio data stream, several MP3 audio data streams
to one or several MP3 decoders 102, 102' ..., which
themselves decode the respectively received MP3 audio data
stream to a respective audio signal and pass it on to
respective loudspeakers disposed according to the channel
configuration.
A particularly simple way of reconstructing the original
MP3 audio data streams of an MPEG-4 audio data stream
generated according to Fig. 5, will be described with
reference to Fig. 5 below and Fig. 7, wherein these steps
are performed by the MP3 reconstructor of Fig. 6.
First, the MP3 reconstructor 100 verifies in a step 110
that the MPEG-4 audio data stream received at the input is
a reformatted MP3 audio data stream, by checking the call
parameter audioObjectType in the file header according to
the AudioSpecificConfig whether the same includes the value
29. If this is the case (line 7 in the
AudioSpecificConfig) , the MP3 reconstructor 100 proceeds
with parsing the file header of the MPEG-4 audio data
stream and reads the redundant part of all frame headers of
the original MP3 audio data stream from part-
MPEG_l_2_SpecificConfig from which the MPEG-4 audio data
stream has been obtained (step 112).
After evaluating the MPEG_l_2_SpecificConfig, the MP3
reconstructor 100 replaces In the step 114 In every channel
element 74a-74c In the respective header hF, he, h3 one or
several parts of the channel elements by components of the
MPEG_l_2_SpecificConfig, particularly the channel element
length indication by the synchronization word from
MPEG_l_2_SpecificConfig to obtain the original MP3 audio
data stream frame headers HF, Hc and Hs again, as indicated
by arrows 116. In a step 118, the MP3 reconstructor 100
modifies the side information Sf, Sc and Sa in the MPEG-4
audio data stream in every channel element. Particularly,
the backpointer is set to 0 to obtain new side information
S'f, S'c and S's. The manipulation according to step 118 is
indicated in Fig. 5 by arrows 120. Then, in a step 122, the
MP3 reconstructor 100 sets the bit rate index in every
channel element 74a-74c in the frame header HF, Hc, H3
provided in step 114 with the synchronization word instead
of the channel element length indication to the highest
allowable value. In the end, the resulting headers differ
from the original ones, which is indicated in Fig. 5 by an
apostrophe, i.e. H'F, H'c and H's. The manipulation of the
channel elements according to step 122 is also indicated by
arrow 116.
For illustrating the changes of steps 114-122 again,
individual parameters are listed in Fig. 5 for the header
H'F and the side index part S'F. In 124, individual
parameters of the header H'F are indicated. The frame
header H'r begins with the parameter syncword. Syncword is
set to the original value (step 114) as it is the case in
every MP3 audio data stream, namely to the value OxFFF.
Generally, a frame header H'F as resulting after steps 114-
122 differs from the original MP3 frame header as included
in the original MP3 audio data stream 10 only by the fact
that the bit rate index is set to the highest allowable
value, which is OxE according to MP3 standard.
The purpose of changing the bit rate index is to obtain a
new frame length or data block length, respectively, for
the newly to be generated MP3 audio data stream, which is
greater than the one of the original MP3 audio data stream,
from which the MPEG-4 audio data stream with access unit 78
has been generated. The trick hereby is that the frame
length in bytes in MP3 format always depends on the bit
rate, according to the following equation:
for MPEG 1 layer 3:
frame length[Bit]=1152*bit rate[Bit/s]/sample rate[Bit/s] +
+ 8*paddingbit[Bit]
for MPEG 2 layer 3:
frame length[Bit]=576*bit rate[Bit/s]/sample rate[Bit/s] +
+ 8*paddingbit[Bit]
In other words, the frame length of an MP3 audio data
stream according to the standard is directly proportional
to the bit rate and indirectly proportional to the sample
rate. As additional value, the value of the padding bits is
added, which is indicated in the MP3 frame headers hF, hc,
hs and can be used to set the bit rate exactly. The sample
rate is fixed, since it determines with what speed the
decoded audio signal is played. The conversion of the bit
rate compared to the original setting allows to accommodate
such MP3 channel elements 74-74c in a data block length of
the newly to be generated MP3 audio data stream, which are
longer than the original, since for generating the original
audio data stream the main data have been generated by
taking bits from the bit reservoir.
Thus, while in the present embodiment the bit rate index is
always set to the highest allowable value, it would further
be possible to increase the bit rate index only to a value
sufficient to result in a data block length according to
the MP3 standard, so that even the longest MP3 channel
elements 74a-74c would fit from their length.
At 126, it is illustrated that the backpointer
main_data_begin is set to 0 in the resulting side
information. This only means that in the MP3 audio data
stream generated according to the method of Fig. 7 the data
blocks are always self-contained, so that the main data for
a certain frame header and the side information always
begin directly after the side information and end within
the same data block.
Steps 114, 118, 122 are performed at every channel element,
by extracting each of the same from their access units,
wherein the channel element length indications are useful
during extraction.
Then, in a step 128, that amount of fill data or don't care
bits are added to every channel element 74a-74c to increase
the length of all MP3 channel elements unitarily to the MP3
data block length as set by the new bit rate index OxE.
These fill data are indicated at 128 in Fig. 5. The amount
of fill data can be calculated for every channel element,
for example, by evaluating the channel element length
indication and the padding bit.
Then, in a step 130, the channel elements shown in Fig. 5
at 74a'-74c' modified according to the previous steps, are
passed on to a respective MP3 decoder or an MP3 decoder
entity 134a-134c as data blocks of an MP3 audio data stream
in the order of the coded time marks. The MPEG-4 file
header is omitted. The resulting MP3 audio data streams are
indicated in Fig. 5 generally by 132a, 132b and 132c. The
MP3 decoder entities 134a-134c have, for example, been
initialized before, the same number as channel elements are
included in the individual access units.
The MP3 reconstructor 100 knows which channel elements 74a-
74c in an access unit 78 of the MPEG-4 audio data stream
pertain to which of the to-be-generated MP3 audio data
streams 132a-132c from an evaluation of the call parameter
channelConfiguration in the AudioSpecificConfig of the
MPEG-4 audio data stream. Thus, the MP3 decoder entity 134a
connected to the front loudspeaker receives the audio data
stream 132a corresponding to the front channel, and
correspondingly the MP3 decoder entities 134b and 134c
receive the audio data streams 132b and 132c associated to
the center and surround channel and output the resulting
audio signals to respectively disposed loudspeakers for
example to a subwoofer or to loudspeakers disposed at the
back left and back right, respectively.
Of course, for real-time coding of the MPEG-4 audio data
stream by the arrangement of Fig. 6 with the decoder
entities 102, 102' or 134a-134c it is required to transmit
the newly generated MP3 audio data streams 132a-132c with
the bit rate increased in step 122, which is higher than in
the original audio data stream 10, which is, however, no
problem since the arrangement between MP3 reconstructor 100
and the MP3 decoders 102, 102' or 134a-134c is fixed, so
that here the transmission paths are correspondingly short
and can be designed with correspondingly high data rate
with low cost and effort.
According to the embodiment described with reference to
Fig. 7, an MPEG-4 multi-channel audio data stream obtained
according to Fig. 5 from original audio data streams 10 has
not been reconverted exactly to the original MP3 audio data
streams, but other MP3 audio data streams have been
generated from the same, wherein in contrast to the
original audio data streams, all backpointers are set to 0
and the bit rate index is set to the highest value. The
data blocks of these newly generated MP3 audio data streams
are thus also self-contained insofar as all data associated
to a certain time mark are included in the same data block
74'a-74'c, and fill data have been used to increase the
data block length to a unitary value.
Fig. 8 shows an embodiment for a method according to which
it is possible to reconvert an MPEG-4 audio data stream
generated according to the embodiments of Figs. 1-5 into
the original MP3 audio streams or the original MP3 audio
data stream, respectively.
In that case, the MP3 reconstructor 100 tests again in a
step 150 exactly as in step 110 whether the MPEG-4 audio
data stream is a reformatted MP3 audio data stream. The
subsequent steps 152 and 154 also correspond to steps 112
and 114 of the procedure of Fig. 7.
Instead of changing the backpointers in the side
information and the bit rate index in the frame headers,
the MP3 reconstructor 100 reconstructs, according to the
method of Fig. 8, in step 156 the original data block
length in the original MP3 audio data streams converted to
the MPEG-4 audio data stream, based on the sample rate, the
bit rate and the padding bit. The sample rate and the
padding indication are indicated in the
MPEG_l_2_SpecificConfig, and the bit rate in every channel
element, if the latter is different from frame to frame.
The equation for calculating the original frame length of
the original and to-be-reconstructed audio data stream is
again as above mentioned :
for MPEG 1 layer 3:
frame length[Bit]=1152*bit rate[Bit/s]/sample rate[Bit/s] +
+ 8*paddingbit[Bit]
for MPEG 2 layer 3:
frame length[Bit]=576*bit rate[Bit/s]/sample rate[Bit/s] +
+ 8*paddingbit[Bit]
Then, the MP3 audio data stream or the MP3 audio data
streams, respectively, are generated by arranging the
respective frame headers from the respective channel in an
interval of the calculated data block length and the gaps
are filled up by inserting the audio date or main data,
respectively, at the positions indicated by the pointers in
the side information. Different from the embodiments of
Fig. 7 or 5, respectively, the main data associated to the
respective header or the respective side information,
respectively, are inserted into the MP3 audio data stream
at the beginning of the position indicated by the
backpointer. Or, in other words, the beginning of the
dynamic main data is offset corresponding to the value of
main_data_begin. The MPEG-4 file header is omitted. The
resulting MP3 audio data stream or the resulting MP3 audio
data streams, respectively, correspond to the original MP3
audio data streams on which the MPEG-4 audio data stream
was based. These MP3 audio data streams could thus be
decoded by conventional MP3 decoders into audio signals,
like the audio data streams of Fig. 7.
With regard to the previous description, it should be noted
that the MP3 audio data streams described as single-channel
MP3 audio data streams had at some positions actually
already been two-channel MP3 audio data streams defined
according to ISO/IEC standard 13818-3, wherein, however,
the description did not go into detail about that since it
does not change anything with regard to the understanding
of the present invention. Matrix operations from the
transmitted channels for retrieving the input channel on
decoder side and the usage of several backpointers in these
multi-channel signals have not been discussed, but
reference is made to the respective standard.
The above embodiments made it possible to store MP3 data
blocks in altered form in MPEG-4 file format. MPEG-1/2-
audio-layer-3, short MP3 or proprietary formats like
MPEG2.5 or mp3PR0 derived therefrom can be packed into an
MPEG-4 file based on these procedures, so that this new
representation represents a multi-channel representation of
an arbitrary number of channels in a simple way. Using the
complicated and hardly used method from the standard
ISO/IEC 13818-3 is not required. Particularly, the MP3 data
blocks are packed such that every block - channel element
of access unit - pertains to a defined time mark.
In the above embodiments for changing the format of the
digital signal representation, parts of the representation
have been overwritten with different data. In other words,
information required or useful for the decoder are written
across the part of the MP3 data block that is constant for
different blocks within a data stream.
By packing several mono or stereo data blocks into an
access unit of the MPEG-4 file format, a multi-channel
representation could be obtained, which is significantly
easier to handle compared to the representation from
standard ISO/IEC 13818-3.
In the previous embodiments, the representation of an MP3
data block has been formatted in such a different way that
all data pertaining to a certain time mark are also
included within one access unit. This is generally not the
case in MP3 data blocks, since the element main_data_begin
or the backpointerin the original MP3 data block,
respectively, can point to earlier data blocks.
The reconstruction of the original data stream could also
be performed (Fig. 8) . This means, as shown, that the
retrieved data streams can be processed by every conforming
decoder.
Above that, the above embodiments allow coding or decoding
7>f more than" two trtiaime-l's". Further, in' the" ab'bve
embodiments, tfie ready-coded MP3 data only have to be
reformatted by simple operations to obtain a multi-channel
format. On the other hand, on the coder side, only this
operation or these operations, respectively, had to be
reversed.
While an MP3 data stream usually Includes data blocks of
differing lengths, since the dynamic data pertaining to one
block can be packed into previous blocks, the previous
embodiments bundled the dynamic data directly behind the
side information. The resulting MPEG-4 audio data stream
had a constant medium bit rate, but data blocks of
differing lengths. The element main_data_begin or the
backpointer, respectively, is transmitted in an unaltered
way to ensure reproduction of the original data stream.
Further, with reference to Fig. 5, an extension of the
MPEG-4 syntax has been described to pack several MP3 data
blocks as MP3 channel elements to one multi-channel format
within an MPEG-4 file. All MP3 channel element entries
pertaining to one point of time were packed in one access
unit. Corresponding to the MPEG-4 standard, the suitable
information for configuration on the coder side can be
taken from the so-called AudioSpecificConfig. Apart from
the audioObjectType, the sample rate and channel
configuration etc., the same includes a descriptor relevant
for the respective audioObjectType. This descriptor has
been described above with regard to the
MPEG_l_2_SpecificConfig.
According to the previous embodiments, the 12-bit MPEG-1/2
syncword in the header has been replaced by the length of
the respective MP3 channel element. According to ISO/IEC
13818-3, 12 bits are sufficient therefore. The remaining
header has not been modified any further, which can,
however, happen for shortening, for example, the frame
header and the residual redundant part except the syncword
to reduce the amount of information to be transmitted.
Different variations of the above embodiments can easily be
carried out. Thus, the sequence in the steps in Figs. 3, 7,
8 can be altered, particularly steps 42, 50, 56, 60 in Fig.
3, 11, 114, 118, 122 and 128 in Fig. 7, and 152, 154, 156
in Fig. 8.
Further, with regard to Figs. 3, 7, 8 it should be noted
that the steps shown there are performed by respective
features in the converter or reconstructor, respectively,
of Figs. 2 or 6, respectively, which can, for example, be
embodied as a computer or a hard-wired circuit.
In the embodiment of Fig. 7, the manipulation of the
headers of the side information, respectively, (steps 118,
122) has been performed for the MP3 decoders on receiver or
decoder side, respectively, on the MP3 data stream slightly
changed compared to the original MP3 data stream. In many
application cases, it can be advantageous to perform these
steps on coder or transmitter side, respectively, since the
receiver devices are often mass-produced devices, so that
savings in electronics on the receiver side allow
significantly higher gains. According to an alternative
embodiment, it can thus be provided that these steps are
already performed during MP3-MPEG-4 data format conversion.
The steps according to this alternative format conversion
method are shown in Fig. 9, wherein steps identical to the
ones in Fig. 3 are provided with the same reference numbers
and are not described again to avoid repetitions.
First, the MP3 audio data stream to be converted is
received in step 40, and in step 42 the audio data
pertaining to a time mark or representing a coding of a
time period of the audio signal to be coded by the MP3
audio data stream pertaining to the respective time mark,
respectively, are combined into a contiguous block, and
this for all time marks. The headers are added again to the
contiguous blocks to obtain the channel elements (step 50).
However, the headers are not only modified by replacing the
synchronization word with the length of the respective
channel element as in step 56. Rather, in steps 180 and 182
corresponding to steps 118 and 122 of Fig. 7, further
modifications follow. In step 180, the pointer in the side
information of every channel element is set to zero, and in
step 182, the bit rate index in the header of every channel
element is changed such that as described above, the MP3
data block length depending on the bit rate is sufficient
to include all audio data of this channel element or the
pertaining time mark, respectively, together with the size
of the header and the side information. Step 182 might also
comprise converting the padding bits in the headers of the
successive channel elements to produce an exact bit rate
later when supplying the MPEG-4 audio data stream formed by
the method of Fig. 9 to a decoder operating according to
the method of Fig. 7 but without steps 118 and 122. The
padding can of course also be performed on the decoder side
within step 128.
In step 182, it can useful to set the bit rate index not to
the highest possible value as described with regard to step
122. The value can also be set to the minimum value, which
is sufficient to take up all audio data, the header and the
side information of a channel element in a calculated MP3
frame length, which can also mean that in the case of
passages of the coded audio piece that can be coded with a
lesser amount of coefficients, the bit rate index is
reduced.
After these modifications, in steps 60 and 62, merely the
file header (AudioSpecificConfig) is generated, and the
same is output together with the MP3 channel elements as
MPEG-4 audio data stream. The same can, as has already been
mentioned, be played according to the method of Fig. 7,
wherein, however, steps 118 and 122 can be omitted, which
eases the implementation on the decoder side. However,
steps 42, 50, 56, 180, 182 and 60 can be performed in any
order.
The previous description related merely exemplarily to MP3
data streams with fixed data block bit length. Of course,
MP3 data streams with variable data block length can be
processed according to the previous embodiments, wherein
--- ---_..__._ uvuuiuc iui J.XJ.C lucmac conversion
could also be implemented In software. The Implementation
can be made on a digital memory medium, particularly a disk
or a CD with electronically readable control signals, which
can cooperate with a programmable computer system such that
the respective method is performed. Thus, generally, the
invention consists also of a computer program__psoduct with
a program code stored on a machine-readable carrier for
performing the inventive method when the computer program
product runs on a computer. In other words, the invention
can also be realized as a computer program with a program
code for performing the method when the computer program
runs on a computer.
WE CLAIM:
1. A method for converting a first audio data stream (10) representing a
coded audio signal comprising time periods and having a first file format into a
second audio data stream representing the coded audio signal and having a
second file format, wherein a time period comprises a number of audio values,
and wherein, according to the first file format, the first audio data stream is
divided into subsequent data blocks (10a-10c), wherein a data block comprises a
determination block (14, 16) and data block audio data (18), wherein
determination block audio data are associated to the determination block (14,
16), which are obtained by coding a time period, wherein the determination
block comprises a pointer pointing to a beginning of the determination block
audio data (12a-12c), and wherein an end of the determination block audio data
(12a-12c) lies prior to a beginning of determination block audio data (12b, 12c)
in the audio data stream associated to a next data block, comprising the steps
of:
combining (42) the determination block audio data (44, 46) associated to a
determination block of at least two data blocks to obtain contiguous
determination block audio data (48) forming part of the second audio data
stream;
adding (50) the contiguous determination block audio determination block audio
data (48) to the determination block (14, 16) to which the determination block
audio data (44, 46) are associated, from which the contiguous determination
block audio data are obtained, to obtain a channel element (52a);
arranging the channel elements to obtain the second audio data stream; and
modifying (56) the channel element (54a-54c) so that the same includes a length
indication indicating the amount of data of the channel element (54a-54c) or an
amount of data of the contiguous determination block audio data, wherein the
step of modifying comprises replacing (56) a redundant part identical for all
determination blocks by the length indication.
2. The method as claimed in claim 1, comprising the steps of:
placing (60, 62) an overall determination block in front of the second audio data
stream, wherein the overall determination block has the redundant part identical
for all determination blocks.
3. The method as claimed in claim 1 or 2, wherein the step of combining
comprises the sub-steps of:
reading the pointer in a determination block;
reading a first part of the determination block audio data included in data block
audio data of one of the at least two data blocks and comprising the beginning
of the determination block audio data to which the pointer of the determination
block points;
reading a second part of the determination block audio data included in data
block audio data of the other of the at least two data blocks and comprising the
end of the determination block audio data; and
combining the first and second parts.
4. A method for combining a first audio data stream representing a coded
first audio signal and a second audio data stream representing a coded second
audio signal into a multi-channel audio data stream, comprising the steps of:
conveying the first audio data stream into a first sub-audio data stream
according to the method of one of claims 1 to 3, and
converting the second audio data stream into a second sub-audio data stream
according to the method of one of claims 1 to 3,
wherein the steps of arranging are performed such that the two sub-audio data
streams together form the multi channel audio data stream, and that in the multi
channel audio data stream the channel elements (70a) of the first sub-audio data
stream and the channel element (72a) of the second sub-audio data stream
containing contiguous determination block audio data obtained by coding time
periods equal in time are arranged successively in a contiguous access unit (78).
5. The method as claimed in claim 4, comprising the steps of:
placing an overall determination block in front of the second data audio stream,
the overall determination block including a format indication indicating in which
order the channel elements (70a) of the first sub-audio data stream and the
second sub-audio stream (70b) are arranged in the access units (78).
6. The method as claimed in one of the previous claims, wherein the data
blocks are data blocks of equal or predetermined variable size depending on a
sample rate indication and a bit rate indication in the determination block of the
same.
7. A method for converting a first audio data stream representing a coded
audio signal comprising time periods and having a first file format, into a second
and having a second file format, wherein a time period comprises a number of
audio values, and wherein, according to the first file format, the first audio data
stream is divided into subsequent data blocks, wherein a data block comprises a
determination block and data block audio data, comprising the step of:
modifying the data blocks so that the same includes a length indication indicating
the amount of data of the data blocks or an amount of data of the data block
audio data to obtain channel elements forming the second audio data stream
from the data blocks, wherein the step of modifying includes..-replacing a
redundant part identical for all determination blocks by the length indication.
8. The method as claimed in one of claims 1 to 3, comprising the steps of:
resetting (180) the pointers in the determination blocks, so that the same
indicate as a beginning of the determination block audio data that the
determination block audio data begin immediately after the respective
determination block; and
changing (182) the bit rate indications in the determination blocks such that a
data block length depending on a bit rate indication according to the first audio
file format is sufficient to take up the respective determination block and the
associated determination block audio data.
9. A method for decoding a second audio data stream representing a coded
audio signal comprising time periods and having a second file format, based on a
decoder, which is able to decode a first audio data stream representing the
coded signal and having a first file format, into an audio signal, wherein a time
period comprises a number of audio values, and wherein according to the first
file format, the first audio data stream is divided into successive data blocks
(lOa-lOc), wherein a data block has a determination block (14, 16), wherein the
determination block includes a pointer pointing to a beginning of the
determination block audio data (12a-12c), and wherein an end of the
determination block audio data (12a-12c) is prior to a beginning of determination
block audio data (12a-12c) in the audio data stream associated to a next data
block, and wherein the second audio data stream is divided into channel
elements according to the second file format, wherein a channel element
comprises contiguous determination block audio data (44, 46) obtained by
combining determination block audio data associated to a determination block
from two data blocks, and the associated determination block in a form wherein
a previously redundant part, which is identical for all determination blocks, is
modified to be replaced by a length indication indicating the amount of data of
the respective channel element or an amount of data of the respective
contiguous determination block data, comprising the steps of:
forming an input data stream representing the coded audio signal and having a
first file format, from the second audio data stream by
parsing the second audio data stream with the help of the length indications;
resetting the pointers in the determination blocks of the channel elements of the
second audio data stream, so that the same indicate as a beginning of the
determination block audio data that the determination block audio data begin
immediately after the respective determination block to obtain reset
determination blocks;
changing a bit rate indication in the determination blocks of the channel
elements of the second audio data stream so that a data block length depending
on the bit rate indication according to the second audio file format is sufficient to
take up the respective determination block and the associated determination
block audio data to obtain bit rate-changed and reset determination blocks; and
inserting bits between every channel element and the subsequent channel
element, so that the length of every channel element plus the inserted bits is
adapted to the changed bit rate indication, and
supplying the input data stream to the decoder according to the changed bit rate
indication to obtain the audio signal.
10. An apparatus for converting a first audio data stream (10) representing a
coded audio signal comprising time periods and having a first file format, into a
second audio data stream representing the coded audio signal and having a
second file format, wherein a time period comprises a number of audio values,
and wherein according to the first file format, the first audio data stream is
divided into subsequent data blocks (lOa-lOc), wherein a data block comprises a
determination block (14, 16) and data block audio data (18), wherein
determination block audio data are associated to the determination block (14,
16), which are obtained by coding a time period, wherein the determination
block comprises a pointing to a beginning of the determination block audio data
(12a-12c), and wherein an end of the determination block audio data (12a-12c)
lies prior to a beginning of determination block audio data (12b, 12c) in the
audio data stream associated to a next data block, comprising:
a means for combining (42) the determination block audio data (44, 46)
associated to a determination block of two data blocks to obtain contiguous
determination block audio data (48) forming part of the second audio data
stream;
a means for adding (50) the contiguous determination block audio data (48) to
the determination block (14, 16) to which the determination block audio data
(44,46) are associated, form which the contiguous determination block audio
data are obtained, to obtain a channel element (52a);
a means for arranging the channel elements to obtain the second audio data
stream; and
a means for modifying (56) the channel element (54a-54c), so that the same
includes a length indication indicating the amount of data of the channel element
(54a-54c) or the amount of data of the contiguous determination block audio
data, wherein the means for modifying (56) is formed to replace a redundant
part, which is identical for all determination blocks, by the length indication.
11. An apparatus for converting a first audio data stream representing a
coded audio signal comprising time periods and having a first file format, into a
second audio data stream representing the coded audio signal and having a
second file format, wherein a time period comprises a number of audio values,
and wherein, according to the first file format, the first audio data stream is
divided into subsequent data blocks, wherein a data block comprises a
determination block and data block audio data, comprising
a means for modifying the data blocks so that the same includes a length
indication indicating the amount of data of the data blocks or an amount of data
of the data block audio data to obtain channel elements forming the second
audio data stream from the data blocks, wherein the step of modifying includes
replacing a redundant part, which is identical for all determination blocks, by the
length indication.
12. An apparatus for decoding a second audio data stream representing a
coded audio signal comprising time periods and having a second file format,
based on a decoder, which is able to decode a first audio data stream
representing the coded signal and having a first file format, into an audio signal,
wherein a time period comprises a number of audio values, and wherein
according to the first file format, the first audio data stream is divided into a
successive data blocks (lOa-lOc), wherein a data block has a determination
block (14, 16) and data block audio data (18), wherein determination block audio
data, which are obtained by coding a time period, are associated to the
determination block (14, 16), wherein the determination block includes a pointer
pointing to a beginning of the determination block audio data (12a-12c), and
wherein an end of the determination block audio data (12a-12c) is prior to a
beginning of determination block audio data (12a-12c) in the audio data stream
associated to a next data block, and wherein the second audio data stream is
divided into channel elements according to the second file format, wherein a
channel element comprises contiguous determination block audio data (44, 46)
obtained by combining determination block audio data associated to a
determination block from two data blocks, and the associated determination
block, in a form wherein a previously redundant part, which is identical for all
determination blocks, is modified to be replaced by a length indication indicating
the amount of data of the respective channel element or an amount of data of
the respective contiguous determination block data comprising:
a means for forming an input data stream representing the coded audio signal
and having a first file format, from the second audio data stream by
parsing the second audio data stream with the help of the length indications;
resetting the pointers in the determination blocks of the channel elements of the
second audio data stream, so that the same indicate as a beginning of the
determination block audio data that the determination block audio data begin
immediately after the respective determination block to obtain reset
determination blocks;
changing a bit rate indication in the determination blocks of the channel
elements of the second audio data stream so that a data block length depending
on the bit rate indication according to the second audio file format is sufficient to
take up the respective determination block and the associated determination
block audio to obtain bit rate-changed and reset determination blocks; and
inserting bits between every channel element and the subsequent channel
element, so that the length of every channel element plus the inserted bits is
adapted to the changed bit rate indication, and
a means for supplying the input data stream to the decoder according to the
changed bit rate indication to obtain the audio signal.

A method for converting a first audio data stream (10) representing a coded
audio signal comprising time periods and having a first file format into a second
audio data stream representing the coded audio signal and having a second file
format, wherein a time period comprises a number of audio values, and wherein,
according to the first file format, the first audio data stream is divided into
subsequent data blocks (10a-10c), wherein a data block comprises a
determination block (14, 16) and data block audio data (18), wherein
determination block audio data are associated to the determination block (14,
16), which are obtained by coding a time period, wherein the determination
block comprises a pointer pointing to a beginning of the determination block
audio data (12a-12c), and wherein an end of the determination block audio data
(12a-12c) lies prior to a beginning of determination block audio data (12b, 12c)
in the audio data stream associated to a next data block..