Error concealment in digital audio receiver

Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission

Reexamination Certificate

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Details

C704S228000, C704S229000

Reexamination Certificate

active

06687670

ABSTRACT:

The invention relates in general to detection and concealment of errors in a signal transmitted in digital form from a transmitter to a receiver. In particular the invention relates to detection and concealment of transmission errors in an audio signal processed in the form of frames by a digital audio receiver.
Transmission of an audio signal in digital form from a transmitter to a receiver is known as such and it is going be become more common as digital television and broadcasting systems replace older systems based on analog frequency modulation. Known telecommunications standards dealing with the transmission of digital audio signals include the ETS 300 401 standard by the European Broadcasting Union (EBU) and European Telecommunications Standards Institute (ETSI) and the ISO/IEC 11172-3 and ISO/IEC 13818-3 standards by the International Standard Organization (ISO) and International Electrotechnical Commission (IEC). These standards specify a certain frame structure for the transmission of a digital audio signal. The ETS 300 401 standard, which is also called the DAB (Digital Audio Broadcasting) standard, specifies a frame structure which in a way is a special case of the frame structure specified in the ISO/IEC 11172-3 and ISO/IEC 13818-3 standards as it contains additional specifications concerning frame structure particulars left open in the earlier standards. With an audio signal sampling frequency of 48 kHz the DAB standard is based on the ISO/IEC 11172-3 standard and with a sampling frequency of 24 kHz on the ISO/IEC 13818-3 standard. To illustrate the background of the invention, the structure of the audio frame according to the aforementioned standards and its processing in transmitter and receiver apparatuses is described in brief below.
FIG. 1
is a simplified block diagram of an apparatus
1
according to the ISO/IEC 11172-3 and 13818-3 Layer II standards generating DAB frames from a pulse-code-modulated (PCM) audio signal. The apparatus comprises an input port
2
, output port
3
, and between them, a filter bank
4
, quantising and coding block
5
, and a frame generating block
6
, connected in series. In parallel with the filter bank
4
, there is a psychoacoustic model block
7
the input signal of which is the same as the filter bank input signal. The outputs of blocks
4
and
7
are taken to a bit allocation block
8
the output of which controls quantising and coding in block
5
. The apparatus also comprises a data port
9
such that digital program associated data brought thereto is taken to the frame generating block
6
which incorporates the program associated data in the frame structure.
FIG. 2
is a simplified block diagram of an apparatus
10
according to the ISO/IEC 11172-3 and 13818-3 Layer II standards decoding the frames generated by the transmitter shown in
FIG. 1
into a pulse-code-modulated audio signal. It comprises an input port
11
, output port
12
, and between them, a frame decoding block
13
, reconstructing block
14
and an inverse filter bank
15
, connected in series. The frame decoding block
13
is also connected with a data port
16
to take program associated data to other circuits of the receiver apparatus.
The audio signal is transmitted as frames between apparatuses according to
FIGS. 1 and 2
. The amount of data in a single frame corresponds to a 24- or 48-ms-long audio signal part. In addition to audio data proper the frame contains header information, checksums, information related to the processing of audio data, and program associated data, PAD. Since transmission paths are not ideal, errors may occur in the contents of the frames which affect the operation of the receiver in different ways depending on the location of the error in the frame.
FIG. 3
shows the structure of an audio frame
17
according to the DAB standard. The frame comprises an integer number of eight-bit bytes (not shown). It starts with a 32-bit header
18
, followed by a 16-bit CRC word
19
. The length of the bit allocation part
20
is 26 to 176 bits depending on the audio mode (single channel, dual channel, stereo, joint stereo) and sampling frequency used as well as on the bit rate used for transmitting the audio program. An SCFSI part contains instructions for the interpretation of the scale factor part
22
following it. The scale factors in the latter provide information about how the various parts of the signal were emphasised at the frame generation stage. Each scale factor is represented by a six-bit codeword (not shown) and the number of codewords in the frame varies according to how much variation there is in the different parts of the audio signal during the period represented by the frame. Part
23
contains the sampled values proper which represent the sampled audio signal. If the bits representing the sampled values do not fill the length of the space reserved for them, the empty part is filled with padding bits
24
.
There are in the end of the frame
17
, from right to left in the Figure, a fixed program associated data (F-PAD) field
25
, scale factor cyclic redundancy check (SCF CRC) error protection
26
for the audio data, and an extended program associated data (X-PAD) field
27
. The latter is not necessarily included in every audio frame. In accordance with the ETS 300 401 standard, the program associated data fields
25
and
27
are intended for the transmission of data that are closely related to the audio data proper included in the frame and that may have synchronisation requirements concerning the audio data. Their use is not mandatory. The F-PAD and X-PAD fields together form the program associated data (PAD) part. The F-PAD field particularly includes a two-bit X-PAD indicator (not shown) to indicate whether the frame includes an X-PAD field and if so, whether it is a four-byte, so-called short X-PAD field or a variable size X-PAD field.
FIG. 4
shows in more detail an audio frame header
18
the length of which is 32 bits (four bytes). The description to follow concerns both the ISO/IEC 11172-3 and ISO/IEC 13818-3 standards and the DAB standard so that the specifications required by the DAB standard are mentioned separately. The first twelve bits form a synchronisation word
29
in which all bits are ones. The next bit
30
is a so-called ID bit wherein value “1” corresponds to the application of the ISO/IEC 11172-3 standard and value “0” corresponds to the application of the ISO/IEC DIS 13818-3 standard in the audio signal processing. The length of the Layer field
31
is two bits and its value corresponds to the layer of the ISO/IEC 11172-3 standard in use. The DAB standard allows values “10” (Layer II) and “00” (reserved for future expansion). The protection bit
32
indicates whether there is a checksum in the frame, and its value according to the DAB standard is “0”, meaning a checksum is used. The next four-bit field
33
represents the bit rate of the audio program in use. The ISO/IEC 11172-3 and ISO/IEC 13818-3 standards do not allow the value “1111” in the field
33
. Furthermore, the DAB standard does not allow the value “0000”. The sampling frequency field
34
includes two bits representing the sampling frequency of the original pulse-code-modulated signal. According to the DAB standard, values “00” and “01” are not allowed in this field
34
. Value “01” corresponds to a 48-kHz sampling frequency if the ID bit is “1”, and to a 24-kHz sampling frequency if the ID bit is “0”. Value “11” is reserved for future expansion. A padding indicator bit
35
is “0” according to the DAB standard because there are no padding bits in the audio frame formed from a 48-kHz or 24-kHz PCM signal. According to the ISO/IEC 11172-3 and ISO/IEC 13818-3 standards, bit
35
is “1” if there are padding bits in the audio frame. The Private bit
36
, which is reserved for private use, has no significance according to the DAB, ISO/IEC 11172-3 and ISO/IEC 13818-3 standards.
A two-bit field
37
indicates the audio program's transmission mode which can be stereo (“00”), joint stereo (“01”), dual channel (“10”) or single c

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