METHOD, DEVICE AND/OR SYSTEM FOR TRANSMITTING AND/OR RECEIVING DIGITAL INFORMATION
The present invention provides a method, device and/or system for transmitting and/or receiving digital information, wherein a personal computer comprising a storage medium for the information is adapted such that the information, irrespective of its format, is automatically readable/writable from and to the storage medium.
The present invention enables storage and/or further processing of digital audio information from different types of equipment without loss of quality due to repeat - ed compression. The required equipment of standard manufacture is inexpensive and a single device will suffice for the different types of information.
Existing equipment for transmitting digital information audio files, for instance coded in accordance with MPEG layer II, often makes use of recorders which can transmit digital data files to a studio, for instance for radio broadcasts, via the ISDN net or also remotely via GSM. Such equipment is rather expensive, while the possibilities of editing and co-transmission of additional information are limited.
According to the present invention it is possible to suffice with a personal computer of existing manufacture and a program which can be specifically tailored to the wishes of the user. Real time transmission of pre-record- ed audio becomes possible without loss of quality, while existing equipment can likewise communicate with a personal computer provided with the computer program.
The present invention will be elucidated on the basis of the following description of a preferred embodi- ent thereof, wherein reference is made to the annexed drawings, in which: fig. 1 shows a diagram of a preferred embodiment of the present invention; fig. 2 shows a more detailed diagram of the diagram of fig. 1;
fig. 3 is a diagram explaining the system shown schematically in fig. 1 and 2 ,* fig. 4 shows a flow chart of an operation of an operation of the system shown in fig. 1, 2 and 3 ; fig. 5 shows a flow chart of a detail of fig. 4; fig. 6 shows a flow chart of a detail of fig. 5 ; fig. 7 shows a detail of the flow chart of fig. 4; fig. 8 shows a detail of the flow chart of fig. 4 ; fig. 9 shows a flow chart of a detail of the flow chart of fig. 4 ; fig. 10 shows a flow chart of a detail of the flow chart of fig. 4; fig. 11 shows a flow chart of a detail of the flow chart of fig. 4; fig- 12 shows a flow chart of a detail of the flow chart of fig. 4 ;
A preferred embodiment of a system 1 (fig. 1) comprises a personal computer 2, preferably with a 486 processor or above, which includes an analog audio board 3 of known design which has an analog input 4 and an analog output 5 and a digital output 6. The audio board 3 is provided with an analog/digital converter 7, a digital/analog converter 8, a compression unit 9 for coding, for instance in accordance with MPEG layer II, and a conversion unit 10 for converting from digital to digital. The signal coded for the first time in accordance with MPEG layer-II is stored on a digital storage means, for instance a hard disk 11, in a section schematically designated with 12. Likewise situated on the receiving side is a personal computer 22 of the type 486 or above and an audio board 23 with components of a nature similar to audio board 3. The PC 22 is likewise provided with a storage means 24, for instance in the form of a hard disk, which can communicate with hard disk 11 via a computer network, modem, an ISDN line or a diskette as schematically designated by block 25. The figure further shows schematically the section 26 onto which is also recorded the digital signal which has been coded for the first time.
By way of comparison with the present invention, figure 2 shows an existing system consisting of two so- called dedicated devices 30 and 31 which are provided with a telephone dialler and can communicate with each other via a public ISDN net, designated schematically with 32. The telephone diallers are designated with 33 respectively 34. Use is increasingly being made of such systems for broadcasting on-the-spot sound recordings, for instance of an interview or the like. In figure 2, AD designates an analog/digital conversion, DA designates digital/analog conversion, C a compression step, ED an originally digital signal in accordance with MPEG-2 layer- II and NOD a not originally digital signal in accordance with MPEG layer- II, i.e. after decompression and renewed compression. U designates the linear decompression from digital to digital.
Without discussing existing systems in further detail, such as those made commercially available by applicant for instance under the name Digicorder, it is apparent that in particular conditions loss of quality occurs due to the digital decompression step, either during transmission via the public network 32 or through the respective digital outputs 35, 36.
Figure 3 shows a diagram of the present invention, wherein a dedicated device 30' is connected via a network 41 to a personal computer 42 which is provided with a digital storage means such as a hard disk 43 and which is provided with an ISDN board 44 such as is available commercially. The software of the system according to the present invention is designated schematically with 45. The software ISYS makes use of a compression algorithm in order to transmit audio of good quality over the ISDN line. In this preferred embodiment use is made of the MPEG standard, wherein the user of the software is able to change particular settings, such as the number of bits per second (bit rate) produced by the audio board, for instance from 32 kbits per second, the number of samples per second of the audio signal, from 16 kHz, the recording mode, mono, stereo or joint stereo, wherein in the case of stereo and joint stereo the bit rate doubles,
the header bits of the digital data for transmitting and/or an analog digital input signal. It is further possible to use algorithms other than MPEG, for instance G.711 and G.722. Because a computer of standard manufacture is used in the system according to the present invention which is then provided with the software, the equipment can be used for other purposes, such as Internet, editing of audio using the audio board, word processing, planning and the like. Existing prior art hardware is only suitable for audio transmission. The ISYS software can select and transmit audio files which have been digitally recorded and stored without the loss of quality occurring as outlined above with reference to fig. 2. The operation of the software will be further elucidated with reference to figures 4-12. At the start of a live recording 50 a check is first of all made at 51 whether a connection has been made. In the negative case recording and/or playback is stopped at 57 and the soft- ware is placed in rest mode at 58. In the affirmative case a start is made at 52 and a check is made at 53 as to whether the connection is present. At 54 playing is activated, at 55 recording and at 56 the ISDN connection. At 51 (fig. 5) the number of required channels is determined. The number is first set to 0 at 61, the required number is subsequently determined at 62 from the bit rate divided by the number 64, whereafter at 63 a channel link is made. At 64 a check is made as to whether this is actually present, whereafter at 65 the channel counter is increased by 1. At 66 is checked whether the number is already sufficient and in the negative case the loop is repeated. At 67 the program returns to the main program either after the connection has been effected at 68 or the connection has not been effected at 69. Activate play 54 (fig. 6) commences with the determination 71 as to whether an audio frame has been received. If it is determined at 72 that no earlier frames were received, the play timer is activated at 73. If it is determined at 74 that the play timer is active, a check is made at 75 whether 12 milliseconds have passed.
The play timer is stopped at 76, whereafter playback commences at 77. A frame is subsequently transmitted to the audio board at 78, whereafter the program returns to the main program at 79. Activation of recording 55 (fig. 7) begins with determining whether an audio frame is recorded at 81, whereafter at 82 the bit sequence of each byte in the frame is reversed. At 83 the frame is transmitted via ISDN and the program returns to the main program at 84. On activation of the ISDN according to 56 (fig. 8), it is determined at 85 whether data has been received, whereafter in the affirmative case data is copied to a buffer at 86. At 87 the frame is read, while the program returns to the main program at 88. A frame is subsequently read according to 87, by determining at 91, as designated in fig. 9, whether audio has been found, determining at 92 whether a header has been found and performing a check at 93, whereafter it is determined at 94 whether one frame has been received and is ready for playback so that the program can return to the main program at 95.
Whether audio has been found according to 91 (fig. 10) is established by determining whether according to the MPEG standard from a position 0, which is fixed at 102, fourteen ones have been found at this level (FF F in hexadecimal notation) . As long as it is determined at 103 that the position is smaller than the bytes in the buffer and it is determined at 104 that the position is greater than 0, it is determined via steps 105, 106 and 107 whether the present byte is unequal to OxFF and whether the previous byte is equal to OxFF, whereafter it is determined at 109 that audio has been found, while the program returns to the main program at 111. When the position in the buffer equals 0, the value of the previ- cus byte is made equal to 0x00 at 112 because no previous byte exists. At 110 it is determined that no audio has been found, while the program returns to the main program at 111. At 112 the value of the previous byte is determined .
Whether audio has been found according to 92 (fig. 11) is established by determining whether, according to the MPEG standard, from the position which has been found in figure 10 fourteen ones have been found (FF F in hexadecimal notation) . As long as it is determined at 113 that the position is smaller than the number of bytes in the buffer, the number of ones is determined per byte. At 114 the position of the byte is determined, while at 115 the position in the byte, the bit position, is deter- mined. Because a byte consists of 8 bits, it is determined at 116 whether the position in the byte is smaller than eight. If the position is greater than or equal to eight, a jump is then made to the next byte in 120 and the program returns to 113. As long as the position is smaller than eight, it is then determined at 117 whether there is a one at the present position in the byte. If there is a one at this position, the number of ones is increased by one at 118 and at 119 a jump is made to the next position in the byte. The program then returns to 116. When it is determined at 117 that there is not a one, it is determined at 121 whether the number of ones is more than or equal to fourteen. If less than fourteen ones are counted, a jump is made to 119 and the counting of the ones from the present position is continued. When it is determined at 121 that fourteen or more ones in succession have been counted, it is determined at 122 whether the position in the byte is greater than or equal to six. If this is the case, the present position in the buffer is decreased by one at 123. If this is not the case, the present position in the buffer is decreased by two at 124. In accordance with the MPEG standard the fourteen ones must always start at the beginning of a . byte. This therefore implies that the position in the byte (the bit position) must be equal to 0. In order to achieve this, it is determined in 125 how many steps all bits must be shifted in order to place the first one at position 0 in the byte. At 126 it is determined that an audio header has been found. When it is determined at 113 that the end of the buffer has been reached, it is deter-
mined at 127 that no audio header has been found. The program returns to the main program at 128.
Whether according to 93 a frame is complete and playable (fig. 12) is determined by checking at 129 whether all bytes required for a frame are present. If this is not the case, it is determined at 134 that the frame is not complete and the program returns to the main program at 135. When it is determined at 129 that the frame is complete, it is determined at 130 whether all bits have to be shifted (how much all bits must be shifted has been determined in fig. 11) . If it is determined at 130 to shift the bits, this is done at 131. At 132 is checked whether the first four bytes of the header conform to the MPEG standard. If these bytes are correct, it is then determined at 133 that the frame is playable and the program returns to the main program at 135. In the other case it is determined at 134 that the frame is not playable and the program returns to the main program at 135.
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