ITU-T J 42-1988 CHARACTERISTICS OF EQUIPMENT FOR THE CODING OF ANALOGUE MEDIUM QUALITY SOUND-PROGRAMME SIGNALS FOR TRANSMISSION ON 384-kbit s CHANNELS《用于在384kbit s通道中传输的中等质量模拟声音节目信.pdf

1、INTERNATIONAL TELECOMMUNICATION UNION)45G134 * TELECOMMUNICATIONSTANDARDIZATION SECTOROF ITU4%,%6)3)/.G0G0!.$G0G03/5.$G0G042!.3-)33)/.#(!2!#4%2)34)#3G0G0/ amended at Melbourne, 1988)1 General1.1 This Recommendation gives the characteristics of equipment for the coding of 7 kHz monophonic analoguesou

2、nd-programme signals into a digital signal. Two monophonic digital signals can be combined to form a 384-kbit/ssignal already specified in Recommendation J.41.1.2 Equipment for coding of analogue sound-programme signals, as specified in this Recommendation, can be:a) A stand alone encoder/decoder wi

3、th a digital interface at 384 kbit/s. The encoder operation and the decoderoperation may be performed in two separate equipments or in the same equipment.b) A combined encoder-multiplex decoder-demultiplex with a digital interface at 1544 or 2048 kbit/s. Theencoder-multiplex operation and the decode

4、r-multiplex operation may be performed in two separateequipments or in the same equipment.In case b) it is not mandatory to provide an external digital sound-programme access port at 384 kbit/s.2 Transmission performanceThe transmission performance per encoder/decoder pair shall be such that the lim

5、its specified inRecommendation J.23 (CCIR Recommendation 503) are not exceeded by three encoder/decoder pairs connected intandem at audio frequencies.3 Method of encoding3.1 The recommended encoding laws are as specified in 1.3.2 These encoding laws are based on a uniformly quantized, 14-bit per sam

6、ple PCM technique with compandingand employ either:a) eleven-segment 14 to 11 bit instantaneous A-law companding, orb) five-range 14 to 10 bit near-instantaneous companding.3.3 Equipment characteristics common to both methods of encoding are:Nominal audio bandwidth: 0.05 to 7 kHzAudio interface: see

7、 Recommendation J.23, 2.Sampling frequency: 16 (1 5 105) kHz.Pre/de-emphasis: Recommendation J.17 with6.5 dB attenuation at 800 Hz.Note Pre-emphasis and de-emphasis are not used by the Administrations of Canada, Japan and the UnitedStates of America on their national circuits and on international ci

8、rcuits between each other, but are used oninternational circuits to other countries.4 Equipment using instantaneous companding4.1 Coding table4.1.1 The coding law is specified in Table 1/J.41.2 Fascicle III.6 - Rec. J.424.1.2 The allocation of character signals (PCM code words) is also given in Tabl

9、e 1/J.41. Two variants (A and B) ofcharacter signals are allowed.Note In the case of digital interconnection between variants A and B, the conversion from one set of charactersignals to the other set in Table 1/J.41, can be done without any performance degradation. In the case of analogueinterconnec

10、tion, a reduction in the S/N ratio, in the order of 3 dB, is expected.4.2 Bit ratesNominal source coding bit rate (16 kHz 11 bit/sample) 176 kbit/sError protection (16 kHz 1 bit/sample) 16 kbit/sTransmission bit rate per sound-programme signal 192 kbit/sChannel bit rate for 2 sound-programme signals

11、 384 kbit/s4.3 Overload levelThe overload level for a sine-wave signal at zero dB insertion loss frequency (2.1 kHz) of the pre-emphasisis+ 15 dBm0s.4.4 Digital signal formatThe character signal bit sequences for variants A and B, are shown in Figure 1/J.41.4.4.1 Variant AWhen transmitting two monop

12、honic digital signals as one 384 kbit/s signal, with respect to the code wordinterleaving shown in Figure 1/J.41, the first two 12 bit code words are allocated to 7 kHz channel No. 1 and thesecond two 12 bit code words are allocated to 7 kHz channel No. 2.4.4.2 Variant BThe 12 bit code word assignme

13、nts when transmitting two monophonic digital signals as one 384-kbit/s signal isunder study.4.5 Bit error protectionOne parity bit is added to each 11-bit character signal.4.5.1 Variant AThe five most important bits of each sample are protected against errors by means of a parity bit. In theconverte

14、r of the transmitting part, the parity bit is added as the 12th bit to each code word. Its value is fixed so that the6 bit parity block always contains only an odd number of one values. In order that even bit error structures can alsoresult in parity violations, the protected and unprotected bits of

15、 each code word are interleaved in ascending anddescending sequence, as shown in Figure 1/J.41.4.5.2 Variant BThe added parity bit shall be based on the 7 most significant bits of the 11-bit PCM word. These are bits S, X,Y, Z, A, B, C. The parity of “ones” bit shall be even. Since the chord bits (X,

16、 Y, Z) always contain a one, the minimumnumber of ones per sample is 2, resulting in a minimum ones density of 1/6.4.5.3 Error concealmentIf a parity violation is detected, an error concealment technique should be applied (for instance, replacement byinterpolation, extrapolation or repetition). For

17、multiple parity violation (error bursts), a muting technique should beapplied.4.6 Digital interface at 384 kbit/sUnder study (see Recommendations G.735 and G.737).4.7 SynchronizationThe coding equipment operates in synchronism with the clock of subsequent multiplex equipment or thenetwork clock. In

18、cases where the digital interface is provided, bit and byte (24 bit, as shown in Figure 1/J.41) timinginformation is required.Fascicle III.6 - Rec. J.42 3Variant A: A solution for synchronous access is given in the Recommandations G.735 and G.737.Variant B: The solution for synchronous access is und

19、er study.4.8 Fault condition and consequent actions4.8.1 Variant AWhere a 384-kbit/s digital interface is provided, the same principles for fault conditions and subsequent actionsas those outlined in Recommendation G.732, should be followed.4.8.2 Variant BUnder study.5 Equipment using near-instantan

20、eous companding5.1 IntroductionThe equipment described in this section uses the near-instantaneous method of companding in the coding ofmedium quality sound-programme signals into digital form.A two-stage process is used in the encoding equipment:a) Conversion of a 7 kHz channel into a 169 kbit/s st

21、ream.Note The value of 169 kbit/s has been chosen to allow for the possible multiplexing of 12 channels into a2048 kbit/s dedicated frame format.b) Asynchronous insertion of two synchronous 169 kbit/s streams into a 384 kbit/s stream.Note The asynchronous insertion of two synchronous 169 kbit/s stre

22、ams into a 384 kbit/s stream allows theuse, at the encoder location, of a clock not necessarily synchronous to the network clock. It can beadvantageous when the encoder equipment and the insertion equipment (see Recommendations G.735 andG.737) are located in different places, and when the transmissi

23、on link between them is unidirectional,and the reverse processes in the decoding equipment.5.2 Conversion from 7 kHz to 169 kbit/s and constitution of the 338-kbit/s signal5.2.1 Overload levelThe overload level for a sine-wave signal at the zero dB insertion loss frequency (2.1 kHz) of the pre-empha

24、siscircuit is + 12 dBm0s.5.2.2 CompandingThe same near-instantaneous companding procedure with a block of 32 samples (2 ms) as described in 5.2.2of Recommendation J.41, is used. The character signal is coded in 2s complement form.5.2.3 Constitution of the 338-kbit/s signalTwo 7-kHz channels (C1 and

25、C2) are contained in one 338-kbit/s stream. The frame structure of the 338 kbit/sstream is defined in 5.2.5 and in Figure 3/J.41. The following numbering of the samples within a given multiframe isdefined as follows (see Figure 3/J.41):Sample n of the multiframe is sample (n 96i) of frame i0 n 191 i

26、 = 0 or 1Using the above notation, the following relationship between the bits of the 338 kbit/s multiframe and channelsC1 and C2 can be defined:Sample 2n of the multiframe corresponds to sample n of channel C1Sample (2n + 1) of the multiframe corresponds to sample n of channel C20 n 9 5Range coding

27、 information associated with block (2n 1) of the multiframe is allocated to block n of channelC1 (derived from C1 samples in blocks (2n 1) and (2n) of the multiframe).4 Fascicle III.6 - Rec. J.42Range coding information associated with block (2n) of the multiframe is allocated to block n of channel

28、C2(derived from C2 samples in blocks (2n 1) and (2n) of the multiframe).1 n 3The range coding information and its protection, the sample format and the sample error protection are definedand transmitted as specified in this Recommendation and in 5.2.3 to 5.2.5 of Recommendation J.41.The criteria for

29、 loss and recovery of frame alignment at 338 kbit/s is defined in 5.2.8 of RecommendationJ.41.5.3 Conversion from 338 kbit/s to 383 kbit/sSee Recommendation J.41, 5.3.5.4 Digital interface at 384 kbit/sUnder study.5.5 Fault conditions and consequent actionUnder study.6 Digital interface between equipments using different coding standardsUnder studyReferences1 CCIR Recommandation Transmission of analogue high-quality sound-programme signals on mixed analogueand digital circuits using 384 kbit/s channels, Vol. XII, Rec. 660, ITU, Geneva, 1986.