.rs .\" Troff code generated by TPS Convert from ITU Original Files .\" Not Copyright ( c) 1991 .\" .\" Assumes tbl, eqn, MS macros, and lots of luck. .TA 1c 2c 3c 4c 5c 6c 7c 8c .ds CH .ds CF .EQ delim @@ .EN .nr LL 40.5P .nr ll 40.5P .nr HM 3P .nr FM 6P .nr PO 4P .nr PD 9p .po 4P .rs \v | 5i' .ce 1000 APPENDIX\ I .ce 0 .ce 1000 (to Recommendation G.722) .sp 9p .RT .ce 0 .ce 1000 \fBNetworking aspects\fR .sp 1P .RT .ce 0 .PP The purpose of this Appendix is to give a broad outline of the interaction of 64\ kbit/s (7\ kHz) audio coding with other parts of the digital network. Some general guidance is also offered. .sp 1P .RT .PP The establishment of the connection is beyond the scope of this Recommendation. .sp 1P .LP I.1 \fINetwork characteristics\fR .sp 9p .RT .PP This Recommendation is applicable to systems operating in networks which exhibit each of the following characteristics: .RT .LP i) availability of network octet timing at the terminals; .LP \fINote\fR \ \(em\ Octet timing may also be derived from control signals within the frame structure defined in Recommendation\ G.725; .LP ii) plesiochronous networking where the reference clocks meet the timing requirements given in Recommendation\ G.811, or synchronous networking; .LP iii) 64 kbit/s connection types having either of the following characteristics: .LP \(em full 64 kbit/s transparency, .LP \(em pulse density restriction as described in Recommendation\ G.802. .PP \fINote\fR \ \(em\ 64 kbit/s (7 kHz) audio coding can also operate in networks where there is substitution of a signalling bit for the 8th bit of the octet as described in Recommendation\ G.704, \(sc\ 3.1 or where there is 56\ kbit/s transparency only. However, a reduction of the audio bit rate and auxiliary data channel capacity occurs and only two modes of operation, denoted 1\ \fIbis\fR (unframed) and 3\ \fIbis\fR , are possible as follows: .LP \(em Mode 1 \fIbis\fR | 56 kbit/s for audio coding and no data channel; .LP \(em Mode 3 \fIbis\fR | 48 kbit/s for audio coding, a 6.4\ kbit/s data channel and 1.6\ kbit/s for service channel framing and mode control. .sp 1P .LP I.2 \fIIntegration into the telecommunications network\fR .sp 9p .RT .PP It is foreseen that the 64 kbit/s (7 kHz) audio coding system will be used for point\(hyto\(hypoint, multipoint and broadcast applications. Examples of particular uses are: commentary quality channels for broadcasting purposes and high quality speech for audio and video conferencing applications. .PP The coding system can operate over any 64 kbit/s bearer channel (see \(sc\ I.1), e.g.\ the public switched telephone network, leased circuits or over an ISDN. .PP Processes such as digital speech interpolation, echo control and digital pads must be disabled for the transmission of 64\ kbit/s (7\ kHz) audio coding. The disabling protocol is not the subject of this Recommendation. .PP It should be noted however that signal processing may occur in a multipoint conference unit (see \(sc\ I.7). .RT .sp 2P .LP I.3 \fIAudio performance of the 64 kbit/s (7 kHz) audio coding\fR \fIsystem\fR .sp 1P .RT .sp 1P .LP I.3.1 \fISpeech\fR .sp 9p .RT .PP The speech performance of the 64 kbit/s (7 kHz) audo coding system has been quantified in terms of \fIQ\fR\d\fIW\fR\u\(hyvalues, where\ \fIQ\fR\d\fIW\fR\uis a measure of the signal\(hyto\(hycorrelated noise ratio of the wideband system, measured in dB. Detailed information on \fIQ\fR \(hyvalue measurements may be found in Recommendation\ P.81. This Recommendation, although primarily intended for telephony bandwidth applications, has been used for the evaluation of wideband systems \(em\ signified by the subscript\ \fIW\fR \ \(em by use of an appropriate filter (50\(hy7000\ Hz). .PP For guidance purposes only, a \fIQ\fR\d\fIW\fR\uvalue of 38 dB corresponds approximately to a 128\ kbit/s (7\ kHz) PCM system (sampling frequency 16\ kHz, coding law as in Recommendation\ G.711), whereas a \fIQ\fR\d\fIW\fR\u\ value of 45\ dB is approximately equivalent to the audio parts of the coder interconnected without the intermediate SB\(hyADPCM coding process. .bp .PP Table I\(hy1/G.722 indicates the relative performance in \fIQ\fR\d\fIW\fR\uvalues for nominal input values. .RT .ce \fBH.T. [T24.722]\fR .ce TABLE\ I\(hy1/G.722 .ce \fBRelative levels of speech performance\fR .ce \fB(\fR .ce \fIQ\fI .ce | fBvalues)\fR .ps 9 .vs 11 .nr VS 11 .nr PS 9 .TS center box; cw(60p) | cw(24p) sw(60p) sw(60p) , ^ | c | c | c. Mode of operation Transcodings 1 { 4 Analogue according to Fig. I\(hy1/G.722 } { 4 Digital according to Fig. I\(hy2/G.722 } _ .T& cw(60p) | cw(24p) | cw(60p) | cw(60p) . 1 (64 kbit/s) 45 38 41 _ .T& cw(60p) | cw(24p) | cw(60p) | cw(60p) . 2 (56 kbit/s) 43 36 38 _ .T& cw(60p) | cw(24p) | cw(60p) | cw(60p) . 3 (48 kbit/s) 38 29 34 _ .TE .nr PS 9 .RT .ad r \fBTable I\(hy1/G.722 [T24.722], p.\fR .sp 1P .RT .ad b .RT .PP .sp 2 The performance of the 64 kbit/s (7 kHz) audio coding system has been found to be substantially unaffected by randomly distributed errors at BER levels as high as\ 1\(mu10\uD\dlF261\u4\d. High error ratios approaching\ 1\(mu10\uD\dlF261\u3\d produce perceptible degradation which may be considered tolerable in certain applications. .PP No particular problems have been experienced in the multiple talker condition and hence correct operation under normal conference conditions can safely be assumed. .PP The performance under conditions of mode mismatch (i.e. where the variant used in the decoder for a given octet does not correspond to the mode of operation) is considered in \(sc\ I.5. .RT .sp 1P .LP I.3.2 \fIMusic\fR .sp 9p .RT .PP Although primarily designed for speech, no significant distortions may be expected when coding a wide range of music material in Mode\ 1. Further study on the effects on music signals is a matter of Study Group CMTT. .RT .sp 2P .LP I.4 \fIAudio performance when interconnected with other coding systems on\fR \fIan analogue basis\fR .sp 1P .RT .sp 1P .LP I.4.1 \fI64 kbit/s PCM\fR .sp 9p .RT .PP Informal subjective tests carried out over a path consisting of an analogue interconnected combination of a 64\ kbit/s PCM link conforming to Recommendation\ G.711 and a 64\ kbit/s (7\ kHz) audio coding link has indicated that no interworking problems will occur. However, the performance of the combination will not be better than that of 64\ kbit/s PCM. .PP Interconnection of the two coding systems on a digital basis is the subject of \(sc\ I.8. .RT .sp 1P .LP I.4.2 \fI32 kbit/s ADPCM\fR .sp 9p .RT .PP An analogue interconnected combination of a 32 kbit/s ADPCM link conforming to Recommendation\ G.721 and a 64\ kbit/s (7\ kHz) audio coding link is not expected to pose any interworking problems. However, the performance of the combination will not be better than that of 32\ kbit/s ADPCM. .PP Interconnection of the two coding systems at a digital level is the subject of further study. .bp .RT .sp 1P .LP I.5 \fIAudio performance under\fR \fImode switching\fR .sp 9p .RT .PP It is recommended that mode switching should be performed synchronously between the transmitter and the receiver to maximize the audio performance. However, asynchronous mode switching may be considered since the condition of mode mismatch will probably be of limited duration and hence the corresponding performance is likely to be acceptable. Although not desirable, operation under permanent mode mismatch may be contemplated in exceptional circumstances. Table\ I\(hy2/G.722 indicates the relative performance under all mode mismatch combinations for nominal input levels. .RT .ce \fBH.T. [T25.722]\fR .ce TABLE\ I\(hy2/G.722 .ce \fBRelative speech performance under mode mismatch\fR .ce \fB(\fR .ce \fIQ\fI .ce | fBvalues)\fR .ps 9 .vs 11 .nr VS 11 .nr PS 9 .TS center box; cw(72p) | cw(60p) sw(60p) , ^ | c | c. { Bit rate used for audio reception } { Bit rate used for audio transmission } 56 kbit/s 48 kbit/s _ .T& cw(72p) | cw(60p) | cw(60p) . 64 kbit/s 41 35 _ .T& cw(72p) | cw(60p) | cw(60p) . 56 kbit/s \(em 36 .TE .LP \fINote\fR \ \(em\ The bits not used for audio coding have been replaced by bits of a pseudorandom sequence. .nr PS 9 .RT .ad r \fBTable I\(hy2/G.722 [T25.722], p.\fR .sp 1P .RT .ad b .RT .LP .sp 5 .sp 1P .LP I.6 \fIAuxiliary data channel\fR \fIperformance\fR .sp 9p .RT .PP The available combinations of audio and data channel bit rates depends on the connection types described in \(sc\ I.1\ iii). .PP The data channel is unaffected by the characteristics of the audio signal since the audio and data channels are effectively decoupled. The transparency of the data channel is limited only by the choice of signalling sequences which could be used to derive the terminal identification. If these sequences are chosen to be of a suitable format, the possibility of their simulation by audio or data bits can be made extremely low. Hence, for all practical purposes, the data channel may be assumed to be transparent. .PP The control of the data channel capacity is considered in Recommendation\ G.725. .PP Although the format of the data channel is not part of this Recommendation, it may be noted that the use of two completely independent 8\ kbit/s data channels when the total data channel capacity is 16\ kbit/s is not prohibited by the algorithm. .PP Under transmisssion error conditions the data channel is not subject to error multiplication due to the audio coding algorithm. .PP \fINote\fR \ \(em\ It might be possible to obtain additional data channel capacity by substituting data for the two bits normally allocated to the higher sub\(hyband with the consequent penalty of a reduction in the audio bandwidth. However, such an approach is likely to require a more stringent specification for the receive filter characteristics in order to minimize aliasing effects. .bp .RT .sp 1P .LP I.7 \fIMulti\(hypoint conference configuration\fR .sp 9p .RT .PP The specific features of a multipoint conference arrangement including control of the data channel, echo control, and handling of control messages between terminals, are beyond the scope of this Recommendation. However, the audio coding algorithm has been chosen to maintain maximum flexibility for multipoint conference arrangements which are likely to emerge. There are a number of general guidelines which should be noted: .RT .LP \(em To maximize audio performance, the highest audio bit rate possible, consistent with the transmitted data channel bit rate requirement, should be used for transmission into and out of the signal summing facility of the multipoint conference unit. .LP \fINote\fR \ \(em\ The signal summation must be carried out on a linear representation of the signals. .LP \(em The transmit and receive modes of a terminal or port of a multipoint conference unit do not necessarily have to be the same. .LP \(em Signal summing at the sub\(hyband uniform PCM level is preferred for the following reasons: .LP i) the hardware is minimized in the multipoint conference unit (MCU) by eliminating the need for quadrature mirror filters, .LP ii) signal quality is maximized and additional signal delay is eliminated by avoiding additional filtering, .LP iii) echo control is likely to be simpler to perform at the sub\(hyband level. .LP Figure I\(hy3/G.722 indicates a possible arrangement at the multipoint conference bridge with signal summing at the sub\(hyband level; .LP \(em For reasons of audio performance, the number of tandem connected multipoint conference units interconnected with 64\ kbit/s (7\ kHz) audio coding is limited to three, see Figure\ I\(hy4/G.722). .LP \(em In the case where the multipoint conference unit includes 64\ kbit/s PCM ports, digital transcoding principles equivalent to that described in \(sc\ I.8 should be used to derive the higher and lower sub\(hyband signals. .LP .rs .sp 9P .ad r \fBFigure I\(hy1/G.722, p.\fR .sp 1P .RT .ad b .RT .LP .rs .sp 15P .ad r \fBFigure I\(hy2/G.722, p.\fR .sp 1P .RT .ad b .RT .LP .bp .LP .rs .sp 22P .ad r \fBFigure I\(hy3/G.722, p.\fR .sp 1P .RT .ad b .RT .LP .rs .sp 12P .ad r \fBFigure I\(hy4/G.722, p.\fR .sp 1P .RT .ad b .RT .sp 1P .LP I.8 \fIDigital transcoding between the 64 kbit/s (7 kHz) audio coding\fR \fIsystem and 64 kbit/s PCM\fR .sp 9p .RT .PP Figure I\(hy5/G.722 indicates the method recommended for the digital interconnection of the 64\ kbit/s (7\ kHz) audio coding system and 64\ kbit/s PCM to Recommendation\ G.711. .PP The principle of transcoding from 64\ kbit/s PCM to 64\ kbit/s (7\ kHz) audio coding involves the conversion from A\(hylaw or \(*m\(hylaw PCM to uniform PCM and the insertion of interleaved alternate samples of zero amplitude to the 8\ kHz sampled uniform PCM signal to form a 16\ kHz sampled signal. This signal is then passed through a digital low pass filter sampled at 16\ kHz which does not significantly modify the baseband frequency response up to 3.4\ kHz and which attenuates the frequency components above 4.6\ kHz. The resulting signal is then applied to the sub\(hyband ADPCM encoder as shown in Figure\ I.3/G.722. .PP It should be noted that the use of the lower sub\(hyband alone to carry the information in a signal emanating from a 64\ kbit/s PCM link to Recommendation\ G.711 should be avoided. .bp .PP An alternative method of deriving two sub\(hyband signals from a 64\ kbit/s PCM signal using the low pass (LP) and high pass (HP) QM filter designs already employed for the 64\ kbit/s (7\ kHz) audio coding scheme is given in Figure\ I\(hy6/G.722. The objective is to generate a higher sub\(hyband signal which will eventually cancel the aliasing distortion introduced into the lower sub\(hyband signal. The 64\ kbit/s PCM signal is converted to uniform PCM and upsampled to 16\ kHz by inserting alternate zero\(hyvalued samples. The factor\ 2 multiplier is inserted to preserve unity gain. The lower sub\(hyband signal is derived by two identical stages of HP QM filtering following by 2:1 subsampling. The higher sub\(hyband signal is derived by two filtering stages, HP followed by LP, a factor 1/2 gain reduction, sign inversion, followed by 2:1 subsampling. When these two signals are input to the QM synthesis filter of Recommendation\ G.722, an appropriate 7\ kHz form of the original PCM is obtained. .PP Note that the upsampling and subsampling process should be synchronized so that instants of sample deletion correspond to the instants of zero\(hysample insertion. .PP Transcoding from 64 kbit/s (7 kHz) audio coding to 64\ kbit/s PCM can be achieved by taking the output signal from the sub\(hyband ADPCM decoder and performing the following three processes in turn: .RT .LP \(em digital low pass filtering (16 kHz sampling), which does not significantly modify the baseband frequency response up to 3.4\ kHz and which attenuates the frequency components above 4.6\ kHz; .LP \(em the deletion of alternate samples from the resulting 16\ kHz sampled signal; .LP \(em conversion from the resulting 8 kHz sampled uniform PCM signal to A\(hylaw or \(*m\(hylaw PCM. .PP \fINote\fR \ \(em\ The derivation of a 64 kbit/s PCM signal solely from the lower sub\(hyband of the 64\ kbit/s (7\ kHz) signal is subject to further study. .LP .rs .sp 26P .ad r \fBFigure I\(hy5/G.722, p.\fR .sp 1P .RT .ad b .RT .LP .bp .LP .rs .sp 17P .ad r \fBFigure I\(hy6/G.722, p.\fR .sp 1P .RT .ad b .RT .ce 1000 APPENDIX\ II .ce 0 .ce 1000 (to Recommendation G.722) .sp 9p .RT .ce 0 .ce 1000 \fBDigital test sequences\fR .sp 1P .RT .ce 0 .PP This Appendix gives information concerning the digital test sequences which should be used to aid verification of implementations of the ADPCM codec part of the wideband coding algorithm. Copies of the sequences are available on flexible disks (see \(sc\ II.4). .sp 1P .RT .sp 1P .LP II.1 \fIInput and output signals\fR .sp 9p .RT .PP Table II\(hy1/G.722 defines the input and output signals for the test sequences. It contains some signals (indicated by\ ##) peculiar to these test sequences in order to facilitate the interface between the test sequence generator/receiver and the encoder/decoder. 16\(hybit word formats for these input and output signals are shown in Figures\ II\(hy1/G.722, II\(hy2/G.722 and\ II\(hy3/G.722. .RT .sp 1P .LP II.2 \fIConfiguratins for the application of test sequences\fR .sp 9p .RT .PP Two configurations (Configuration 1 and Configuration 2) are appropriate for use with test sequences. In both configurations, a TEST signal is used to make the encoder and decoder ready to be tested with the digital test sequences. When the TEST signal is provided, the QMFs are by\(hypassed and the test sequences are applied directly to the ADPCM encoders or decoders. An RSS signal is extracted from the input test sequences\ X## (I## in decoder) and results in a reset signal RS for the encoder and decoder. The RS signal will be used to initialize state variables (those indicated by * in Table\ 13/G.722 to zero or specific values. .RT .sp 1P .LP II.2.1 \fIConfiguration 1\fR .sp 9p .RT .PP Configuration 1 shown in Figure II\(hy4/G.722 is a simplified version of Figures\ 4/G.722 and\ 5/G.722. The encoder input signals, XL and XH, are described in Table\ 12/G.722. These input signals are directly fed to the respective lower and higher sub\(hyband ADPCM encoders, by\(hypassing the QMF. The encoder output signals, IL and IH, are defined in the sub\(hyblock QUANTL and QUANTH, respectively. .PP This sequence is used for testing the quantizer/predictor feedback loop in the encoder. .bp .RT .ce \fBH.T. [T26.722]\fR .ce TABLE\ II\(hy1/G.722 .ce \fBDescription of input and output signals for test sequence\fR .ps 9 .vs 11 .nr VS 11 .nr PS 9 .TS center box; cw(24p) | cw(192p) . Name Description _ .T& lw(24p) | lw(192p) . XL { 15\(hybit uniformly quantized input signal to the lower sub\(hyband encoder } .T& lw(24p) | lw(192p) . XH { 15\(hybit uniformly quantized input signal to the higher sub\(hyband encoder } .T& lw(24p) | lw(192p) . X## { Input test sequence with 16\(hybit word format as shown in Figure\ II\(hy1/G.722 } .T& lw(24p) | lw(192p) . IL { 6\(hybit lower sub\(hyband ADPCM codeword } .T& lw(24p) | lw(192p) . ILR { Received 6\(hybit lower sub\(hyband ADPCM codeword } .T& lw(24p) | lw(192p) . IH { 2\(hybit higher sub\(hyband ADPCM codeword } .T& lw(24p) | lw(192p) . I## { Output (in Configuration 1) and Input (en Configuration 2) test sequence with 16\(hybit word format as shown in Figure II\(hy2/G.722 } .T& lw(24p) | lw(192p) . RL { 15\(hybit uniformly quantized output signal from the lower sub\(hyband decoder } .T& lw(24p) | lw(192p) . RH { 15\(hybit uniformly quantized output signal from the higher sub\(hyband decoder } .T& lw(24p) | lw(192p) . RL## { Output test sequence with 16\(hybit word format as shown in Figure II\(hy3/G.722 } .T& lw(24p) | lw(192p) . RH## { Output test sequence with 16\(hybit word format as shown in Figure II\(hy3/G.722 } .T& lw(24p) | lw(192p) . RSS Reset/synchronization signal .T& lw(24p) | lw(192p) . VI Valid data indication signal _ .TE .nr PS 9 .RT .ad r \fBTableau II\(hy1/G.722 [T26.722], p. 9\fR .sp 1P .RT .ad b .RT .LP .rs .sp 8P .ad r \fBFigure II\(hy1/G.722, p. 10\fR .sp 1P .RT .ad b .RT .LP .rs .sp 12P .ad r \fBFigure II\(hy2/G.722, p. 11\fR .sp 1P .RT .ad b .RT .LP .bp .LP .rs .sp 8P .ad r \fBFigure II\(hy3/G.722, p. 12\fR .sp 1P .RT .ad b .RT .LP .rs .sp 13P .ad r \fBFIGURE II\(hy4/G.722, p. 13\fR .sp 1P .RT .ad b .RT .sp 1P .LP II.2.2 \fIConfiguration 2\fR .sp 9p .RT .PP Configuration 2 shown in Figure II\(hy5/G.722 is a simplified version of Figures\ 7/G.722 and\ 8/G.722. The test signals, ILR and IH, and the MODE signal are described in Table\ 12/G.722. The corresponding decoder output signals, RL and RH, are defined in the sub\(hyblocks LIMIT in \(sc\(sc\ 6.2.1.6 and\ 6.2.2.5. For the lower sub\(hyband, the ADPCM decoder output signals are derived for three basic modes of operation (Modes\ 1, 2 and\ 3). By\(hypassing the QMF, the output signals, RL and RH, are separately obtained from the lower and higher sub\(hyband ADPCM decoders, respectively. .PP Configuration 2 is used for testing the inverse quantizer operation and the predictor adaptation without a quantizer/predictor feedback loop in the decoder. .RT .LP .rs .sp 16P .ad r \fBFigure II\(hy5/G.722, p.\fR .sp 1P .RT .ad b .RT .LP .bp .sp 1P .LP II.2.3 \fIReset/synchronization signal (RSS)\fR \fI and\fR \fIvalid data\fR \fIindication (VI)\fR .sp 9p .RT .PP All memory states in the two test configurations must be initialized to the exact states specified in this Recommendation prior to the start of an input test sequence in order to obtain the correct output values for the test. .PP In Configuration 1, the input test sequence, X##, is composed of encoder input test signals and the reset/synchronization signal (RSS) as shown in Figure\ II\(hy1/G.722. The RSS signal is located at the first LSB of the input sequence. If RSS is \*Q1\*U, the lower and higher sub\(hyband encoders are initialized, and the outputs of the encoders are set to \*Q0\*U, i.e., IH\ =\ \*Q0\*U and IL\ =\ \*Q0\*U. This normally forbidden output code is used to indicate \*Qnon\(hyvalid data\*U of the outputs. After the RSS signal goes to \*Q0\*U, the input test sequence will be valid and the ADPCM algorithm begins to operate. .PP In Configuration 2, the input test sequence, I##, is composed of the first 8\ bits of lower and higher sub\(hyband decoder input codewords, and the last 8\ bits consists of 7\(hybit zeroes and \*QRSS\*U in the LSB as shown in Figure\ II\(hy2/G.722. The RSS signal has the same role as in Configuration\ 1. That is, if the RSS signal equals \*Q1\*U, the lower and higher sub\(hyband decoders are initialized. After the RSS signal goes to \*Q0\*U, the ADPCM algorithm will be in the operational state. The output test sequences, RL## and RH##, are made up of a decoder output signal of 15\ bits and a valid data indication signal (VI) as shown in Figure\ II\(hy3/G.722. While the RSS signal to the decoder is \*Q1\*U, the signal \*QVI\*U is set to \*Q1\*U and the decoder output set to \*Q0\*U, which indicates \*Qnon\(hyvalid data\*U of the output. When \*QVI\*U is \*Q0\*U, the output test sequence is valid. .PP In order to establish the connection between the test sequence generator/receiver and the encoder/decoder, four sub\(hyblocks, INFA, INFB, INFC, INFD in Figures\ II\(hy4/G.722 and\ II\(hy5/G.722 are provided. A detailed expansion of these sub\(hyblocks is described below using the same notations specified in \(sc\ 6.2. .RT .ad r .ad b .RT .sp 1P .ce 1000 INFA .sp 9p .RT .ce 0 .sp 1P .LP Input: X## .LP Outputs: XL, XH, RS .LP Function: Extract reset/synchronization signal and input signals to lower and higher sub\(hyband ADPCM encoder. .LP RS = X## & 1 | Extract RSS signal .LP XL = S## > > 1 | Lower sub\(hyband input signal .LP XH = XL | Higher sub\(hyband input signal .ad r .ad b .RT .sp 1P .ce 1000 INFB .sp 9p .RT .ce 0 .sp 1P .LP Inputs: IL, IH, RS .LP Outputs: I## .LP Function: Create an output test sequence by combining lower and higher sub\(hyband ADPCM encoder output signals and the reset/synchronization signal. [Formula Deleted] .LP